- sr-users - kamailio.org

[Kamailio-Users] two endpoints
by Alex R.S.M 06 Nov '08

06 Nov '08

Hi, If ENUM lookup returns 2 destinations, does kamailio fork the message and send two INVITE? What function can be used to force kamailio to fork a message? Regards, Alex _________________________________________________________________ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster…

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[Kamailio-Users] rewriteuser with AVP
by Samuel Muller 06 Nov '08

06 Nov '08

Hello all, I've a little question : I got a username in an AVP by avp_load_radius, and I would rewrite the uri with it. The objectives is to replace the r-uri by an ipbx uri, then forward the call to it by a new branch (to not loose the phone number requested). in this case : 0123452000 is behind a spa9000. The user 0123452000 has an AVP called "ipbx", that i'm using to do some groups and permissions (plus the rewrite). actually, it does not work, i tried many ways : # load radius attributes of the callee if (!avp_load_radius("callee")) { xlog("L_INFO","-> user unknown in radius usr db : $ru"); route(15); # route PSTN } # verify the AVPs we got (caller and callee) xlog("L_INFO", " -AVP------------------------------------- "); avp_print(); xlog("L_INFO", " ----------------------------------------- "); # callee is behind an ipbx (avp ipbx) ? if (is_avp_set("$avp(s:callee_ipbx)")) { xlog("L_INFO", "-> callee is behind an ipbx : $avp(s:callee_ipbx)"); setflag(14); # flag IPBX #subst_user('/$rU$/$avp(s:callee_ipbx)/'); #rewriteuri("sip:$avp(s:callee_ipbx)@$rd"**); rewriteuser($avp(s:callee_ipbx)); } # callee in usrloc ? if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); xlog("L_INFO","-> registered user called : $tu"); route(7); # route FORWARD } else { xlog("L_INFO","-> 480: local user not found in usrloc : $tu"); sl_send_reply("480","Temporarily Unavailable"); drop; } and in the debugs log : Nov 5 13:04:51 ser0 kamailio[2930]: -ROUTE--INBOUND-------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: rc_auth Success Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP 'callee_asserted_id'/0='0123452000'/0 has been added Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP 'callee_ipbx'/0='spa9000'/0 has been added Nov 5 13:04:51 ser0 kamailio[2930]: -AVP------------------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5cfe8, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<callee_ipbx> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<spa9000 / 7> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5cf88, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<callee_asserted_id> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<0123452000 / 10> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5ced8, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<caller_asserted_id> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<0123451011 / 10> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5ce38, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: ----------------------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: -> callee is behind an ipbx : spa9000 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:registrar:lookup: '$avp( s(a)sip.720.fr' Not found in usrloc Nov 5 13:04:51 ser0 kamailio[2930]: -> 480: local user not found in usrloc : sip:0123452000@sip.720.fr <sip%3A0123452000(a)sip.720.fr> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:check_via_address: params 77.246.81.162, 192.168.0.134, 0 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:destroy_avp_list: destroying list 0x7fb3d7e5d040 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:receive_msg: cleaning up -> But, if I did it by : if (uri =~ "^sip:012345200{1}") { rewriteuri("sip:spa9000@sip.720.fr<sip%3Aspa9000(a)sip.720.fr> "); } if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); xlog("L_INFO","-> registered user called : $tu"); route(7); # route FORWARD } it works great, and the call is ok (the ipbx forward correctly the request the phone behind him). Anyone has an idea ? thanks in advance ! -- Samuel MULLER

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[Kamailio-Users] Fwd: Openser as Proxy and Asterisk as Registrar
by Cosimo Fadda 06 Nov '08

06 Nov '08

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 > > > >

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[Kamailio-Users] Fwd: Openser as Proxy and Asterisk as Registrar
by Cosimo Fadda 06 Nov '08

06 Nov '08

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 > >

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[Kamailio-Users] reminder: irc devel meeting in 1h20
by Daniel-Constantin Mierla 06 Nov '08

06 Nov '08

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

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Re: [Kamailio-Users] SER/Asterisk interworking mailing list
by luzango mfupe 06 Nov '08

06 Nov '08

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......."

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Re: [Kamailio-Users] Is it possible to break a "if" stament?
by ingdavidcespedes＠cable.net.co 06 Nov '08

06 Nov '08

----- 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 >

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[Kamailio-Users] high processing delay
by Fabian Borot 06 Nov '08

06 Nov '08

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/

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[Kamailio-Users] Is it possible to break a "if" stament?
by Iñaki Baz Castillo 05 Nov '08

05 Nov '08

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

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Re: [Kamailio-Users] high processing delay
by Daniel-Constantin Mierla 05 Nov '08

05 Nov '08

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

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[Kamailio-Users] Openser as Proxy and Asterisk as Registrar
by Cosimo Fadda 05 Nov '08

05 Nov '08

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.

2 1

[Serusers] DNS failover
by Nuno Ribeiro 05 Nov '08

05 Nov '08

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

2 2

[Kamailio-Users] SER/Asterisk interworking mailing list.
by Alex Balashov 05 Nov '08

05 Nov '08

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

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[Kamailio-Users] Load userblacklist table to cache w/o restarting?
by Pablo Bertuzzi 05 Nov '08

05 Nov '08

Hello forum, I was wondering whether there is a fifo command or workaround to load the userblacklist table into memory without restarting. I have a large carrierroute table and it takes about 20 secs to start. best regards, PB

2 3

Re: [Kamailio-Users] rewriteuser with AVP
by Samuel Muller 05 Nov '08

05 Nov '08

Hello Ravi , i'm using radiator but not freeradius, actually my users are stored into an user flat file. here you can have an example of user : 0123452000(a)sip.720.fr User-Password = "2000" User-Name = "0123452000", Sip-User-ID = "0123452000", Sip-User-Realm = "sip.720.fr", Sip-URI-User = "0123452000(a)sip.720.fr", Sip-Rpid = "0123452000", Sip-Group = "full", Sip-AVP = "asserted_id:0123452000" and concerning services attributes, you have to declare them like this : => in kamailio.cfg (or openser.cfg, ser.cfg, opensips.cfg) : modparam("acc|auth_radius", "service_type", 15) modparam("avp_radius", "caller_service_type", 30) modparam("avp_radius", "callee_service_type", 31) if you're using peering module : modparam("peering", "verify_destination_service_type", 21) modparam("peering", "verify_source_service_type", 22) => in the radius dictionary (for radiusclient-ng, and the dictionary used by your radius server) ### Service-Type Values ### VALUE Service-Type Call-Check 10 # RFC2865, uri_radius VALUE Service-Type Group-Check 12 # Proprietary, group_radius VALUE Service-Type Sip-Session 15 # Schulzrinne, acc/auth radius VALUE Service-Type Sip-Verify-Destination 20 # Proprietary, peering VALUE Service-Type Sip-Verify-Source 21 # Proprietary, peering VALUE Service-Type Sip-Caller-AVPs 30 # Proprietary, avp_radius VALUE Service-Type Sip-Callee-AVPs 31 # Proprietary, avp_radius => then, in your radius server config, you must add one handler per service-type example for radiator, just a base config based on flat user file: <Handler Service-Type=Sip-Verify-Destination> RewriteUsername s/^sip:(.*)/$1/ <AuthBy FILE> NoCheckPassword Filename %D/user.peer NoDefault </AuthBy> </Handler> I can't help you actually for a radius DB, 'cause i'm working on it. cheers, .Sam. -- Samuel MULLER Ingénieur Reseaux & Telecom 720 DEGRES +33 (0)663 128 505 sml(a)720.fr On Wed, Nov 5, 2008 at 1:28 PM, LetMeKnow < sunkara.raviprakash.feb14(a)gmail.com> wrote: > Hello Samuel, > Can you check the radius services attributes > > is you using users.conf or radius databases. > > can see this url > http://www.kamailio.org/docs/openser-radius-1.0.x.html#freeradius_users > > > > Thanks &Regards > Ravi Prakash Sunkara > VoIP Architect & JAVA-SIP Developer > +91-9999882776 > > > 2008/11/5 Samuel Muller <sml(a)720.fr> > >> Hello all, >> >> I've a little question : >> I got a username in an AVP by avp_load_radius, and I would rewrite the uri >> with it. >> >> The objectives is to replace the r-uri by an ipbx uri, then forward the >> call to it by a new branch (to not loose the phone number requested). >> in this case : 0123452000 is behind a spa9000. The user 0123452000 has an >> AVP called "ipbx", that i'm using to do some groups and permissions (plus >> the rewrite). >> >> actually, it does not work, i tried many ways : >> >> # load radius attributes of the callee >> if (!avp_load_radius("callee")) >> { >> xlog("L_INFO","-> user unknown in radius usr db : $ru"); >> route(15); # route PSTN >> } >> >> # verify the AVPs we got (caller and callee) >> xlog("L_INFO", " -AVP------------------------------------- "); >> avp_print(); >> xlog("L_INFO", " ----------------------------------------- "); >> >> # callee is behind an ipbx (avp ipbx) ? >> if (is_avp_set("$avp(s:callee_ipbx)")) >> { >> xlog("L_INFO", "-> callee is behind an ipbx : >> $avp(s:callee_ipbx)"); >> setflag(14); # flag IPBX >> #subst_user('/$rU$/$avp(s:callee_ipbx)/'); >> #rewriteuri("sip:$avp(s:callee_ipbx)@$rd"**); >> rewriteuser($avp(s:callee_ipbx)); >> } >> >> # callee in usrloc ? >> if (lookup("location")) >> { >> append_hf("P-hint: usrloc applied\r\n"); >> xlog("L_INFO","-> registered user called : $tu"); >> route(7); # route FORWARD >> } >> else >> { >> xlog("L_INFO","-> 480: local user not found in usrloc : >> $tu"); >> sl_send_reply("480","Temporarily Unavailable"); >> drop; >> } >> >> and in the debugs log : >> >> Nov 5 13:04:51 ser0 kamailio[2930]: >> -ROUTE--INBOUND-------------------------- >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: rc_auth >> Success >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP >> 'callee_asserted_id'/0='0123452000'/0 has been added >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP >> 'callee_ipbx'/0='spa9000'/0 has been added >> Nov 5 13:04:51 ser0 kamailio[2930]: >> -AVP------------------------------------- >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5cfe8, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Iname=<callee_ipbx> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Ival_str=<spa9000 / 7> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5cf88, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Iname=<callee_asserted_id> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Ival_str=<0123452000 / 10> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5ced8, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Iname=<caller_asserted_id> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> ^I^I^Ival_str=<0123451011 / 10> >> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: >> p=0x7fb3d7e5ce38, flags=0x0003 >> Nov 5 13:04:51 ser0 kamailio[2930]: >> ----------------------------------------- >> Nov 5 13:04:51 ser0 kamailio[2930]: -> callee is behind an ipbx : spa9000 >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:registrar:lookup: '$avp( >> s(a)sip.720.fr' Not found in usrloc >> Nov 5 13:04:51 ser0 kamailio[2930]: -> 480: local user not found in >> usrloc : sip:0123452000@sip.720.fr <sip%3A0123452000(a)sip.720.fr> >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:parse_headers: >> flags=ffffffffffffffff >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:check_via_address: params >> 77.246.81.162, 192.168.0.134, 0 >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:destroy_avp_list: destroying >> list 0x7fb3d7e5d040 >> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:receive_msg: cleaning up >> >> >> -> But, if I did it by : >> >> if (uri =~ "^sip:012345200{1}") >> { >> rewriteuri("sip:spa9000@sip.720.fr<sip%3Aspa9000(a)sip.720.fr> >> "); >> } >> if (lookup("location")) >> { >> append_hf("P-hint: usrloc applied\r\n"); >> xlog("L_INFO","-> registered user called : $tu"); >> route(7); # route FORWARD >> } >> >> it works great, and the call is ok (the ipbx forward correctly the request >> the phone behind him). >> >> >> Anyone has an idea ? thanks in advance ! >> >> >> -- >> Samuel MULLER >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

1 0

Re: [Kamailio-Users] Users Digest, Vol 42, Issue 11
by luzango mfupe 04 Nov '08

04 Nov '08

> 9. Re: the sip router project (Bogdan-Andrei Iancu) > > > Mmmh! guys, i am missing some points here, does this has anything to do > with OPENSIPS-YATE Collabo? a counter attack move of some sort??? > > > ------------------------------ > > Message: 9 > Date: Tue, 04 Nov 2008 17:24:15 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [Kamailio-Users] the sip router project > To: Johansson Olle E <oej(a)edvina.net> > Cc: devel <devel(a)lists.kamailio.org>, serusers(a)lists.iptel.org, > sr-dev(a)lists.sip-router.org, users <users(a)lists.kamailio.org>, SER > developer mailing list <serdev(a)iptel.org>, > business(a)lists.kamailio.org > Message-ID: <4910691F.9020509(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi Olle, > > Thank you for your thoughts. On a first view, it looks interesting, but I'm > missing some points here (important points): > > 1) as OpenSER was forked from SER because different views (and the OpenSER > view proved to be a very popular and successful one), I wonder > why, Kamilio is getting back to SER? not sharing any more the OpenSER view > as claimed? because such merging will definitely have a great > impact on the dynamical and openness of the projects (like releases, > contributions, driving the project) > > 2) this major change of perspective (at least for kamilio) was a backstage > decision, kept secret from the community - shouldn't be in the > interest of the community to say if going back to the roots (merging into > SER) is something wanted or not? it somehow contradicts the self > existence of OpenSER, right? > > 3) the benefits you mentions are mainly optimization of the internal > project activities and not optimizations of the outcome - what the > project will deliver. And I guess this is the most important. We already > went though the experience of large devel community, frameworks, etc but > with no outcome for more than 2 years... > > >From my personal perspective, the new project looks more like SER > absorbing Kamilio (considering the sizes, the companies behind each > project, the resources, and the man-power behind each project). > > And at the moment I would like preserve the OpenSER vision and to have an > open source project (a standalone one), far away from the "control" of any > Big Brother ;) > > Regards, > Bogdan > > > > Johansson Olle E wrote: > > Congratulations to both projects! > > > > I think this is very beneificial for both communities as well as for > > all the companies out there using the two products. > > > > I am very happy with this result and hope that the third kid in the > > SER family - OpenSIPS - will consider joining the effort too. > > > > Best regards, > > /Olle > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 42, Issue 11 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] the sip router project
by Daniel-Constantin Mierla 04 Nov '08

04 Nov '08

Hello Everybody, We are pleased to announce to you the SIP Router project. It aims to build a solid open source SIP routing platform, based on collaboration of the SIP Express Router (SER) and Kamailio (OpenSER) teams. Developers of these two projects believe that an united and non-conflicting environment will bring many benefits to them, community members and companies: * bring together the developers and user communities of both projects * reduce maintenance overhead * avoid duplicated efforts in development * develop a core framework that is flexible, extensible and scalable * promote and build a solid open source SIP server project * ensure business credibility * make future forking undesirable, this harms everybody, affects credibility and business You are welcome to join! Visit the web site at: http://sip-router.org There is a meeting in Karlsruhe, Germany, on Nov 10, 2008, hosted by 1&1, where the developers and community members have the chance to discuss and tune the last aspects of the new project. We are looking to see many of you there: http://sip-router.org/index.php/meeting/ We hope this is a great news for you, thanks to the effort of main developers and management teams of the two projects. We invite you to join the new mailing list for further discussions: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev Daniel, Jiri Kamailio Management Team SER Management Team

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[Kamailio-Users] Unknown column 'from_uri' in 'field list'
by Kaeman Chris 04 Nov '08

04 Nov '08

Hello, modparam("acc", "db_extra", "from_uri=$fu; to_uri=$tu") added 2 new fields in the acc table. Logs giving following errors while making calls Nov 4 07:14:11 [4567] ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'from_uri' in 'field list' Nov 4 07:14:11 [4567] ERROR:core:db_do_insert: error while submitting query Nov 4 07:14:11 [4567] ERROR:acc:acc_db_request: failed to insert into database Is this error due to the added fields? Thanks, KChris Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

3 2

[Kamailio-Users] SER config file adaptation to Kamailio ?
by Samuel Muller 04 Nov '08

04 Nov '08

Hello all, I tried during one week to adapt my old SER configuration file (ser.cfg, for SER 2.0.0), who works really great, to Kamailio v1.4.2. I used to change all the stuff needed (many functions, onsend route, ... disappeared or was replaced, ...). That's ok, concepts are the same - registration are ok, NAT (seems to) works with rtpproxy, and I'm using radius server (Radiator). MySQL is only here to store locations, domains, and trusted stuff like gateways. Everything looks fine (load kamalio : no pbm, REGISTER, OPTIONS methods and SIP ping ok, ...) Right now, RTP does not work, and no SIP phone (linksys spa9xx, thomson st2030, ...) can't call - there's no ringing, even if all the sip phones are registered and known by OpenSER ... It seems OpenSER can't send the SIP requests to the callee ??? Did anyone had the same problem and resolved it ? Or, eventually anyone does have some config sample config file (as you can have in SER examples stuff) ? I would like to let here a debug log of a call from a linksys spa942 to a sjphone but it's a bit enormous (debug log + config file) ... so i let it in attached file. the scenario is : 0123451011 (sjphone) calls 0123451012 (linksys spa9xx) the 2 UACs are on the same LAN, through the same router, with the same realm. there're registered and are stored into the location table in mySQL. but the problem is present in all the case we can have (different realms, different WAN connections, different SIP UACs ...) Thanks in advance ! -- Samuel MULLER

2 2

[Kamailio-Users] Remote-Party-ID header
by ALAEDDINE abbech 04 Nov '08

04 Nov '08

hi all, Please, what is the role of Remote-Party-ID header? thanks

5 12

[Kamailio-Users] PUA API send to certain IP
by Schumann Sebastian 04 Nov '08

04 Nov '08

Anca, All, I have the following scenario UE - OpenSER Proxy/Presence Server - OpenSER Registrar The proxy acts as proxy and presence server. If the UE sends a subscribe to user@registrar, it handles this subscription. If it publishes its state user@registrar it does as well. All registration/call control is proxied to the registrar. I try now to use the PUA module API to create PUBLISH on the Proxy/Presence Server. This works fine, the only issue is that it has the request-uri user@registrar, like UE generated subscriptions. This is actually not an issue, but wanted. The problem is that the proxy forwards the package to the registrar acc. R-URI. I want it sent to another IP. Is there the possibility to define the IP where the package should be sent to, i.e. presence server IP, to handle the packet there? Thanks! Best regards, Sebastian

3 6

[Serusers] How to use SRV records
by Someone beneath the blue sky 04 Nov '08

04 Nov '08

Hello all, I'm pretty new to the SIP world and appreciate if somebody can help me out. In the SerWeb login page I noticed a link "Have-My-Domain!", and it looks like that the result would be having something similar to email forwarding service. Suppose that I have my own domain (e.g. mydomain.com), and I have added an SRV record to it and pointed it to iptel's sip server. Then following the link I was able to create a new username (e.g. user01). Now based on what I've understood, whenever somebody calls the sip uri user01(a)mydomain.com my soft phone that is registered with the above account should ring (my soft phone is using user01 for username, mydomain.com for server and iptel.org for registerar and it seems that it is registered without any problem). My questions: 1. What prevents others from impersonating a new username in my domain ? It seems that if somebody knows that mydomain.com has an SRV record pointed to iptel.org, can create another username (user02(a)mydomain.com) and use it without my knowledge, what is worse is that my DNS provider doesn't have any logging options for DNS requests, so I'll never find out about it. 2. In my account in iptel ( user01(a)mydomain.com ), there is another sip uri ( something like 123456(a)mydomain.com ), which I guess is supposed to be used by somebody who uses a device without alphabets (for example if somebody wants to call me via sipbroker gateways), I've already tried *478 - 123456 ( *478 is the code for iptel), but it doesn't work and it goes to voice mail, how can I use sipbroker to call user01(a)mydomain.com ? Thanks a lot in advance and sorry for the long message ! Someone

2 1

[Kamailio-Users] Proxy IAX ?
by BERGANZ François 04 Nov '08

04 Nov '08

Is there a proxy IAX ? Openser can do it ? Thank you Cordialement BERGANZ François

7 10

[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

>> Hi all, >> i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: >> >> A -------------------------- SER ---------------------------------- B >> SIPp SIPp >> >> >> i have a problem using SER : >> I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. >> >> the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. >> >> i m putting below my config file ser.cfg >> >> i have also some questions: >> >> 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. >> 2/ also i have noticed that when using the command : serctl moni , i got the following message: >> root@amar-desktop:~# serctl moni >> [: 1074: ==: unexpected operator >> [: 1074: 0: unexpected operator >> >> despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. >> >> 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. >> >> i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. >> >> >> Amar >> >> >> >> >> _________________________________________________________________ >> You live life beyond your PC. So now Windows goes beyond your PC. >> http://clk.atdmt.com/MRT/go/115298556/direct/01/ > > _________________________________________________________________ > You live life beyond your PC. So now Windows goes beyond your PC. > http://clk.atdmt.com/MRT/go/115298556/direct/01/ _________________________________________________________________ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/

2 1

[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

> > Hi all, > i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: > > A -------------------------- SER ---------------------------------- B > SIPp SIPp > > > i have a problem using SER : > I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. > > the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. > > i m putting below my config file ser.cfg > > i have also some questions: > > 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. > 2/ also i have noticed that when using the command : serctl moni , i got the following message: > root@amar-desktop:~# serctl moni > [: 1074: ==: unexpected operator > [: 1074: 0: unexpected operator > > despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. > > 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. > > i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. > > > Amar > > > > > _________________________________________________________________ > You live life beyond your PC. So now Windows goes beyond your PC. > http://clk.atdmt.com/MRT/go/115298556/direct/01/ _________________________________________________________________ You live life beyond your PC. So now Windows goes beyond your PC. http://clk.atdmt.com/MRT/go/115298556/direct/01/

1 0

[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

Hi all, i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: A -------------------------- SER ---------------------------------- B SIPp SIPp i have a problem using SER : I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. i m putting below my config file ser.cfg i have also some questions: 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. 2/ also i have noticed that when using the command : serctl moni , i got the following message: root@amar-desktop:~# serctl moni [: 1074: ==: unexpected operator [: 1074: 0: unexpected operator despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. Amar _________________________________________________________________ You live life beyond your PC. So now Windows goes beyond your PC. http://clk.atdmt.com/MRT/go/115298556/direct/01/

1 0

[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 03 Nov '08

03 Nov '08

Hello, being the time for a new IRC devel meeting, I propose Thursday, Nov 6, 2008, at 15:00UTC, on channel #kamailio, server irc.freenode.net The focus is to sync the developers related the work since the last major release, roadmap to next major release as well as minor releases scheduling. Please add to the agenda at: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-agenda-dr… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 2

Re: [Kamailio-Users] re mote clients don't contact other remote clients
by Ricky Gutierrez 03 Nov '08

03 Nov '08

________________________________ Monday, November 3, 2008 3:28:30 AM arif.zaman Wrote: > Did you check the location table whether callee is registered before > forwarding any call? that is strange that I see in the table location it, registered : +----+----------+------------------------------------------+------------------------+------------------------------+ | id | username | contact | received | user_agent | +----+----------+------------------------------------------+------------------------+------------------------------+ | 1 | 122 | sip:122@192.168.0.60:5065 | sip:190.184.35.4:5065 | Linksys/SPA942-5.2.8 | | 2 | 201 | sip:201@192.168.11.30:5060;transport=udp | sip:192.168.11.30:5060 | Grandstream BT200 1.1.4.18 | | 4 | 202 | sip:202@192.168.10.30:5064;transport=udp | sip:190.212.5.155:5064 | Grandstream GXP2020 1.1.6.16 | +----+----------+------------------------------------------+------------------------+------------------------------+ regardsss rickygm ________________________________

1 0

[Kamailio-Users] INVITE accounting problem
by Kaeman Chris 03 Nov '08

03 Nov '08

Hello, I have a problem in accounting INVITE. When the call is connected with duration the INVITE, ACK, BYE are accounted in db. But the problem is when the call is disconnected before it is actually attempting to connect. i am getting only CANCEL accounted whereas INVITE is not accounted for the cancelled call. I have the xlogs showing INVITE and CANCEL both for this call only no accounting is performed for INVITE. following is the config for INVITE & CANCEL; if (is_method("INVITE")) { setflag(1); setflag(2); # MISSED if (allow_trusted()) { rewritehostport("xx.xxx.xy.xyz:5060"); route(1); } else { sl_send_reply("401", "Unauthorized"); route(1); exit; } } # CANCEL processing if (is_method("CANCEL")) { setflag(1); setflag(3); if (t_check_trans()) t_relay(); exit; } t_check_trans(); Please let me know what should be changed so that for the Call that is cancelled before it actually connects can have INVITE also accounted along w/ CANCEL. Thanks, KChris. Connect with friends all over the world. Get Yahoo! India Messenger at http://in.messenger.yahoo.com/?wm=n/

3 5

[Kamailio-Users] restrict calls from gateway to some extensions
by muhammad akl 03 Nov '08

03 Nov '08

Hey all , I want to make some ACL for the INVITES come from a PSTN-GW to certain users on my OpenSER server. for ex: I have the following extensions on my OpenSER server : 11001 11002 | | | | 11111 What I want to implement is something similar to this : Caller <-------> PSTN-GW <------->OpenSER <----> (complete this call to this extension ) 11001 Caller <-------> PSTN-GW <------->OpenSER <----> (reject this call for this user ) 11002 and so on for other extensions . any way to implement this feature in openser ?

2 2

[Kamailio-Users] remote clients don't contact other remote clients
by Ricky Gutierrez 03 Nov '08

03 Nov '08

Hello list, I have another question and a very strange detail, I have a couple of connected remote clients to my kamailio 1.3.2 and mediaproxy 1.9.1, when the client 122 calls to the client 202 this very contacts or, if I call a minute I contact him very well, but if I call to the client 122 after some minutes I don't contact with that client and the jumps me to the voicemail . I see in the chart location the address ip of the UAC that it can be happening? regards rickygm 122 UAC <-> NAT <-> internet==== kamailio 1.3.2 / mediaproxy ==== internet <-> ADSL <-> NAT <-> UAC 202 public ip: x.x.x.x.x interface: any filter: (ip) and ( port 5060 ) # U +0.890415 190.212.35.223:5064 -> 190.184.22.152:5060 INVITE sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=202 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000286 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9;received=190.212.35.223 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea16 Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 INVITE Proxy-Authenticate: Digest realm="190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.197976 190.212.35.223:5064 -> 190.184.22.152:5060 ACK sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea16 Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.073962 190.212.35.223:5064 -> 190.184.22.152:5060 INVITE sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: replaces, timer, path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.003823 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK7288d140481a11e9;rport=5064;received=190.212.35.223 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000053 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.485936 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.999801 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +1.999612 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.999219 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.999220 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +4.001208 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.599535 190.184.22.152:5060 -> 192.168.11.1:5070 INVITE sip:u122@192.168.11.1:5070 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000362 190.184.22.152:5070 -> 190.184.22.152:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1;received=190.184.22.152 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Length: 0 # U +0.000283 190.184.22.152:5070 -> 190.184.22.152:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1;received=190.184.22.152 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Type: application/sdp Content-Length: 289 v=0 o=root 9069 9069 IN IP4 190.184.22.152 s=session c=IN IP4 190.184.22.152 t=0 0 m=audio 15334 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.002152 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Type: application/sdp Content-Length: 289 P-hint: onreply_route|usemediaproxy v=0 o=root 9069 9069 IN IP4 190.184.22.152 s=session c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.236892 190.212.35.223:5064 -> 190.184.22.152:5060 ACK sip:u122@190.184.22.152:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK0fd7eaebf7c6f6c6 Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.000248 190.184.22.152:5060 -> 190.184.22.152:5070 ACK sip:u122@190.184.22.152:5070 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.3 Via: SIP/2.0/UDP 192.168.10.30:5064;received=190.212.35.223;branch=z9hG4bK0fd7eaebf7c6f6c6 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: LR|fixcontact,setflag6 exit 40 received, 0 dropped

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 02 Nov '08

02 Nov '08

Hi list, always looking for as solving my audio problem with mediaproxy asterisk and openser, there will be some form of telling to the openser that when he comes from the from sip:asterisk@192.168.10.1:5070 that doesn't use the mediaproxy or the onreply_route[1] , SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK36b7f619;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as42edbc9b;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as42edbc9b To: <sip:113@192.168.10.1>;tag=6d45d2188218c8ef Call-ID: 5a1c60382f18bd832fa3bdc54dc6ab13(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35064 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 making other tests if I change this form the onreply_route[1], I have audio in the openser extension, but the one that this behind the pstn doesn't have audio or he doesn't listen to me onreply_route[1] { # #-- On-replay block routing -- # if (client_nat_test("1")) { append_hf("P-hint: Onreply-route - fixcontact \r\n"); fix_nated_contact(); }; if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (search("^Content-Type:[ ]*application/sdp")) { append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); }; }; exit; } my best regardss rickygm

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

________________________________ Friday, October 31, 2008 2:59:58 AMDaniel-Constantin Mierla wrote: > do you have to bridge the RTP between the two network interfaces? >I do not know mediaproxy, but rtpproxy (nathelper module) has bridge mode >that can be used in such scenarios. Hi Daniel, I don't have a bridge with the two net cards, I had read some rtpproxy that can be used in bridge mode, but I didn't find alone examples some comments, but he gave me an error and I could not make it work this problem has me sad... and I am not seeing an exit.. regards .. rickygm

1 0

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

Thank you Luciano for your answer but doesn't it still work me, do the calls enter me of the pstn and don't I listen anything... where I can look for? , there will be some of form so that the IP of where the calls come from asterisk does not use mediaproxy, somebody has an example. any help is appreciated.. thank you rickygm ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> Sent: Friday, October 31, 2008 5:19:25 AM Subject: Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy Hi Ricky, I think you should increase the mediaproxy natping interval to 30 and then add use_media_proxy() line after you setflag(6). check below:--- ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); USE MEDIA PROXY HERE---> use_media_proxy(); }; route(1); } else { sl_send_reply("404", "Not here"); } exit; } On Fri, Oct 31, 2008 at 5:21 AM, Ricky Gutierrez <xserverlinux(a)yahoo.com> wrote: I show you here my openser.cfg my best regards rickygm loadmodule "nathelper.so" loadmodule "mediaproxy.so" modparam("registrar", "received_avp", "$avp(i:42)") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 4) modparam("nathelper","rtpproxy_sock", "") modparam("nathelper","natping_interval", 0) modparam("nathelper","received_avp", "$avp(i:42)") modparam("mediaproxy","natping_interval",20) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!is_method("REGISTER")) { if(nat_uac_test("19")){ record_route(";nat=yes"); } else { record_route(); }; }; if (has_totag()) { # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if(method=="BYE") { #Account BYE transactions setflag(2); }; if (method=="INVITE" && (!allow_trusted())) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if(method=="BYE" || method=="CANCEL") { end_media_session(); }; ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); }; route(1); } else { sl_send_reply("404", "Not here"); } exit; } ##Loose_route packets if (loose_route()) { if(!has_totag()){ sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; }; route(1); }; #CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { end_media_session(); t_relay(); }; exit; } t_check_trans(); if (method=="REGISTER"){ route(2); } else { route(3); }; } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { ; if (method=="INVITE" || method=="ACK") { ; end_media_session(); ; }; # sl_reply_error(); }; exit; } route[2] { # # -- Register request handler -- # if (is_uri_host_local()) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (!check_to()) { sl_send_reply("403", "Forbidden"); exit; }; sl_send_reply("100", "Trying"); if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { setbflag(6); fix_nated_register(); force_rport(); }; save("location"); exit; } else if { sl_send_reply("403", "Forbidden"); }; } route[3] { # # -- INVITE request handler -- # if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); }; if (is_from_local()){ # From an internal domain -> check the credentials and the FROM if(!allow_trusted()){ if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if (client_nat_test("3")) { append_hf("P-hint: route(3)|setflag7,forcerport,fix_contact\r\n"); setbflag(7); force_rport(); fix_contact(); }; if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(4); route(1); exit; } consume_credentials(); # Verify aliases lookup("aliases"); if (is_uri_host_local()) { # -- Inbound to Inbound route(10); } else { # -- Inbound to outbound route(11); }; } else { # From an external domain -> do not check credentials #Verify aliases, if found replace R-URI. lookup("aliases"); if (is_uri_host_local()) { #-- Outbound to inbound route(12); } else { # -- Outbound to outbound route(13); }; }; } route[6] { # # -- NAT handling -- # if (isbflagset(6) || isbflagset(7)) { append_hf("P-hint: Route[6]: mediaproxy \r\n"); use_media_proxy(); }; } route[10] { #from an internal domain -> inbound #Native SIP destinations are handled using the location table append_hf("P-hint: inbound->inbound \r\n"); if (uri=~"^sip:[2346578][0-9]{6}@.*") { if (is_user_in("credentials", "local")){ # rewritehostport("192.168.10.1:5070"); route(6); route(4); exit; } else { sl_send_reply("403", "No tienes permiso para llamadas locales"); exit; }; }; if (!lookup("location")) { if (does_uri_exist()) { revert_uri(); prefix("u"); rewritehostport("192.168.10.1:5070"); route(6); route(1); } else { sl_send_reply("404", "Not Found"); exit; }; sl_send_reply("404", "Not Found"); exit; }; route(6); route(1); } route[11] { # from an internal domain -> outbound # Simply route the call outbound using DNS search append_hf("P-hint: inbound->outbound \r\n"); route(1); } route[12] { # From an external domain -> inbound # Verify aliases, if found replace R-URI. lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; route(1); } route[13] { #From an external domain outbound #we are not accepting these calls append_hf("P-hint: outbound->inbound \r\n"); sl_send_reply("403", "Forbidden"); exit; } route[4] { rewritehostport("192.168.10.1:5070"); route(1); } ### aqui se detecta cunado una llamada es no disponible , ocupada o falla failure_route[1] { if (t_was_cancelled()) { exit; }; if (t_check_status("486")) { revert_uri(); prefix("b"); xlog("L_ERR","Stepped into the 486 ruri=<$ru>"); rewritehostport("192.168.10.1:5070"); append_branch(); route(1); exit; }; if (t_check_status("408") || t_check_status("480")) { revert_uri(); prefix("u"); xlog("L_ERR","Stepped into the 480 ruri=<$ru>"); rewritehostport("192.168.10.1:5070"); append_branch(); route(1); exit; }; } onreply_route[1] { # #-- On-replay block routing -- # if (client_nat_test("1")) { append_hf("P-hint: Onreply-route - fixcontact \r\n"); fix_nated_contact(); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (search("^Content-Type:[ ]*application/sdp")) { append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); }; }; exit; } -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Serusers] SER/FIFO not accessible
by Shruti Sanadhya 31 Oct '08

31 Oct '08

Hi all, I am trying to setup a SER to support calls on Kphone. I am trying to add users to the mysql SER database using: serctl 1000 1000 1000(a)mydomain.com but I am getting the 'SER/FIFO not accessible' error. I tried it using both the SER and root password for MySql, but no luck. I have also added 'fifo_mode=0777" in the configuration file, but the error still persists. Please advise, Thanks!!

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by luzango mfupe 31 Oct '08

31 Oct '08

Hi Ricky, I think you should increase the mediaproxy natping interval to 30 and then add use_media_proxy() line after you setflag(6). check below:--- ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); *USE MEDIA PROXY HERE---> use_media_proxy(); *}; route(1); } else { sl_send_reply("404", "Not here"); } exit; } On Fri, Oct 31, 2008 at 5:21 AM, Ricky Gutierrez <xserverlinux(a)yahoo.com>wrote: > > I show you here my openser.cfg > > my best regards > > rickygm > > loadmodule "nathelper.so" > loadmodule "mediaproxy.so" > > > modparam("registrar", "received_avp", "$avp(i:42)") > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "nat_bflag", 4) > > modparam("nathelper","rtpproxy_sock", "") > > modparam("nathelper","natping_interval", 0) > modparam("nathelper","received_avp", "$avp(i:42)") > modparam("mediaproxy","natping_interval",20) > modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") > modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") > > modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") > modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") > > > route{ > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > }; > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > if (!is_method("REGISTER")) { > if(nat_uac_test("19")){ > record_route(";nat=yes"); > } else { > record_route(); > }; > }; > > if (has_totag()) { > # subsequent messages withing a dialog should take the > # path determined by > record-routing > if (loose_route()) { > if(method=="BYE") { > #Account BYE transactions > setflag(2); > }; > if (method=="INVITE" && (!allow_trusted())) { > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > exit; > } else if (!check_from()) { > sl_send_reply("403", "Forbidden, use From=ID"); > exit; > > }; > > }; > if(method=="BYE" || method=="CANCEL") { > end_media_session(); > }; > > ##Detect requests in the dialog behind NAT and flag with 6 > if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ > append_hf("P-hint: LR|fixcontact,setflag6\r\n"); > fix_contact(); > setbflag(6); > }; > route(1); > } else { > sl_send_reply("404", "Not here"); > } > exit; > } > > ##Loose_route packets > if > (loose_route()) { > if(!has_totag()){ > sl_send_reply("403", "Initial Loose-Routing Rejected"); > exit; > }; > > > route(1); > }; > > > #CANCEL processing > if (is_method("CANCEL")) { > if (t_check_trans()) { > end_media_session(); > t_relay(); > }; > exit; > } > t_check_trans(); > > if (method=="REGISTER"){ > route(2); > } else { > route(3); > }; > } > > route[1] { > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > t_on_reply("1"); > t_on_failure("1"); > if (!t_relay()) { > ; if (method=="INVITE" || method=="ACK") { > ; end_media_session(); > ; }; > > > # sl_reply_error(); > }; > exit; > } > route[2] { > # > # -- Register request handler -- > # > if (is_uri_host_local()) { > if (!www_authorize("", "subscriber")) { > > www_challenge("", "0"); > exit; > }; > if (!check_to()) { > sl_send_reply("403", "Forbidden"); > exit; > }; > sl_send_reply("100", "Trying"); > if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { > setbflag(6); > fix_nated_register(); > force_rport(); > }; > > save("location"); > exit; > } else if { > sl_send_reply("403", "Forbidden"); > }; > } > > route[3] { > # > # -- INVITE request handler -- > # > if (method=="INVITE") { > # Account INVITE packets > setflag(2); > # Account Missed calls > setflag(3); > }; > if (is_from_local()){ > # From an internal domain -> check the credentials and the FROM > if(!allow_trusted()){ > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > exit; > } else if (!check_from()) { > sl_send_reply("403", "Forbidden, use From=ID"); > exit; > > }; > > }; > if (client_nat_test("3")) { > append_hf("P-hint: route(3)|setflag7,forcerport,fix_contact\r\n"); > setbflag(7); > force_rport(); > fix_contact(); > }; > > if(avp_db_load("$ru/username","$avp(s:callfwd)")) { > avp_pushto("$ru", "$avp(s:callfwd)"); > route(4); > route(1); > > exit; > } > > consume_credentials(); > > # Verify aliases > lookup("aliases"); > if (is_uri_host_local()) { > # -- Inbound to Inbound > > route(10); > } else { > # -- Inbound to outbound > route(11); > }; > > } else { > # From an external domain -> do not check credentials > #Verify aliases, if found replace R-URI. > lookup("aliases"); > if (is_uri_host_local()) { > #-- Outbound to inbound > route(12); > } else { > # -- Outbound to outbound > route(13); > }; > }; > } > route[6] { > > # > # -- NAT handling -- > # > if (isbflagset(6) || isbflagset(7)) { > append_hf("P-hint: Route[6]: mediaproxy \r\n"); > use_media_proxy(); > }; > } > > route[10] { > #from an internal domain -> inbound > #Native SIP destinations are handled using the location table > append_hf("P-hint: inbound->inbound \r\n"); > if (uri=~"^sip:[2346578][0-9]{6}@.*") { > if (is_user_in("credentials", "local")){ > # rewritehostport("192.168.10.1:5070"); > route(6); > route(4); > exit; > } else { > sl_send_reply("403", "No tienes permiso para llamadas locales"); > exit; > }; > }; > > if (!lookup("location")) { > if (does_uri_exist()) { > revert_uri(); > prefix("u"); > rewritehostport("192.168.10.1:5070"); > route(6); > route(1); > } else { > sl_send_reply("404", "Not Found"); > exit; > }; > sl_send_reply("404", "Not Found"); > exit; > }; > > > route(6); > route(1); > } > > route[11] { > # from an internal domain -> outbound > # Simply route the call outbound using DNS search > append_hf("P-hint: inbound->outbound \r\n"); > route(1); > } > route[12] { > # From an external domain -> inbound > # Verify aliases, if found replace R-URI. > lookup("aliases"); > if (!lookup("location")) { > sl_send_reply("404", "Not Found"); > exit; > }; > route(1); > } > route[13] { > #From an external domain outbound > #we are not accepting these calls > append_hf("P-hint: outbound->inbound \r\n"); > sl_send_reply("403", "Forbidden"); > exit; > } > > > route[4] { > rewritehostport("192.168.10.1:5070"); > route(1); > > } > > ### aqui se detecta cunado una llamada es no disponible , ocupada o falla > failure_route[1] { > if (t_was_cancelled()) { > exit; > }; > > if (t_check_status("486")) { > revert_uri(); > > prefix("b"); > xlog("L_ERR","Stepped into the 486 ruri=<$ru>"); > rewritehostport("192.168.10.1:5070"); > append_branch(); > route(1); > exit; > }; > if (t_check_status("408") || t_check_status("480")) { > revert_uri(); > prefix("u"); > xlog("L_ERR","Stepped into the 480 ruri=<$ru>"); > rewritehostport("192.168.10.1:5070"); > append_branch(); > route(1); > exit; > }; > } > onreply_route[1] { > # > #-- On-replay block routing -- > # > if (client_nat_test("1")) { > append_hf("P-hint: Onreply-route - fixcontact \r\n"); > fix_nated_contact(); > }; > > if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { > if (search("^Content-Type:[ ]*application/sdp")) { > append_hf("P-hint: onreply_route|usemediaproxy \r\n"); > use_media_proxy(); > }; > }; > > exit; > } > > > > > > > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

Hi list is making tests with openser 1.3.2 and mediaproxy to solve the nat, I have gotten myself an ip it public with my supplier, I have two network cards in the pc that I am using for openser and mediaproxy together with asterisk, making tests with mediaproxy 1.9.1 when I receive a call from the pstn through asterisk I don't have audio, if I call to the pstn they listen to me well . From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1>;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2 Contact: <sip:112@192.168.10.30:5060;transport=udp> Supported: path Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7492 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.022110 192.168.10.30:5060 -> 192.168.10.1:5060 INVITE sip:2685249@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03 From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Contact: <sip:112@192.168.10.30:5060;transport=udp> Supported: replaces, timer, path Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:2685249@192.168.10.1", nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e", response="c40a9387fdf5de29115c1edadc7f79db" Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 358 v=0 o=112 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 0 18 3 97 2 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.003938 192.168.10.1:5060 -> 192.168.10.30:5060 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03;rport=5060 From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000115 192.168.10.1:5060 -> 192.168.10.1:5070 INVITE sip:2685249@192.168.10.1:5070 SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03 From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Contact: <sip:112@192.168.10.30:5060;transport=udp> Supported: replaces, timer, path Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 358 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=112 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35040 RTP/AVP 0 18 3 97 2 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000471 192.168.10.1:5070 -> 192.168.10.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03 Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:2685249@192.168.10.1:5070> Content-Length: 0 I don't have a lot of experience with mediaproxy, and I have some doubts that such you see they can help me to clarify, inside the file mediaproxy.ini some options appear which I have configured them but I am not sure if it is the best way. my scenario is the following one: <-> UAC<-> NAT <-> ADSL <-> Internet <-> eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan (192.168.11.1) <-> UAC [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = openser listen = None allow = None proxyIP = x.x.x.x (public ip) ;portRange = 60000:65000 portRange = 35000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 [Accounting] ; one of none, radius or database accounting = none [Database] user = dbuser password = dbpass host = dbhost database = radius table = radacct [Radius] secret = secret server = localhost authport = 1812 acctport = 1813 dictionaries = /etc/radiusclient-ng/dictionary, /etc/openser/radius/dictionary, /usr/share/mediaproxy/dictionary retries = 2 timeout = 3 this couple of you line inside the openser, I don't still understand them according to the guide of ser getting started they are for asymmetric clients, but I don't find an example modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") somebody that can give me a good help... regards rickygm

2 1

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

I don't have problems when I make calls to the pstn I listen well and people listen to me well, the problem is when I receive a call from the pstn I don't listen anything and they don't listen to me, inside the sip.conf already has configured the values nat, externip localnet . I believe that the problem is that openser detects as nat the ip of my asterisk, eye > "I have the openser and the mediaproxy with asterisk in the same pc" ### Sip Log Asterisk #### <--- SIP read from 192.168.10.1:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK5839d960;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: Onreply-route - fixcontact <-------------> --- (12 headers 0 lines) --- -- SIP/openser-08c0ea58 is ringing xserver*CLI> <--- SIP read from 192.168.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK5839d960;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (15 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.64:35004 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.64:35004 list_route: hop: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> set_destination: Parsing <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> for address/port to send to set_destination: set destination to 192.168.10.1, port 5060 Transmitting (NAT) to 192.168.10.1:5060: ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK32b6019c;rport Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/openser-08c0ea58 answered Zap/4-1 xserver*CLI> <--- SIP read from 192.168.10.1:5060 ---> BYE sip:asterisk@192.168.10.1:5070 SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=a72df908ec08f63d> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK71b2.32123901.0 Via: SIP/2.0/UDP 192.168.10.30:5062;branch=z9hG4bK02603cb0e798dac0 From: <sip:113@192.168.10.1>;tag=a72df908ec08f63d To: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f Supported: path Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 9793 BYE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: LR|fixcontact,setflag6 <-------------> --- (14 headers 0 lines) --- Sending to 192.168.10.1 : 5060 (NAT) <--- Transmitting (NAT) to 192.168.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK71b2.32123901.0;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.30:5062;branch=z9hG4bK02603cb0e798dac0 Record-Route: <sip:192.168.10.1;lr=on;ftag=a72df908ec08f63d> From: <sip:113@192.168.10.1>;tag=a72df908ec08f63d To: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 9793 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:asterisk@192.168.10.1:5070> Content-Length: 0 ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> Hi RickyI should have seen how you handle NAT in kamaiilo.conf but you can also edit sip.conf in Asterisk and try to put Nat=yes Rgds,

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[Kamailio-Users] Fw: version cdrtool mediaproxy 1.9.1
by Ricky Gutierrez 30 Oct '08

30 Oct '08

OK thank you Nuno to answer, but I continue thinking that that version is not compatible with mediaproxy 1.9.1, for comments that read by the web, but better we ask to ag-projects Dan .. my best regards rickygm ________________________________ From: Nuno Marques <yangsengopenser(a)gmail.com> To: Ricky Gutierrez <xserverlinux(a)yahoo.com> Cc: users(a)lists.kamailio.org Sent: Wednesday, October 29, 2008 5:24:35 PM Subject: Re: version cdrtool mediaproxy 1.9.1 Hi Ricky, I'm using CDRTool 6.6.10 (latest). I still can't establish connection between CDRTool and mediaproxy database... but i'm almost there ! :) Nuno 2008/10/29 Ricky Gutierrez <xserverlinux(a)yahoo.com> Hi nuno, excuse me but that version of cdrtool you used for mediaproxy 1.9.1? , I am trying to do the same a server billing with openser and asterisk .. best regardss rickygm

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[Kamailio-Users] Dimensioning a telephony system based on openser!
by Nuno Marques 30 Oct '08

30 Oct '08

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work with this requisites: - 10000 users; - 100 VoIP to VoIP calls simultaneously capacity; - 30 VoIP to PSTN calls simultaneously capacity; Can anyone point me some ideas of how can i design such a system (how many servers, how to distribute the services among them, etc.). I have this prototype mounted with VMWare, so i think that even making tests with sipp aren't going to be reliable. Thanks in advance, Nuno

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Re: [Kamailio-Users] Kamailio/Openser - improve the performance with children process
by Daniel Cardoso Alves 30 Oct '08

30 Oct '08

Hi Henning Westerholt, Thanks for you reply... But I would like to know if there is a number that improve the performance... For example: If come 100 INVITE at the same moment, is there a NUMBER OF CHILDREN that has the best perfomance? The number of children depends on something? Thanks --- Em qua, 29/10/08, Henning Westerholt <henning.westerholt(a)1und1.de> escreveu: De: Henning Westerholt <henning.westerholt(a)1und1.de> Assunto: Re: [Kamailio-Users] Kamailio/Openser - improve the performance with children process Para: users(a)lists.kamailio.org Cc: dcardosoa(a)yahoo.com.br Data: Quarta-feira, 29 de Outubro de 2008, 11:25 On Wednesday 29 October 2008, Daniel-Constantin Mierla wrote: > > Is there a number of children that the Kamailio/Openser for improve > > the performance? > > > > Each process takes care of a request? > > yes, each process takes care of a SIP message (request/reply). Hi Daniel Cardoso, take a look to the core cookbook in our wiki [1] for more informations how you can control the number of childrens per protocol (UDP, TCP..). Cheers, Henning [1] http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:devel#children Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

4 3

[OpenSER-Users] Openser1.3.2 + snmp
by Ricardo Alves Ferreira Silva 30 Oct '08

30 Oct '08

Hi all, Trying to install Openser1.3.2.... everything is OK and fine... but when I try to add support for snmp there are errors.. Dig dep into google, recheck all dependencies and didn't succeed... My env is FreeBSD 7.0-STABLE i386 The full net-snmp package is installed: atila# pkg_info | grep net-snmp net-snmp-5.3.2_3 An extendable SNMP implementation These are the error messages.... make modules/libsnmpstats modules.... Compiling alarm_checks.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -DMOD_NAME='"snmpstats"' -DNAME='"openser"' -DVERSION='"1.3.2-notls"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 4.2.1"' -D__CPU_i386 -D__OS_freebsd -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -DHAVE_TIMEGM -DHAVE_NETINET_IN_SYSTM -DHAVE_KQUEUE -DHAVE_SELECT -c alarm_checks.c -o alarm_checks.o In file included from alarm_checks.c:38: openserObjects.h:41:38: error: net-snmp/net-snmp-config.h: No such file or directory openserObjects.h:42:40: error: net-snmp/net-snmp-includes.h: No such file or directory openserObjects.h:43:52: error: net-snmp/agent/net-snmp-agent-includes.h: No such file or directory If I edit each file and hardcode the path ... /usr/local/include/net-snmp/... there are no errors... What am I missing in making sure the path to the include files are OK... Tried a couple options but didn't succeed... tehre are tons of include files into the src.... Regards Ricardo Ferreira Yes, we do run Openser www.vipway.com.br Enterprise VoIP Services

5 10

[Serusers] Error in jabber module - XJAB:extract_aor: Error while parsing URI
by Tomasz Zieleniewski 30 Oct '08

30 Oct '08

Hi, We are using ser 2.0 We encountered the following error: 7(407) route[PRESENCE_SUBSCRIBE](SUBSCRIBE cseq:933): (handle_subscription(jabber)) 7(407) XJAB:extract_aor: Error while parsing URI But still SER sends an 200 OK response and after then a NOTIFY. Here is the message received by ser: SUBSCRIBE sip:lkl%touk.pl@jabber.rd.touk.pl SIP/2.0 Via: SIP/2.0/UDP 192.168.0.165;rport;branch=z9hG4bKxakietyi Max-Forwards: 70 Proxy-Authorization: Digest username="tzl",realm="touk.pl",nonce="490987cb2702bb9db8d631e016825ad1c9cd12ac",uri="sip:lkl%touk.pl@jabber.rd.touk.pl",response="bd4474cf8af3d1ec8a7246872c0ecb46",algorithm=MD5,cnonce="mk57n39293",qop=auth,nc=00000001 To: <sip:lkl%touk.pl@jabber.rd.touk.pl> From: "Tomasz Zieleniewski" <sip:tzl@sip.rd.touk.pl>;tag=cejrf Call-ID: vxiulpsynbnidjo(a)sen.rd.touk.pl CSeq: 933 SUBSCRIBE Contact: <sip:tzl@192.168.0.165> Accept: application/pidf+xml Event: presence Expires: 3600 User-Agent: Twinkle/1.2 Content-Length: 0 Jabber config: # -- jabber params --- modparam("jabber", "jaddress", "touk.pl") modparam("jabber", "jport", 5222) modparam("jabber", "registrar", "serimgw(a)sip.rd.touk.pl") modparam("jabber","workers",2) modparam("jabber","max_jobs",10) modparam("jabber","cache_time",200) modparam("jabber","delay_time",60) modparam("jabber","jdomain","jabber.rd.touk.pl=%") Please give us a hint what could be wrong. Kind regards Tomasz

1 0

[Serusers] Configuring SER as an outbound proxy for multiple Ekigas on a LAN
by Enrico Maria Crisostomo 30 Oct '08

30 Oct '08

Hi all. I'm a first-time would-be user of SER and I have the following problem: I was planning to use Ekiga on a LAN with several workstations and I was looking for documentation about the options I have and the best setup I can achieve. How to solve the problem of NAT traversal for multiple Ekigas was answered by Ekiga documentation: use a SIP proxy. The workstations are all equipped with Solaris 10 and I have a small Solaris 10 server too, so I thought it was natural for me to use SER as a SIP proxy. Unfortunately I'm finding no documentation about how to configure it as a simple outbound proxy and to say the truth I found very little documentation about configuring a SIP proxy in general. My idea would be just having SER behave as an outbound SIP proxy and use ekiga.net accounts for everybody (no need for a PBX as Asterisk). It seems a kind of basic configuration but I have no idea where to start from. Do you have suggestions and pointers to documentation to help me achieve this goal? Thank you in advance, Enrico. -- Ελευθερία ή θάνατος GPG key: 1024D/FD2229AF fpr: 9E07 D40E 33A5 5993 6FC5 09A8 5BCF B1F2 FD22 29AF

2 1

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 30 Oct '08

30 Oct '08

Hi luzano thank you for your help and time, this it is my full ngrep. I have asterisk, mediaproxy and openser together in the same pc I have the doubt that when a incoming call from the PSTN by asterisk, sends it to an extension of openser, to which I do not request authentication to him within invites, I believe that there it is where I have problems with mediaproxy that it leaves you want to see of the openser.cfg, or everything? regards .. interface: any filter: (ip) and ( port 5060 ) # U +0.063740 192.168.10.1:5070 -> 192.168.10.1:5060 INVITE sip:113@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 9850 9850 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.10.1 t=0 0 m=audio 14750 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.004203 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000275 192.168.10.1:5060 -> 192.168.10.30:5062 INVITE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=root 9850 9850 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.1.64 t=0 0 m=audio 35058 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.026845 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Content-Length: 0 # U +0.009886 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.000169 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: Onreply-route - fixcontact # U +0.000140 190.184.35.4:5060 -> 192.168.1.64:5064 OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 To: <sip:130@192.168.1.64:5064> Contact: <sip:asterisk@190.184.35.4> Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064 OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 To: <sip:130@192.168.1.64:5064> Contact: <sip:asterisk@190.184.35.4> Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.008208 192.168.10.19:5064 -> 190.184.35.4:5060 SIP/2.0 200 OK To: <sip:130@192.168.1.64:5064>;tag=9611a90af8ab9321i1 From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces # U +0.000014 192.168.1.64:5064 -> 190.184.35.4:5060 SIP/2.0 200 OK To: <sip:130@192.168.1.64:5064>;tag=9611a90af8ab9321i1 From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces # U +0.956449 192.168.10.19:5064 -> 190.184.35.4:5060 NOTIFY sip:190.184.35.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4> Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.000027 192.168.1.64:5064 -> 190.184.35.4:5060 NOTIFY sip:190.184.35.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4> Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.156145 190.184.35.4:5060 -> 192.168.1.64:5064 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4>;tag=as2002c003 Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4>;tag=as2002c003 Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +5.895003 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.001400 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35058 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000557 192.168.10.1:5070 -> 192.168.10.1:5060 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 # U +0.000215 192.168.10.1:5060 -> 192.168.10.30:5062 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.2 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 P-hint: LR|fixcontact,setflag6 # U +0.886545 192.168.10.28:5060 -> 192.168.10.1:5060 ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> To: users(a)lists.kamailio.org Sent: Wednesday, October 29, 2008 5:12:29 AM Subject: Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy Hi Ricky, Where is your Kamailio config?? is this your full ngrep capture?? Rgds, Luzango.

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[Kamailio-Users] kamailio and rtpproxy-no audio
by Fabian Borot 30 Oct '08

30 Oct '08

hello I am trying to use rtpproxy and kamailio. Below please find my kamailio.cfg file. But this is what is happening: 1- ext 101[IP 76.109.183.2] is able to send audio to kamailio's IP [65.111.185.187]but it does not reach ext 100 IP [76.109.15.75]. 2- ext 100 does not send RTP at all. 3- I still see the private IPs in the SDPs of both exts. I am aattaching a wireshark capture Thank you in advance Fabian # $Id: kamailio.cfg 4576 2008-08-05 16:22:38Z klaus_darilion $ # # Kamailion basic configuration script # by Anca Vamanu <anca(a)voice-system.ro> # # Please refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = '/usr/local/etc/kamailio/tls/user/user-cert.pem' #tls_private_key = '/usr/local/etc/kamailio/tls/user/user-privkey.pem' #tls_ca_list = '/usr/local/etc/kamailio/tls/user/user-calist.pem' port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:65.111.185.187:5060 ####### Modules Section ######## #set module path mpath='/usr/local/lib/kamailio/modules/' /* uncomment next line for MySQL DB support */ loadmodule 'db_mysql.so' loadmodule 'sl.so' loadmodule 'tm.so' loadmodule 'rr.so' loadmodule 'maxfwd.so' loadmodule 'usrloc.so' loadmodule 'registrar.so' loadmodule 'textops.so' loadmodule 'mi_fifo.so' loadmodule 'uri_db.so' loadmodule 'uri.so' loadmodule 'xlog.so' loadmodule 'acc.so' /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule 'auth.so' loadmodule 'auth_db.so' /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule 'alias_db.so' /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see 'multi-module params' section ) */ #loadmodule 'domain.so' /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule 'presence.so' #loadmodule 'presence_xml.so' loadmodule 'nathelper.so' # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam('mi_fifo', 'fifo_name', '/tmp/kamailio_fifo') # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam('rr', 'enable_full_lr', 1) # do not append from tag to the RR (no need for this script) modparam('rr', 'append_fromtag', 0) # ----- rr params ----- modparam('registrar', 'method_filtering', 1) /* uncomment the next line to disable parallel forking via location */ # modparam('registrar', 'append_branches', 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam('registrar', 'max_contacts', 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam('uri_db', 'use_uri_table', 0) modparam('uri_db', 'db_url', '') # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam('acc', 'early_media', 1) modparam('acc', 'report_ack', 1) modparam('acc', 'report_cancels', 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable 'append_fromtag' in 'rr' module */ modparam('acc', 'detect_direction', 0) /* account triggers (flags) */ modparam('acc', 'failed_transaction_flag', 3) modparam('acc', 'log_flag', 1) modparam('acc', 'log_missed_flag', 2) /* uncomment the following lines to enable DB accounting also */ modparam('acc', 'db_flag', 1) modparam('acc', 'db_missed_flag', 2) # ----- usrloc params ----- modparam('usrloc', 'db_mode', 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam('usrloc', 'db_mode', 2) #modparam('usrloc', 'db_url', # 'mysql://openser:openserrw@localhost/openser') modparam('usrloc', 'nat_bflag', 3) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam('auth_db', 'calculate_ha1', yes) modparam('auth_db', 'password_column', 'password') modparam('auth_db', 'db_url', 'mysql://openser:openserrw@localhost/openser') modparam('auth_db', 'load_credentials', '') # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam('alias_db', 'db_url', # 'mysql://openser:openserrw@localhost/openser') # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam('domain', 'db_url', # 'mysql://openser:openserrw@localhost/openser') #modparam('domain', 'db_mode', 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam('alias_db|auth_db|usrloc|uri_db', 'use_domain', 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam('presence|presence_xml', 'db_url', # 'mysql://openser:openserrw@localhost/openser') #modparam('presence_xml', 'force_active', 1) #modparam('presence', 'server_address', 'sip:192.168.1.2:5060') # ----- nathelper params ----- modparam('nathelper', 'natping_interval', 10) modparam('nathelper', 'ping_nated_only', 1) modparam('nathelper', 'rtpproxy_sock', 'udp:localhost:8899') modparam('nathelper', 'rtpproxy_disable_tout', 20) modparam('nathelper', 'rtpproxy_tout', 2) modparam('nathelper', 'rtpproxy_retr', 2) modparam('nathelper', 'sipping_from', 'sip:pinger@mykamailio.com') modparam('nathelper', 'sipping_bflag', 7) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header('10')) { sl_send_reply('483','Too Many Hops'); exit; } if (is_method('REGISTER')) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize('', 'subscriber')) { www_challenge('', '0'); exit; } if (!save('location')) sl_reply_error(); exit; } if (!lookup('location')) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply('404', 'Not Found'); exit; case -2: sl_send_reply('405', 'Method Not Allowed'); exit; } } # when routing via usrloc, log the missed calls also setflag(2); # record routing if (!is_method('REGISTER|MESSAGE')) { record_route(); } # account only INVITEs if (is_method('INVITE')) { setflag(1); # do accounting } # for INVITEs enable some additional helper routes if (is_method('INVITE')) { force_rtp_proxy(); } if (!t_relay()) { sl_reply_error(); }; exit; # CANCEL processing if (is_method('CANCEL|BYE')) { unforce_rtp_proxy(); if (t_check_trans()) t_relay(); exit; } } _________________________________________________________________ Stay organized with simple drag and drop from Windows Live Hotmail. http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_102008

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[Kamailio-Users] version cdrtool mediaproxy 1.9.1
by Ricky Gutierrez 29 Oct '08

29 Oct '08

Hi nuno, excuse me but that version of cdrtool you used for mediaproxy 1.9.1? , I am trying to do the same a server billing with openser and asterisk .. best regardss rickygm

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 29 Oct '08

29 Oct '08

Hi I download Twinkle, and install all the prerequisites, except for Qt. I have some problems with it. First I download Qt for Linux/X11: C++ "qt-x11-opensource-src-4.4.3.tar" I really don't know if it is the one I have to download, because In the Twinkle page, it only says you have to install "Qt for open source (3.3.0 or later, but not Qt4)". Then I do all the steps cd /tmp gunzip qt-x11-opensource-src-4.4.3.tar.gz # uncompress the archive tar xvf qt-x11-opensource-src-4.4.3.tar # unpack it cd /tmp/qt-x11-opensource-src-4.4.3 ./configure #It took more than 6 hours make su -c "make install" Then I retry to install Twinkle entering in Twinkle directory an doing ./configure but I receive the following error: checking for qt-mt >= 3.3.0 qt-mt < 4.0... Package qt-mt was not found in the pkg-config search path. Perhaps you should add the directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable No package 'qt-mt' found Package qt-mt was not found in the pkg-config search path. Perhaps you should add the directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable No package 'qt-mt' found configure: error: Library requirements (qt-mt >= 3.3.0 qt-mt < 4.0) not met; consider adjusting the PKG_CONFIG_PATH environment variable if your libraries are in a nonstandard prefix so pkg-config can find them. So I read the FAQ of Twinke (http://www.xs4all.nl/~mfnboer/twinkle/index.html) an it says that I can skip this error by doing ./configure --disable-qt-check, but now I receive the following error: checking value of $QTDIR... /usr/lib/qt-3.3 checking for qmake... no configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is incorrect. So I realize that I didn't have libqt-mt so I try to recompile QT with "./configure -thread" but it didn't let me, so I try with "./configure -DQT_THREAD_SUPPORT" and the compilation took also mother tan 6 hours, and I retry to install Twinkle, but without success, I receive the same message. checking value of $QTDIR... /usr/lib/qt-3.3 checking for qmake... no configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is incorrect. I don't Know what I am doing wrong. Any Help!!!!! please. David Cespedes. ----- Mensaje original ----- De: Iñaki Baz Castillo <ibc(a)aliax.net> Fecha: Martes, Octubre 28, 2008 8:15 am Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > 2008/10/28 Daniel-Constantin Mierla <miconda(a)gmail.com>: > > is twinkle able now to display rich presence status? Last time I > tried> was showing only online/offline status. > > For now just online/offline status. > > But it has other great features as: > - Indication of Message Composition for Instant Messaging - RFC 3994 > - PingPong TCP keepalive. > - TCP/UDP > - PRACK > - q-value in registration. > > For me it's the most reliable softphone. > > > -- > Iñaki Baz Castillo > <ibc(a)aliax.net> > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 29 Oct '08

29 Oct '08

Excuse me list, it was an error copy to this list, but maybe some one can help me. ----- Mensaje original ----- De: Alex Balashov <abalashov(a)evaristesys.com> Fecha: Miércoles, Octubre 29, 2008 11:31 am Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > This is not the Twinkle list. > > ingdavidcespedes(a)cable.net.co wrote: > > > Hi > > > > I download Twinkle, and install all the prerequisites, except for > Qt. I have some problems with it. First I download Qt for > Linux/X11: C++ "qt-x11-opensource-src-4.4.3.tar" I really don't > know if it is the one I have to download, because In the Twinkle > page, it only says you have to install "Qt for open source (3.3.0 > or later, but not Qt4)". Then I do all the steps > > > > cd /tmp > > gunzip qt-x11-opensource-src-4.4.3.tar.gz # > uncompress the archive > > tar xvf qt-x11-opensource-src-4.4.3.tar # unpack it > > cd /tmp/qt-x11-opensource-src-4.4.3 > > ./configure #It took more than 6 > hours> make > > su -c "make install" > > Then I retry to install Twinkle entering in Twinkle directory an > doing ./configure but I receive the following error: > > > > checking for qt-mt >= 3.3.0 qt-mt < 4.0... Package qt-mt was not > found in the pkg-config search path. Perhaps you should add the > directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment > variable No package 'qt-mt' found Package qt-mt was not found in > the pkg-config search path. Perhaps you should add the directory > containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable > No package 'qt-mt' found > > configure: error: Library requirements (qt-mt >= 3.3.0 qt-mt < > 4.0) not met; consider adjusting the PKG_CONFIG_PATH environment > variable if your libraries are in a nonstandard prefix so pkg- > config can find them. > > > > So I read the FAQ of Twinke > (http://www.xs4all.nl/~mfnboer/twinkle/index.html) an it says that > I can skip this error by doing ./configure --disable-qt-check, but > now I receive the following error: > > > > checking value of $QTDIR... /usr/lib/qt-3.3 > > checking for qmake... no > > configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is > incorrect.> > > So I realize that I didn't have libqt-mt so I try to recompile QT > with "./configure -thread" but it didn't let me, so I try with > "./configure -DQT_THREAD_SUPPORT" and the compilation took also > mother tan 6 hours, and I retry to install Twinkle, but without > success, I receive the same message. > > > > checking value of $QTDIR... /usr/lib/qt-3.3 > > checking for qmake... no > > configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is > incorrect.> > > I don't Know what I am doing wrong. Any Help!!!!! please. > > > > David Cespedes. > > > > ----- Mensaje original ----- > > De: Iñaki Baz Castillo <ibc(a)aliax.net> > > Fecha: Martes, Octubre 28, 2008 8:15 am > > Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > > > >> 2008/10/28 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >>> is twinkle able now to display rich presence status? Last time > I > >> tried> was showing only online/offline status. > >> > >> For now just online/offline status. > >> > >> But it has other great features as: > >> - Indication of Message Composition for Instant Messaging - RFC > 3994>> - PingPong TCP keepalive. > >> - TCP/UDP > >> - PRACK > >> - q-value in registration. > >> > >> For me it's the most reliable softphone. > >> > >> > >> -- > >> Iñaki Baz Castillo > >> <ibc(a)aliax.net> > >> _______________________________________________ > >> Users mailing list > >> Users(a)lists.kamailio.org > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > -- > 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 >

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[Kamailio-Users] Fwd: How to enable local_route feature ?
by Pascal Maugeri 29 Oct '08

29 Oct '08

Hi Following your instructions I have enable the compilation defined in Makefile.defs and recompile everything after a make clean. One capture of the compilation for you to see the define is enable: [...] gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"uri_db"' -DNAME='"kamailio"' -DVERSION='"1.4.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -c checks.c -o checks.o [...] When I try to start the proxy it complains about: CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: syntax error CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: invalid local_route statement As you announced this feature for OpenSER 1.4 release series, I assume this is available for kamailio v1.4.2 ? Best regards, Pascal ---------- Forwarded message ---------- From: Pascal Maugeri <pascal.maugeri(a)gmail.com> Date: Wed, Oct 29, 2008 at 11:17 AM Subject: How to enable local_route feature ? To: Henning Westerholt <henning.westerholt(a)1und1.de> Cc: devel(a)lists.openser.org, openser users <users(a)lists.openser.org> Hi Following your instructions I have enable the compilation defined in Makefile.defs and recompile everything after a make clean. One capture of the compilation for you to see the define is enable: [...] gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"uri_db"' -DNAME='"kamailio"' -DVERSION='"1.4.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -c checks.c -o checks.o [...] When I try to start the proxy it complains about: CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: syntax error CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: invalid local_route statement As you announced this feature for OpenSER 1.4 release series, I assume this is available for kamailio v1.4.2 ? Best regards, Pascal On Thu, Jun 26, 2008 at 1:11 PM, Henning Westerholt < henning.westerholt(a)1und1.de> wrote: > On Friday 06 June 2008, Bogdan-Andrei Iancu wrote: > > OpenSER has a new type of route: local route. This route is to be > > executed when a brand new request is internally generated (openser being > > the UAC) ; such requests are result of : > > - t_uac_dlg MI command > > - msilo, sms, presence, cpl-c, dialog (and others) modules > > [..] > > Hello all, > > the addition of this feature short before the code freeze caused some > conflicts, which could not resolved sufficently in time for this release. > Thus the OpenSER board was asked to decide in this matter. The majority of > the board voted for decativating this feature in the 1.4 release series. > > For more details please take a look at the developer mailing list at [1], > [2] > and [3]. Details about the "dectivate" commit can be found at [4] > > The local route functionality is still available, but its not activated in > the > default installation. In order to use this feature its necessary to set > the "USE_LOCAL_ROUTE" define in the "Makefile.defs" file and compile your > own > binary from the source. Then it will possible to use this feature. > > In the 1.5.0 release this functionality (or something resemble this) will > be > present, fully usable and supported. > > With best regards, > > Henning Westerholt > > > [1] http://lists.openser.org/pipermail/devel/2008-June/014028.html > [2] http://lists.openser.org/pipermail/devel/2008-June/014122.html > [3] http://lists.openser.org/pipermail/devel/2008-June/014204.html > [4] http://lists.openser.org/pipermail/devel/2008-June/014444.html > > > -- > Henning Westerholt - Development Consumer Products / DSL Core > 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] Kamailio/Openser - improve the performance with children process
by Daniel Cardoso Alves 29 Oct '08

29 Oct '08

Hi people... Is there a number of children that the Kamailio/Openser for improve the performance? Each process takes care of a request? Thanks in advance. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by luzango mfupe 29 Oct '08

29 Oct '08

Hi Ricky,Where is your Kamailio config?? is this your full ngrep capture?? Rgds, Luzango. > > Message: 2 > Date: Tue, 28 Oct 2008 23:10:12 -0700 (PDT) > From: Ricky Gutierrez <xserverlinux(a)yahoo.com> > Subject: [Kamailio-Users] I don't have asterisk audio to openser - > mediaproxy > To: users(a)lists.kamailio.org > Message-ID: <654712.75990.qm(a)web59903.mail.ac4.yahoo.com> > Content-Type: text/plain; charset="us-ascii" > > Hi list is making tests with openser 1.3.2 and mediaproxy to solve the nat, > I have gotten myself an ip it public with my supplier, I have two network > cards in the pc that I am using for openser and mediaproxy together with > asterisk, making tests with mediaproxy 1.9.1 when I receive a call from the > pstn through asterisk I don't have audio, if I call to the pstn they listen > to me well . > > > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1> > >;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2 > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: path > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7492 ACK > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Length: 0 > > > # > U +0.022110 192.168.10.30:5060 -> 192.168.10.1:5060 > INVITE sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: replaces, timer, path > Proxy-Authorization: Digest username="112", realm="192.168.10.1", > algorithm=MD5, uri="sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>", > nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e", > response="c40a9387fdf5de29115c1edadc7f79db" > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Type: application/sdp > Content-Length: 358 > > v=0 > o=112 8000 8001 IN IP4 192.168.10.30 > s=SIP Call > c=IN IP4 192.168.10.30 > t=0 0 > m=audio 5004 RTP/AVP 0 18 3 97 2 9 101 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/16000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > # > U +0.003938 192.168.10.1:5060 -> 192.168.10.30:5060 > SIP/2.0 100 Giving a try > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;branch=z9hG4bKf428b928c25dad03;rport=5060 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > Server: OpenSER (1.3.2-notls (i386/linux)) > Content-Length: 0 > > > # > U +0.000115 192.168.10.1:5060 -> 192.168.10.1:5070 > INVITE sip:2685249@192.168.10.1:5070 SIP/2.0 > Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> > Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0 > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;rport=5060;branch=z9hG4bKf428b928c25dad03 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: replaces, timer, path > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 69 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Type: application/sdp > Content-Length: 358 > P-hint: route(3)|setflag7,forcerport,fix_contact > P-hint: inbound->inbound > P-hint: Route[6]: mediaproxy > > v=0 > o=112 8000 8001 IN IP4 192.168.10.30 > s=SIP Call > c=IN IP4 192.168.1.64 > t=0 0 > m=audio 35040 RTP/AVP 0 18 3 97 2 9 101 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/16000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > # > U +0.000471 192.168.10.1:5070 -> 192.168.10.1:5060 > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0;received= > 192.168.10.1 > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;rport=5060;branch=z9hG4bKf428b928c25dad03 > Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:2685249@192.168.10.1:5070> > Content-Length: 0 > > > I don't have a lot of experience with mediaproxy, and I have some doubts > that such you see they can help me to clarify, inside the file > mediaproxy.ini some options appear which I have configured them but I am not > sure if it is the best way. > > my scenario is the following one: > > <-> UAC<-> NAT <-> ADSL <-> Internet <-> > eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan ( > 192.168.11.1) <-> UAC > > > [MediaProxy] > > start = yes > socket = /var/run/mediaproxy.sock > group = openser > listen = None > allow = None > proxyIP = x.x.x.x (public ip) > ;portRange = 60000:65000 > portRange = 35000:65000 > TOS = 0xb8 > idleTimeout = 60 > holdTimeout = 3600 > forceClose = 0 > > [Accounting] > ; one of none, radius or database > accounting = none > > [Database] > user = dbuser > password = dbpass > host = dbhost > database = radius > table = radacct > > [Radius] > secret = secret > server = localhost > authport = 1812 > acctport = 1813 > dictionaries = /etc/radiusclient-ng/dictionary, > /etc/openser/radius/dictionary, /usr/share/mediaproxy/dictionary > retries = 2 > timeout = 3 > > > > > this couple of you line inside the openser, I don't still understand them > according to the guide of ser getting started they are for asymmetric > clients, but I don't find an example > > modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") > modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") > > somebody that can give me a good help... > > regards > > rickygm > > > > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] Fwd: Re: How to enable local_route feature ?
by Henning Westerholt 29 Oct '08

29 Oct '08

On Wednesday 29 October 2008, Pascal Maugeri wrote: (you can reach the kamailio list at lists.kamailio.org, the lists at openser.org goes to the opensips lists) > Following your instructions I have enable the compilation defined in > Makefile.defs and recompile everything after a make clean. One capture of > the compilation for you to see the define is enable: > > [...] > gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall > -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott > -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK > [..] > checks.c -o checks.o > [...] > > When I try to start the proxy it complains about: > > CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: > syntax error > CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: > invalid local_route statement > > As you announced this feature for OpenSER 1.4 release series, I assume this > is available for kamailio v1.4.2 ? Hi Pascal, i just tried it with the trunk (which should show the same behaviour with regards to this feature). My compilation accepts local_route { [..] } in the configuration. Can you quote the relevant part of your configuration? Cheers, Henning

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[Serusers] Error logs for SER
by Shruti Sanadhya 29 Oct '08

29 Oct '08

Hi, I am trying to setup a SER on Ubuntu 8.04 (Hardy heron). I have been following the starter's guide for configuring my SER but I do not know where to look for the error logs. I just get a 'bad config file' error. Where can I get more details? Thanks

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[Kamailio-Users] error when initilazing "uac.so"
by Alex R.S.M 28 Oct '08

28 Oct '08

Hi, I added "uac.so" module to be able to use "uac_replace_from" but got the following error. I tried to initialized the module params I could think of. I am wondering what I am missing . Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: INFO:avpops:avpops_init: initializing... Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: INFO:uac:mod_init: initializing... Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:uac:mod_init: 'append_fromtag' RR param is not enabled! - required by AUTO restore mode Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:core:init_mod: failed to initialize module uac Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:core:main: error while initializing modules Thanks, Alex _________________________________________________________________ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 28 Oct '08

28 Oct '08

Thanks I'm now trying Zoiper, and it works perfectly, I have solved lot of problems that I thought that were because of my Proxy configuration, but they were because of my X-Lite. Now I will try Twinkle. Thanks for your comments. David Cespedes ----- Mensaje original ----- De: Victor Pascual Ávila <victor.pascual.avila(a)gmail.com> Fecha: Sábado, Octubre 25, 2008 12:51 pm Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > On Sat, Oct 25, 2008 at 12:09 PM, Iñaki Baz Castillo > <ibc(a)aliax.net> wrote: > > El Sábado, 25 de Octubre de 2008, ingdavidcespedes(a)cable.net.co > escribió:> > >> since we don't have an upgrade of X-lite for linux > >> systems, could you recommend me another Softphone. > > > > Always Twinkle softphone, the best one ;) > > I agree-- Twinkle is available for Linux only. > -- > Victor Pascual Ávila > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] My contrib to the project - Simple UI
by Graham Wooden 28 Oct '08

28 Oct '08

Hi list, After seeing that CDRTool was a little too powerful (and complicated for that matter) I decided to take it's CDR gathering and add a "simple" web UI to it allowing me to maintain my SIP users and various settings (some aren't shown in this version right now). Let me preface this - this is very plain, meaning it's not stylized. No CSS, no fancy layouts, etc. Mainly because I have this within my existing UI. However, I stripped it out and is functional as described. Again, for the CDR stuff it does require to have the RADIUS accounting and MySQL setup from CDRTool. Link: https://www.leasedminds.com/ui/ If people see value with this, I will be more than happy to work with someone to make it "stylized" and more featured pack. Enjoy and thanks, -graham

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[Kamailio-Users] Proxying media on reinvite
by CSB 28 Oct '08

28 Oct '08

We have the following scenario: Nated UAC receives INVITE Call is established, we proxy audio Far end sends Re-INVITE (to inform UAC to use UDPTL) We don't proxy audio so call fails So I need to understand how the appropriate way to force us to proxy the audio when the far end of a reinvite is NATed. Relevant parts of openser.cfg (OpenSER 1.3.2): We're using mediaproxy. modparam("usrloc", "nat_bflag", 6) route { # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #If it's an INVITE & client is NATed, if (method=="INVITE" && client_nat_test("3")) { #Record-route and specify the record-route header explicitly record_route_preset("64.38.1.1:5060;nat=yes"); # insert IP address #xlog("invite and nated so record_route_preset"); #If not Nated and not REGISTER then normal record-route } else if (method!="REGISTER") { record_route(); xlog("not register and not nated so record_route"); }; #if (method!="REGISTER") { # record_route(); #}; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { #Ensure we are dealing with a re-INVITE. Only connected calls have tag= #entry in the To header. So if it's loose routed and doesn't have totag #and it's an invite or reply then something's wrong xlog("loose_route"); if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); xlog("403 in lr"); return; }; if (method=="INVITE") { if (!allow_trusted()) { xlog("!allow_ trusted for r-uri <$ru>"); if (!proxy_authorize("","subscriber")) { proxy_challenge("domain.com","1"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; #check client is nated or that we've already identified it's nated if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setbflag(6); xlog("Recipient is nated so setbflag 6 and use mediaproxy: r-uri <$ru>"); use_media_proxy(); }; }; route(1); return; }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- #Call reply_route(1) to intercept response messages heading to the client t_on_reply("1"); #Try to relay the message to its destination if (!t_relay()) { #If it can't and it's an INVITE or ACK end the media proxying log(1,"route[1] !t_relay"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); } else { log(1,"route[1] t_relay"); }; } onreply_route[1] #Handles message that are returned to the sender i.e. response to caller's original request { xlog("hit onreply_route(1). rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); if (isbflagset(6)) { xlog("flag(6) is set for $fu . We're currently in onreply_route(1)"); }; if (isbflagset(7)) { xlog("flag(7) is set for $fu . We're currently in onreply_route(1)"); }; if (status=~"(180)|(183)|2[0-9][0-9]") { xlog("status matches for $fu . We're currently in onreply_route(1)"); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #Check the SDP payload length. Assume if we have something we can call mediaproxy if (!search("^Content-Length:[ ]*0")) { log(1,"using media proxy in onreply_route(1)"); use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } The re-invite looks like this: INVITE sip:6422334455@58.28.1.1:5061 SIP/2.0 Via: SIP/2.0/UDP 202.180.1.1:5060;branch=z9hG4bK251fed3b;rport Route: <sip:147.202.1.1;lr=on;ftag=as45a8ade4> From: "user" <sip:022998877@202.180.1.1>;tag=as45a8ade4 To: <sip:022334455@147.202.1.1>;tag=4c813866c894f124i1 Contact: <sip:022998877@202.180.1.1> Call-ID: 0818dfa10f1da0f25daa4759332d3c55(a)202.180.1.1 CSeq: 103 INVITE User-Agent: UAS Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 349 v=0 o=root 5466 5467 IN IP4 202.180.1.1 s=session c=IN IP4 202.180.1.1 t=0 0 m=image 4724 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy Our OK looks like this: SIP/2.0 200 OK To: <sip:022334455@147.202.1.1>;tag=4c813866c894f124i1 From: "user" <sip:022998877@202.180.1.1>;tag=as45a8ade4 Call-ID: 0818dfa10f1da0f25daa4759332d3c55(a)202.180.1.1 CSeq: 103 INVITE Via: SIP/2.0/UDP 202.180.1.1:5060;branch=z9hG4bK251fed3b;rport=5060 Record-Route: <sip:147.202.1.1;lr=on;ftag=as45a8ade4> Contact: 6422334455 <sip:6422334455@58.28.1.1:5061> Server: Linksys/SPA2102-5.1.9 Content-Length: 265 Content-Type: application/sdp v=0 o=- 69685 69685 IN IP4 192.168.1.113 s=- c=IN IP4 192.168.1.113 t=0 0 m=image 16386 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy Unfortunately this code doesn't result in the c record of the OK being updated to the server IP (it is still the private IP of the UAC). In the loose route section, the re-invite doesn't pass (client_nat_test("3") || search("^Route:.*;nat=yes")) so we don't use mediaproxy. In the on-reply route bflag(6) is not set. I don't understand why bflag(6) is not set since the r-uri is to a registered user that has bflag(6) set (at least that's how I understand it). Could anyone point me in the right direction? Thanks Cameron

2 1

[Kamailio-Users] [presence] Modifying NOTIFY headers set
by Pascal Maugeri 27 Oct '08

27 Oct '08

Hi I need to add a SIP header to outgoing NOTIFY messages, just before it is sent to remote users. Is that possible to do that without using the local_route section (using openser 1.3) ? Regards, Pascal

4 7

[Kamailio-Users] [dispatcher] configure the sending of OPTIONS pings
by Pascal Maugeri 27 Oct '08

27 Oct '08

Hi How do you configure dispatcher to send OPTIONS ping to load-balanced proxies ? I'm using kamailio 1.4.2 with this of configuration: [...] modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_ping_from", "sip:fend1@10.132.184.33<sip%3Afend1(a)10.132.184.33> ") modparam("dispatcher", "ds_probing_threshhold", 3) [...] The load-balancing works properly : I see the traffic being sent to my proxies. But I don't see any SIP OPTIONS packets. I've read in module doc "This method is only available, if compiled with the probing of failed gateways enabled. " but I can't find how to activate this on compilation time, so I'm not sure if I have this feature or not enable. Regards, Pascal

3 4

[Kamailio-Users] i want to stop second registration request if user already registered
by Mian M Asif 27 Oct '08

27 Oct '08

Hi All, i want to stop second registration request if use already registered i am using below code but getting no any response. please help me how to solve this problem. if (method=="REGISTER") { if (registered("location") ) { xlog("L_NOTICE","Asif: lookup(location) Passed\n"); sl_send_reply("699", "User Not Found"); return; } route(2); return; route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- xlog("L_NOTICE","Danish: Entering route[2]\n"); # if (registered("location")) { # xlog("L_NOTICE","Asif: lookup(location) failed\n"); # sl_send_reply("404", "User Not Found"); # return; # }; xlog("L_NOTICE","Asif: Register message IPsrcaddr [$si], RecvdIPaddr[$Ri]\n"); if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { xlog("L_NOTICE","Danish: In route[2] Found Nated contact\n"); setflag(6); setbflag(6); # if you want OPTIONS natpings uncomment next setbflag(7); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!radius_www_authorize("abc.com")) { www_challenge("abc.com", "1"); }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } } regards, Asif

4 5

[Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 25 Oct '08

25 Oct '08

Hi I'm using Xlite 2.0 release 1105d, and when I try to REGISTER to IP:5070, it doesn't let me. In another Windows computer I have Xlite 3.0 and there you can user an IP:Port Address to register to. Does anyone know If it is a problem from version 2.0 or I am doing something wrong. If it is a problem of 2.0 version, and since we don't have an upgrade of X-lite for linux systems, could you recommend me another Softphone. Thanks David Cespedes

3 2

[Kamailio-Users] Stateless mode for sip load balancer
by Kaeman Chris 25 Oct '08

25 Oct '08

Hello, I have opensip working as a sip load balancer which balances load between 8 asterisk servers running 4 instances on each. The opensip task is just to load balance the incoming requests for 32 asterisk instances on 8 boxes. Can i run it in the stateless mode? Anything that i should be aware about which will prohibit me to run it on the stateless mode. Thanks, KChris Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

3 2

Re: [Kamailio-Users] kamailio with tls
by ingdavidcespedes＠cable.net.co 24 Oct '08

24 Oct '08

I knowed that you can have TLS connection between UA and other Proxy Servers, but I didn't know that Kamailio supports it, thanks for your comment. So maybe Paulos's problem is that his Client is not trying to REGISTER to the correct port (5061) of his Kamailio. ----- Mensaje original ----- De: Klaus Darilion <klaus.mailinglists(a)pernau.at> Fecha: Viernes, Octubre 24, 2008 12:45 pm Asunto: Re: [Kamailio-Users] kamailio with tls > (cc'ed to the mailinglist) > > ingdavidcespedes(a)cable.net.co schrieb: > > Enable TLS in Kamailio is only for connection between other Kamailio > > (SIP Proxy Servers), it doesn't have anything to do with UA, or > am I > > wrong? . As I understand, you can not implement TLS between UA's and > > Kamailio. > > Of course you can. If the SIP client supports TLS you can also use > TLS > between UA and Kamailio - no problem. > > I use it with eyebeam clients and SNOM phones. Also pjsip supports > TLS, > for example you can test it using QjSimple. > http://www.ipcom.at/index.php?id=560 > > regards > klaus > > > > > > ----- Mensaje original ----- De: Klaus Darilion > > <klaus.mailinglists(a)pernau.at> Fecha: Viernes, Octubre 24, 2008 > 11:04> am Asunto: Re: [Kamailio-Users] kamailio with tls > > > >> > >> paulo leonardo schrieb: > >>> Hi, > >>> > >>> I would like a litle help :D! > >>> > >>> I installed kamailio and everything is ok :D! But i want to use > >> TLS, but > >>> when i setup TLS in kamailio, don't work the REGISTER (i can't > >> register > >>> my sofphone ...). And i compliled kamilio with TLS!!! when i > >> comment the > >>> cofigurantion TLS and use port 5060 works! > >> > >> So, "what" does not work? > >> > >> - Does Kamailio start (ps -Alf|grep kamailio)? - Does K listen to > >> the specified TLS socket (netstat -anp|grep kama)? - which client > >> do you use? - is a TCP connection set up? - is a TLS handshake > >> happening (ssldump)? - what is in the logfile of Kamailio? ..... > >> > >> Your error description is to short! > >> > >> regards klaus > >> > >> > >> > >> > >>> below is my configuration > >>> > >>> ----------------------------------------- disable_tls = no listen > >>> = tls:192.168.170.101:5061 <http://192.168.170.101:5061> > >>> tls_verify_server = 1 tls_verify_client = 1 > >>> tls_require_client_certificate = 1 tls_method = TLSv1 > >>> tls_certificate = > >>> "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key > >>> = "/usr/local/etc/kamailio/tls/user/user- > >> privkey.pem"> tls_ca_list > >>> --------------------------------------- > >>> > >>> if (!www_authorize("192.168.170.101 <http://192.168.170.101>", > >>> "subscriber")) { www_challenge("192.168.170.101 > >> <http://192.168.170.101>", "0"); > >>> exit; } > >>> > >>> ---------------------------------------- > >>> > >>> root@pst:/usr/local/etc/kamailio# ls -R tls/ tls/: ca.conf > >>> README request.conf rootCA user user.conf > >>> > >>> tls/rootCA: cacert.pem certs index.txt private serial > >>> > >>> tls/rootCA/certs: > >>> > >>> tls/rootCA/private: cakey.pem > >>> > >>> tls/user: user-calist.pem user-cert.pem user-cert_req.pem > >>> user-privkey.pem > >>> > >>> > >>> thanks!!! > >>> > >>> > >>> > >>> ---------------------------------------------------------------- > -- > >>> > >> ------ > >>> _______________________________________________ Users mailing > >>> list Users(a)lists.kamailio.org > >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> _______________________________________________ Users mailing > list > >> Users(a)lists.kamailio.org > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

1 0

[Kamailio-Users] Problems with CDRTool installation!
by Nuno Marques 24 Oct '08

24 Oct '08

Hi, I'm trying to follow the INSTALL.txt of the CDRTool to accomplish a nice installation of OpenSER 1.3.2 + Mediaproxy + SerMyAdmin + Asterisk + CDRTool + FreeRadius. All went well until i started CDRTool installation... it's a little confusing for a newbie like me but... all the steps of the installation seem to went well. When i go to the browser to look for the CDRTool portal it gives me this error: Database error: pconnect as cdradmin to sipdb failed. Session halted. I thought it was a mysql users problem... so i went to the shell and tried to enter mysql with login and password for cdradmin, and... No problem. It worked fine. Mysql accepted the user login/password. Is there someone who can tell me where can i look for errors... I'm really stuck on this part... Thanks in advance, Nuno

2 9

[Kamailio-Users] draft sip-outbound
by paulo leonardo 24 Oct '08

24 Oct '08

Hi, I would like to know if kamailio implements draft sip-outbound? Can i use draft sip-outbound? http://tools.ietf.org/html/draft-ietf-sip-outbound-15 thanks!

2 1

Re: [Kamailio-Users] kamailio with tls
by Klaus Darilion 24 Oct '08

24 Oct '08

(cc'ed to the mailinglist) ingdavidcespedes(a)cable.net.co schrieb: > Enable TLS in Kamailio is only for connection between other Kamailio > (SIP Proxy Servers), it doesn't have anything to do with UA, or am I > wrong? . As I understand, you can not implement TLS between UA's and > Kamailio. Of course you can. If the SIP client supports TLS you can also use TLS between UA and Kamailio - no problem. I use it with eyebeam clients and SNOM phones. Also pjsip supports TLS, for example you can test it using QjSimple. http://www.ipcom.at/index.php?id=560 regards klaus > > ----- Mensaje original ----- De: Klaus Darilion > <klaus.mailinglists(a)pernau.at> Fecha: Viernes, Octubre 24, 2008 11:04 > am Asunto: Re: [Kamailio-Users] kamailio with tls > >> >> paulo leonardo schrieb: >>> Hi, >>> >>> I would like a litle help :D! >>> >>> I installed kamailio and everything is ok :D! But i want to use >> TLS, but >>> when i setup TLS in kamailio, don't work the REGISTER (i can't >> register >>> my sofphone ...). And i compliled kamilio with TLS!!! when i >> comment the >>> cofigurantion TLS and use port 5060 works! >> >> So, "what" does not work? >> >> - Does Kamailio start (ps -Alf|grep kamailio)? - Does K listen to >> the specified TLS socket (netstat -anp|grep kama)? - which client >> do you use? - is a TCP connection set up? - is a TLS handshake >> happening (ssldump)? - what is in the logfile of Kamailio? ..... >> >> Your error description is to short! >> >> regards klaus >> >> >> >> >>> below is my configuration >>> >>> ----------------------------------------- disable_tls = no listen >>> = tls:192.168.170.101:5061 <http://192.168.170.101:5061> >>> tls_verify_server = 1 tls_verify_client = 1 >>> tls_require_client_certificate = 1 tls_method = TLSv1 >>> tls_certificate = >>> "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key >>> = "/usr/local/etc/kamailio/tls/user/user- >> privkey.pem"> tls_ca_list >>> --------------------------------------- >>> >>> if (!www_authorize("192.168.170.101 <http://192.168.170.101>", >>> "subscriber")) { www_challenge("192.168.170.101 >> <http://192.168.170.101>", "0"); >>> exit; } >>> >>> ---------------------------------------- >>> >>> root@pst:/usr/local/etc/kamailio# ls -R tls/ tls/: ca.conf >>> README request.conf rootCA user user.conf >>> >>> tls/rootCA: cacert.pem certs index.txt private serial >>> >>> tls/rootCA/certs: >>> >>> tls/rootCA/private: cakey.pem >>> >>> tls/user: user-calist.pem user-cert.pem user-cert_req.pem >>> user-privkey.pem >>> >>> >>> thanks!!! >>> >>> >>> >>> ------------------------------------------------------------------ >>> >> ------ >>> _______________________________________________ Users mailing >>> list Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >

1 0

[Kamailio-Users] kamailio with tls
by paulo leonardo 24 Oct '08

24 Oct '08

Hi, I would like a litle help :D! I installed kamailio and everything is ok :D! But i want to use TLS, but when i setup TLS in kamailio, don't work the REGISTER (i can't register my sofphone ...). And i compliled kamilio with TLS!!! when i comment the cofigurantion TLS and use port 5060 works! below is my configuration ----------------------------------------- disable_tls = no listen = tls:192.168.170.101:5061 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" tls_ca_list --------------------------------------- if (!www_authorize("192.168.170.101", "subscriber")) { www_challenge("192.168.170.101", "0"); exit; } ---------------------------------------- root@pst:/usr/local/etc/kamailio# ls -R tls/ tls/: ca.conf README request.conf rootCA user user.conf tls/rootCA: cacert.pem certs index.txt private serial tls/rootCA/certs: tls/rootCA/private: cakey.pem tls/user: user-calist.pem user-cert.pem user-cert_req.pem user-privkey.pem thanks!!!

2 1

[Serusers] Asterisk -> SER one way audio call issue
by Thorsten 24 Oct '08

24 Oct '08

Hi guys does someone have experience about the interaction of asterisk and SER. I have a strange behavior. I terminate a call from an Asterisk to SER and then to a interconnection partner which runs also on asterisk. Whenever I make a call after a long (3-5 mins) idle time I got a one way audio call because the interconnection partner sends it back to SER instead of asterisk. But the call after works like a charm. And this happens all the time. This is my SER config # ----------- global configuration parameters ------------------------ #debug=4 debug=2 fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=82.98.89.140:5060 port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/ser/current/lib/ser/modules/sl.so" loadmodule "/usr/local/ser/current/lib/ser/modules/exec.so" loadmodule "/usr/local/ser/current/lib/ser/modules/tm.so" loadmodule "/usr/local/ser/current/lib/ser/modules/rr.so" loadmodule "/usr/local/ser/current/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser/current/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser/current/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser/current/lib/ser/modules/textops.so" loadmodule "/usr/local/ser/current/lib/ser/modules/avpops.so" #loadmodule "/usr/local/ser/current/lib/ser/modules/group.so" loadmodule "/usr/local/ser/current/lib/ser/modules/xlog.so" loadmodule "/usr/local/ser/current/lib/ser/modules/auth.so" #loadmodule "/usr/local/ser/current/lib/ser/modules/nathelper.so" loadmodule "/usr/local/ser/current/lib/ser/modules/uri.so" #loadmodule "/usr/local/ser/current/lib/ser/modules/uri_db.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="ACK") { route(1); break; } if (method=="REGISTER") { save("location"); break; }; # only forward if from our nat ip if (method=="INVITE" && (src_ip==82.98.89.146 || src_ip==82.98.89.147 || src_ip==82.98.89.129)) { #record_route(); if (uri =~ "sip:[0-9]*@*") { #record_route(); route(3); break; } } else { sl_send_reply("401", "Unauthorized"); }; if (method=="CANCEL") { route(1); break; }; if (method=="BYE") { route(1); break; } lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[3] { # strip the first 4 characters 1991 #strip(4); #rewritehost("62.134.16.104"); rewritehostport("62.134.16.104:5060"); #t_relay_to_udp("62.134.16.104", "5060"); xlog("L_WARN", "[%Tf]: Call forwarded to BT from: [%fu] to: [%tu] next: [%ds]\r\n"); route(1); break; } The asterisk dialplan looks like this ; route everything to SER exten => _X.,1,Noop(${SIP_HEADER(FROM)}) exten => _X.,n,Set(SIP_CODEC="gsm") exten => _X.,n,Dial(sip/${EXTEN}(a)82.98.89.140) exten => _X.,n,Hangup() The asterisk server has the public IP 82.98.89.146 and SER 82.98.89.140. BT which is our interconnection partner has the IP 62.134.16.104. Could someone help me with this? Thanks Ciao Thorsten Kohlhepp

1 0

[Serusers] registrar: interesting, however rare, bug..
by Alexandre Snarskii 24 Oct '08

24 Oct '08

Hi! This week we saw interesing problem with one of our clients: initially he was not able to configure his CCME to register on our proxy, so we just added static location serctl ul add ..., that's our classic workaround for CCME and another not-so-smart admins. But when he finally configured registration - things gone nuts: clients asks for registration: Date: Fri, 24 Oct 2008 13:16:29 GMT Timestamp: 1224854189 CSeq: 2905600 REGISTER Contact: <sip:NNNNNNN@AA.BBB.CCC.DD:5060> Expires: 3600 Authorization: Digest .... and ser replies with 200 OK, but with Contact: field filled with his static registration: Contact: <sip:NNNNNNN@AA.BBB.CCC.DD>;q=1;expires=4293498731;received="" As you can see, expires value is 4293498731, which means that it will be valid for about 136 years, just sixteen days short than 2^32 seconds... So, by RFC, client's router should wait for ~100 years before attempting registration refresh :) But CCME, i suppose, thinks different, it just adds received expires value to current time, got uint32 overflow and result is now()-16 days. As a result CCME thinks that registration is always expired and tries to re-register immediately... And does so hundredths times per second, creating unnecessary extra load to our proxy and database backend.... Why that is so: modules/registrar/reply.c calculates expires= as int2str((int)(c->expires - act_time), &len) but, c->expires can be less than act_time, so, result will be negative integer (translated to unsigned int on call to int2str, and this unsigned int will have higher bit set on, so it will be big one). Well, there are check if(VALID_CONTACT(c, act_time)) which filters out most of the cases where с->expires <= act_time, but for 'static' registrations these checks are off: ((c->expires > act_time) || (c->flags & FL_PERMANENT)) is always true in case of 'static' contact because of FL_PERMANENT. For now I fixed it with following patch (not perfect, i suppose, it just uses default_expires value for negative expires time, but it was easy to write and it works for me): --- reply.c.orig Thu Jan 13 11:48:08 2005 +++ reply.c Fri Oct 24 14:08:13 2008 @@ -163,7 +163,11 @@ memcpy(p, EXPIRES_PARAM, EXPIRES_PARAM_LEN); p += EXPIRES_PARAM_LEN; - cp = int2str((int)(c->expires - act_time), &len); + if (c->expires > act_time) { + cp = int2str((int)(c->expires - act_time), &len); + } else { + cp = int2str(default_expires, &len); + } memcpy(p, cp, len); p += len; Environment: ser-0.9.6/i386/FreeBSD (ser-2.0 has the same code, so it may be affected too). The only tuning of usrloc and registrar modules is db_url and dbmode = 2.

1 0

Re: [Kamailio-Users] Users Digest, Vol 41, Issue 68
by Mian M Asif 24 Oct '08

24 Oct '08

Dear Abalashov, What do you mean by "no[t] getting any response? is mean this function is not working please let me know i am doing right or wrong, dear Arif, exit also is not working cheers :). regards, Asif > Today's Topics: > > 1. Re: Architecture with load-balancing (dispatcher module) > (Ovidiu Sas) > 2. i want to stop second registration request if user already > registered (Mian M Asif) > 3. Re: i want to stop second registration request if user > already registered (Alex Balashov) > 4. Re: i want to stop second registration request if user > already registered (arif.zaman) > 5. Re: Anyone doing CNAM lookups? (Klaus Darilion) > 6. Kamailio v1.4.2 Released (Daniel-Constantin Mierla) > > > Message: 2 > Date: Thu, 23 Oct 2008 19:49:21 +0500 > From: "Mian M Asif" <asif44pk(a)gmail.com> > Subject: [Kamailio-Users] i want to stop second registration request > if user already registered > To: users(a)lists.kamailio.org > Message-ID: > <b005c33b0810230749v1b00bf9auabb97cfc17e517c6(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi All, > i want to stop second registration request if use already registered i > am using below code but getting no any response. please help me how to > solve this problem. > > if (method=="REGISTER") { > if (registered("location") ) { > xlog("L_NOTICE","Asif: lookup(location) Passed\n"); > sl_send_reply("699", "User Not Found"); > return; > } > route(2); > return; > > route[2] { > > # ----------------------------------------------------------------- > # REGISTER Message Handler > # ---------------------------------------------------------------- > xlog("L_NOTICE","Danish: Entering route[2]\n"); > # if (registered("location")) { > # xlog("L_NOTICE","Asif: lookup(location) failed\n"); > # sl_send_reply("404", "User Not Found"); > # return; > # }; > > xlog("L_NOTICE","Asif: Register message IPsrcaddr [$si], > RecvdIPaddr[$Ri]\n"); > if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { > xlog("L_NOTICE","Danish: In route[2] Found Nated contact\n"); > setflag(6); > setbflag(6); > # if you want OPTIONS natpings uncomment next > setbflag(7); > fix_nated_register(); > force_rport(); > }; > > sl_send_reply("100", "Trying"); > > if (!radius_www_authorize("abc.com")) { > www_challenge("abc.com", "1"); > }; > consume_credentials(); > > if (!save("location")) { > sl_reply_error(); > }; > > } > } > > regards, > Asif > > > > ------------------------------ > > Message: 3 > Date: Thu, 23 Oct 2008 10:51:30 -0400 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [Kamailio-Users] i want to stop second registration > request if user already registered > To: Mian M Asif <asif44pk(a)gmail.com> > Cc: users(a)lists.kamailio.org > Message-ID: <49008F72.1060700(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > What do you mean by "no[t] getting any response?" > > Mian M Asif wrote: > >> Hi All, >> i want to stop second registration request if use already registered i >> am using below code but getting no any response. please help me how to >> solve this problem. >> >> if (method=="REGISTER") { >> if (registered("location") ) { >> xlog("L_NOTICE","Asif: lookup(location) Passed\n"); >> sl_send_reply("699", "User Not Found"); >> return; >> } >> route(2); >> return; >> >> route[2] { >> >> # ----------------------------------------------------------------- >> # REGISTER Message Handler >> # ---------------------------------------------------------------- >> xlog("L_NOTICE","Danish: Entering route[2]\n"); >> # if (registered("location")) { >> # xlog("L_NOTICE","Asif: lookup(location) failed\n"); >> # sl_send_reply("404", "User Not Found"); >> # return; >> # }; >> >> xlog("L_NOTICE","Asif: Register message IPsrcaddr [$si], >> RecvdIPaddr[$Ri]\n"); >> if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { >> xlog("L_NOTICE","Danish: In route[2] Found Nated contact\n"); >> setflag(6); >> setbflag(6); >> # if you want OPTIONS natpings uncomment next >> setbflag(7); >> fix_nated_register(); >> force_rport(); >> }; >> >> sl_send_reply("100", "Trying"); >> >> if (!radius_www_authorize("abc.com")) { >> www_challenge("abc.com", "1"); >> }; >> consume_credentials(); >> >> if (!save("location")) { >> sl_reply_error(); >> }; >> >> } >> } >> >> regards, >> Asif >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > Message: 4 > Date: Thu, 23 Oct 2008 07:59:45 -0700 (PDT) > From: "arif.zaman" <arif.zaman(a)brotecs.com> > Subject: Re: [Kamailio-Users] i want to stop second registration > request if user already registered > To: users(a)lists.kamailio.org > Message-ID: <20132764.post(a)talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > > Mian M Asif wrote: >> >> if (method=="REGISTER") { >> if (registered("location") ) { >> xlog("L_NOTICE","Asif: lookup(location) Passed\n"); >> sl_send_reply("699", "User Not Found"); >> return; >> } >> > > Use "exit;" instead of "return;". > > Cheers, > ARIF >

1 0

[Kamailio-Users] Anyone doing CNAM lookups?
by Brian McCrary 24 Oct '08

24 Oct '08

I've been using OpenSER 1.0.1 for the past couple of years, but I now need to add CNAM lookups. It looks like the presence module might be what I need, although from the looks if it, it's made to respond to lookups, instead of actually generate a query to another server for a lookup. I'm looking at doing CNAM lookups with Verisign which look to interface with a SUBSCRIBE message, in which in return Verising sends back a NOTIFY message with one of the fields being "calling-name" which has the CNAM info in it. Has anyone been succesful in implementing such a lookup with Verisign or another CNAM database provider? Brian

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[Serusers] SER without any fork(), It is possible?
by Samuel Ayet San Andres 24 Oct '08

24 Oct '08

Hi all, I´m trying to use SER (SIP Express Router) in a linux enbedded system with uClinux. In my enbedded processor i dont have any MMU so it cant fork(), it uses vfork() that is a cousin of fork but diferent because qhen father vfork a child, stops until the child make execve() or exit. When I run SER on my embedded plataform in a debug mode it shows me: ERROR: main_loop can´t fork(); Then I try to chage all forks for vfork() but it still not working due to the childs never make execve(). And after the explain of the context, here is the question: can I run SER without any child? Can I run SER without making any fork? I mean that only ONE process (witout deamonize) will run alone. I don´t need to manage more than 1 call...I only want to have one process listening to ONE IP. I have try to change the configuration of the ser 'ser.cfg' asi show: fork=no childrens=0 But still without working and i´m not sure if this is the right way to run SER without forking... Thank you for you support. Samuel. --

1 0

[Kamailio-Users] presence network Agent
by suresh tattikota 24 Oct '08

24 Oct '08

Hi all I am working on Presence network Agent. please let me know any one is working on the same . thank you all suresh

2 1

Re: [Kamailio-Users] Record Route and proxy behind static NAT
by Zappasodi Daniele 23 Oct '08

23 Oct '08

Thanks for your reply. I'm agree with your observations. I had already tried to solve the situations as described in the draft (avoiding spiral) and I have built a configuration that seem to work, at least for the basic call, but my solution needs to much tricks and I'm not sure that what I have done is safe and if it works in more complex call flows. I set the IP address in Record Route depending on position public\private of the callee and I substitute the IP address in Record Route for each reply, when it needs. First of all I suppose to know if the caller and callee is private or public (I check if the Request URI domain with RFC 1918) and I don't know if this is right. Then there is the problem to write the record route. I have moved the record route section in a branch route, because here I know the final destination for each branch. Here there is another doubt: I rewrote the record_route_preset function in order to remove the limit of one record route because in parallel forking I set more than record for each transaction, one record route for each branch. What are the consequences? I risk "only" to loose a call or to kill the proxy? I would prefer to leave the record route section at the beginning, but is it possible to substitute the record route after it is added? I remember that is possible to change only the original message and that there are some limitations on the possibility to change an header added during the message processing (I had some segmentation faults when I tryed with a custom header). Just now I'm looking on what to do with a relay from the failure route to a second destination. Before I proceding on this way I would like to know if I'm going on the wrong direction and if someone has found a simpler solution with this kind of scenarios. Regards, Daniele > Date: Wed, 22 Oct 2008 19:02:35 +0200 > From: " I?aki Baz Castillo " <ibc(a)aliax.net> > Subject: Re: [Kamailio-Users] Record Route and proxy behind static NAT > Cc: users(a)lists.kamailio.org > Message-ID: > <cc1f582e0810221002i7c95570epf585456a248811d1(a)mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > The solution could be adding two Record-Routes, one with private IP > and the other with public IP, but I'm not sure of how to achieve it > with Kamailio. AFAIK Kamailio *already* does it when the transaction > comes via eth0 and leaves the proxy via eth1, but this is not your > case. > I recommend you reading this draft, and later trying to get it in Kamailio: > http://tools.ietf.org/html/draft-ietf-sip-record-route-fix > (5. Double Record-Routing) ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 1

[Kamailio-Users] Kamailio v1.4.2 Released
by Daniel-Constantin Mierla 23 Oct '08

23 Oct '08

Hello everybody, a new release in 1.4 series is out. Kamailio 1.4.2 is based on the latest version of branch 1.4, including many fixes in code and documentation, therefore those running 1.4.0 or 1.4.1 are advised to upgrade. This is a minor release that includes updates since v1.4.1 - configuration file and database compatibility is preserved. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.4.2/src/ <http://www.kamailio.org/pub/kamailio/1.4.2/src> Detailed changelog: http://www.kamailio.org/pub/kamailio/1.4.2/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.4.2/ Project site at SourceForge.net: http://sourceforge.net/projects/openser/ Modules' documentation: http://www.kamailio.org/docs/modules/1.4.x/ <http://www.kamailio.org/docs/modules/1.4.x> What is new in 1.4.x release series is summarized in the announcement of v1.4.0: http://www.kamailio.org/mos/view/KAMAILIO-v1.4.0-Released Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Architecture with load-balancing (dispatcher module)
by Pascal Maugeri 23 Oct '08

23 Oct '08

Hi I was wondering how to achieve an architecture with two or more active load-balancer nodes (kamailio+dispatcher module). I have read how to setup two dispatcher nodes, one node as a master and the other one as a backup. The backup node doesn't process any traffic until master fails. But how to make the traffic being processed by both load-balancers ? In the case for instance the capacity of single node is not enough to process all incoming traffic. Is there any recommended configuration (eg. using frontal application-level swith) Regards, Pascal

4 4

Re: [Kamailio-Users] Architecture with load-balancing (dispatcher module)
by Daniel-Constantin Mierla 23 Oct '08

23 Oct '08

On 10/23/08 15:05, Pascal Maugeri wrote: > > On Thu, Oct 23, 2008 at 1:47 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > a solution I am using is to do shared IP (vrrp/ucarp). The typical > scenario is active-backup, when one share IP is used. > > But you can do active-active by using two shared IP addresses, so > will be > > server1: active_ip_1 - backup_ip_2 > server2: backup_ip_1 - active_ip_2 > > So each server is active and backs-up the other. This can be > scaled, with more than 2 servers, you need an IP per server, and > you will have a chain of active servers backing up next one. Based > on who is backing up who, you may have one or more servers down > without affecting the service > > You have to take care of balancing the traffic among the IP addresses. > > > OK but how do you share traffic among the IP addresses ? You need a > kind of super load-balancer that load-balance traffic among > load-balancers ? one way you can do it is via DNS. Daniel > -pascal > > > > > Cheers, > Daniel > > > > On 10/23/08 13:16, Alex Balashov wrote: > > Any "frontal application-level switch" would simply have the > same liability. > > Kamailio is a very high-throughput proxy that can handle huge > amounts of call setups per second. I think you can count on that. > > Failing over around the load balancer node to a secondary load > balancer or distributing the traffic among multiple load > balancers is a job best left to the sending endpoint. For > example, a DID origination provider's switch or SBC can be set > up to fail over calls to a different IP endpoint for your SIP > trunk if no response is received within a certain amount of > time. That is how this is typically done. At some point > you've got to say that you've done all you can do, and it's up > to the other side. > > Pascal Maugeri wrote: > > > > Hi > > I was wondering how to achieve an architecture with two or > more active load-balancer nodes (kamailio+dispatcher module). > > I have read how to setup two dispatcher nodes, one node as > a master and the other one as a backup. > > The backup node doesn't process any traffic until master > fails. But how to make the traffic being processed by both > load-balancers ? In the case for instance the capacity of > single node is not enough to process all incoming traffic. > Is there any recommended configuration (eg. using frontal > application-level swith) > > Regards, > Pascal > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Usrloc: cache vs database inconsistencies
by CSB 23 Oct '08

23 Oct '08

I have noticed a consistent difference between the location records OpenSER has cached and those stored in the database. e.g. openserctl ul show --brief database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump brief Domain:: location table=512 records=19 max_slot=2 Compared to: select count(*) from ser.location; +----------+ | count(*) | +----------+ | 10 | +----------+ 1 row in set (0.00 sec) This difference is consistent over time. When I look at the database logs OpenSER is trying to update an existing record in the database which, for some reason, doesn't exist. Our config has the following relevant parameters modparam("usrloc", "db_mode", 2) Openser 1.3.2 Any thoughts on how to prevent this circumstance from occurring and how to synchronise the cache with the db in an orderly manner? I had thought that after 120 seconds (timer_interval default)this would happen but it doesn't seem to. Any advice appreciated. Regards Cameron

2 1

[Kamailio-Users] Hold - core dump
by Marian Piater 23 Oct '08

23 Oct '08

Hello, I tried Kamailio 1.4.1+TLS with Mediaproxy 2.1.0. Everything works fine, but when I place call to Hold (using Hold button), Kamailio sometimes crashed. When I disable mediaproxy, Kamailio works without crash. I tried using engage_media_proxy() without any end_media_proxy or use_media_proxy and I also tried using use_media_proxy and end_media_proxy without engage_media_proxy with that same result. Sometimes Kamailio crash after first place call to hold, sometimes crash after few place call to hold/unhold. Sometimes work fine. OS is Debian Etch. Kamailio generate core dump. Core was generated by `kamailio'. Program terminated with signal 11, Segmentation fault. #0 0xb7d1e19a in strnlen () from /lib/tls/i686/cmov/libc.so.6 (gdb) bt #0 0xb7d1e19a in strnlen () from /lib/tls/i686/cmov/libc.so.6 #1 0xb7cf13f1 in vfprintf () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7d0fb01 in vsnprintf () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7cf6945 in snprintf () from /lib/tls/i686/cmov/libc.so.6 #4 0xb7888e10 in end_media_session (callid=<value optimized out>, from_tag={s = 0x2 "", len = 4}, to_tag= {s = 0xbf9ce1b8 "remove\r\ncall_id: \r\nfrom_tag: ?I?E?\177\017openser", len = 0}) at mediaproxy.c:1478 #5 0xb7889147 in __dialog_ended (dlg=0x6, type=2048, _params=0xb7883cbc) at mediaproxy.c:1545 #6 0xb7865378 in run_dlg_callbacks (type=2048, dlg=0xb59e6398, msg=0x0, dir=0, dlg_data=0x0) at dlg_cb.c:252 #7 0xb7876186 in destroy_dlg_table () at dlg_hash.c:118 #8 0xb7861d8e in mod_destroy () at dialog.c:529 #9 0x080b3d41 in destroy_modules () at sr_module.c:381 #10 0x08069b60 in cleanup (show_status=1) at main.c:333 #11 0x0806aaf1 in handle_sigs () at main.c:523 #12 0x0806ecc8 in main (argc=1, argv=0xbf9cecc4) at main.c:865 DEBUG from Kamailio: Oct 22 16:25:34 [7817] DBG:tm:t_relay_to: new transaction fwd'ed Oct 22 16:25:34 [7856] CRITICAL:core:receive_fd: EOF on 15 Oct 22 16:25:34 [7856] DBG:core:handle_ser_child: dead child 8, pid 7817 (shutting down?) Oct 22 16:25:34 [7856] DBG:core:io_watch_del: io_watch_del (0x8175e80, 15, -1, 0x0) fd_no=26 called Oct 22 16:25:34 [7803] INFO:core:handle_sigs: child process 7817 exited by a signal 11 Oct 22 16:25:34 [7803] INFO:core:handle_sigs: core was generated Oct 22 16:25:34 [7803] INFO:core:handle_sigs: terminating due to SIGCHLD Oct 22 16:25:34 [7856] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 144400, used+overhead=169568, free=879008 max used (+overhead)= 169568 dumping free list: hash = 115 fragments no.: 1, unused: 0 bucket size: 920 - 920 (first 920) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 873576) TOTAL: 6 free fragments = 879152 free bytes ----------------------------- Oct 22 16:25:34 [7853] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7850] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7847] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7844] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7829] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127784, free=920792 max used (+overhead)= 127864 dumping free list: hash = 1 fragments no.: 1, unused: 0 bucket size: 8 - 8 (first 8) hash = 10 fragments no.: 1, unused: 0 bucket size: 80 - 80 (first 80) hash = 62 fragments no.: 1, unused: 0 bucket size: 496 - 496 (first 496) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 8 free fragments = 920936 free bytes ----------------------------- Oct 22 16:25:34 [7826] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127776, free=920800 max used (+overhead)= 127784 dumping free list: hash = 1 fragments no.: 1, unused: 0 bucket size: 8 - 8 (first 8) hash = 73 fragments no.: 1, unused: 0 bucket size: 584 - 584 (first 584) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 7 free fragments = 920944 free bytes ----------------------------- Oct 22 16:25:34 [7823] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127776, free=920800 max used (+overhead)= 127784 dumping free list: hash = 1 fragments no.: 1, unused: 0 bucket size: 8 - 8 (first 8) hash = 73 fragments no.: 1, unused: 0 bucket size: 584 - 584 (first 584) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 7 free fragments = 920944 free bytes ----------------------------- Oct 22 16:25:34 [7820] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127768, free=920808 max used (+overhead)= 127768 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920952 free bytes ----------------------------- Oct 22 16:25:34 [7814] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=128304, free=920272 max used (+overhead)= 132536 dumping free list: hash = 1 fragments no.: 7, unused: 0 bucket size: 8 - 8 (first 8) hash = 2 fragments no.: 3, unused: 0 bucket size: 16 - 16 (first 16) hash = 3 fragments no.: 7, unused: 0 bucket size: 24 - 24 (first 24) hash = 4 fragments no.: 25, unused: 0 bucket size: 32 - 32 (first 32) hash = 5 fragments no.: 11, unused: 0 bucket size: 40 - 40 (first 40) hash = 6 fragments no.: 3, unused: 0 bucket size: 48 - 48 (first 48) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) hash = 9 fragments no.: 1, unused: 0 bucket size: 72 - 72 (first 72) hash = 10 fragments no.: 1, unused: 0 bucket size: 80 - 80 (first 80) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) hash = 17 fragments no.: 1, unused: 0 bucket size: 136 - 136 (first 136) hash = 27 fragments no.: 2, unused: 0 bucket size: 216 - 216 (first 216) hash = 40 fragments no.: 1, unused: 0 bucket size: 320 - 320 (first 320) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) hash = 94 fragments no.: 1, unused: 0 bucket size: 752 - 752 (first 752) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 914936) TOTAL: 68 free fragments = 920416 free bytes ----------------------------- Oct 22 16:25:34 [7811] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=128240, free=920336 max used (+overhead)= 132296 dumping free list: hash = 1 fragments no.: 5, unused: 0 bucket size: 8 - 8 (first 8) hash = 2 fragments no.: 4, unused: 0 bucket size: 16 - 16 (first 16) hash = 3 fragments no.: 7, unused: 0 bucket size: 24 - 24 (first 24) hash = 4 fragments no.: 19, unused: 0 bucket size: 32 - 32 (first 32) hash = 5 fragments no.: 12, unused: 0 bucket size: 40 - 40 (first 40) hash = 6 fragments no.: 1, unused: 0 bucket size: 48 - 48 (first 48) hash = 8 fragments no.: 1, unused: 0 bucket size: 64 - 64 (first 64) hash = 9 fragments no.: 1, unused: 0 bucket size: 72 - 72 (first 72) hash = 14 fragments no.: 1, unused: 0 bucket size: 112 - 112 (first 112) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) hash = 27 fragments no.: 2, unused: 0 bucket size: 216 - 216 (first 216) hash = 58 fragments no.: 1, unused: 0 bucket size: 464 - 464 (first 464) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 60 free fragments = 920480 free bytes ----------------------------- Oct 22 16:25:34 [7841] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 111616, used+overhead=136744, free=911832 max used (+overhead)= 136744 dumping free list: hash = 118 fragments no.: 1, unused: 0 bucket size: 944 - 944 (first 944) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 907496) TOTAL: 5 free fragments = 911976 free bytes ----------------------------- Oct 22 16:25:34 [7832] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 159496, used+overhead=184608, free=863968 max used (+overhead)= 184608 dumping free list: hash = 129 fragments no.: 1, unused: 0 bucket size: 1032 - 1032 (first 1032) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 858424) TOTAL: 6 free fragments = 864112 free bytes ----------------------------- Oct 22 16:25:34 [7807] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=127808, free=920768 max used (+overhead)= 127808 dumping free list: hash = 70 fragments no.: 1, unused: 0 bucket size: 560 - 560 (first 560) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920912 free bytes ----------------------------- Oct 22 16:25:34 [7806] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=127808, free=920768 max used (+overhead)= 127808 dumping free list: hash = 70 fragments no.: 1, unused: 0 bucket size: 560 - 560 (first 560) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920912 free bytes ----------------------------- Oct 22 16:25:34 [7805] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=127808, free=920768 max used (+overhead)= 127808 dumping free list: hash = 70 fragments no.: 1, unused: 0 bucket size: 560 - 560 (first 560) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920912 free bytes ----------------------------- Oct 22 16:25:34 [7804] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102744, used+overhead=128088, free=920488 max used (+overhead)= 128088 dumping free list: hash = 13 fragments no.: 1, unused: 0 bucket size: 104 - 104 (first 104) hash = 61 fragments no.: 1, unused: 0 bucket size: 488 - 488 (first 488) hash = 101 fragments no.: 1, unused: 0 bucket size: 808 - 808 (first 808) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 7 free fragments = 920632 free bytes ----------------------------- Oct 22 16:25:34 [7835] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 159496, used+overhead=184608, free=863968 max used (+overhead)= 184608 dumping free list: hash = 129 fragments no.: 1, unused: 0 bucket size: 1032 - 1032 (first 1032) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 858424) TOTAL: 6 free fragments = 864112 free bytes ----------------------------- Oct 22 16:25:34 [7838] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 159496, used+overhead=184608, free=863968 max used (+overhead)= 184608 dumping free list: hash = 129 fragments no.: 1, unused: 0 bucket size: 1032 - 1032 (first 1032) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 858424) TOTAL: 6 free fragments = 864112 free bytes ----------------------------- Oct 22 16:25:34 [7808] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102600, used+overhead=128496, free=920080 max used (+overhead)= 132720 dumping free list: hash = 1 fragments no.: 7, unused: 0 bucket size: 8 - 8 (first 8) hash = 2 fragments no.: 5, unused: 0 bucket size: 16 - 16 (first 16) hash = 3 fragments no.: 9, unused: 0 bucket size: 24 - 24 (first 24) hash = 4 fragments no.: 31, unused: 0 bucket size: 32 - 32 (first 32) hash = 5 fragments no.: 15, unused: 0 bucket size: 40 - 40 (first 40) hash = 6 fragments no.: 3, unused: 0 bucket size: 48 - 48 (first 48) hash = 8 fragments no.: 2, unused: 0 bucket size: 64 - 64 (first 64) hash = 9 fragments no.: 1, unused: 0 bucket size: 72 - 72 (first 72) hash = 12 fragments no.: 1, unused: 0 bucket size: 96 - 96 (first 96) hash = 14 fragments no.: 1, unused: 0 bucket size: 112 - 112 (first 112) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) hash = 27 fragments no.: 2, unused: 0 bucket size: 216 - 216 (first 216) hash = 46 fragments no.: 1, unused: 0 bucket size: 368 - 368 (first 368) hash = 115 fragments no.: 1, unused: 0 bucket size: 920 - 920 (first 920) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 914768) TOTAL: 82 free fragments = 920224 free bytes ----------------------------- Oct 22 16:25:34 [7803] DBG:dialog:destroy_dlg: destroing dialog 0xb5a063a0 Oct 22 16:25:34 [7803] DBG:dialog:run_dlg_callbacks: dialog=0xb5a063a0, type=2048 Any suggestions? With best regards Marian Piater -- Marian Piater LAM plus s.r.o. odd. hlasové komunikace - VoIP 242 413 000 Do práce 605 267 612 Mobilní telefon marian.piater(a)lam.cz LAM plus s.r.o. Mezi vodami 19 143 00 Praha - Modřany P Prosím, uvědomte si svou zodpovědnost za životní prostředí, než vytisknete tento e-mail.

2 1

[Kamailio-Users] OpenSER ActiveWatcher Table
by George Lee 22 Oct '08

22 Oct '08

Hi Expert, When a SUBSCRIBE message is received, OpenSER creates an entries in the ActiveWatcher table. This is used to generate NOTIFY message to watchers. The problem I have seen is that OpenSER keeps on creating new entries to the ActiveWatchers if the SIP UA deregister, register, deregister, register, etc. for several times. This triggers multiple NOTIFY messages to be sent to watchers. How does OpenSER remove entries in the ActiveWatcher table? It deosn't look like a SUBSCRIBE with expires=0 would delete an entry in the table. Thanks, George

3 3

[Kamailio-Users] URI length
by VoIP Forums www.Go4Calls.com 22 Oct '08

22 Oct '08

Hi All, I was trying to make dialplan if dialed number length >= 8 it should go for PSTN call otherwise local call. I did the following but PSTN is working well if length is greater than 8 but when number comes for local call i am getting the error "Message to Big" if ($(rU{s.len}) >= 8) { rewritehostport("89.21.XXX.XX:5060"); route(1); exit; }else{ lookup("location"); route(1); exit; }; Thank psl suggest us how i can do the local call if it is < to 8 thank you _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

2 1

[Kamailio-Users] Record Route and proxy behind static NAT
by Zappasodi Daniele 22 Oct '08

22 Oct '08

Hello, I have my proxy behind a NAT (static NAT) with SIP clients that are both on my LAN and on public IP. UAC1 Private IP <--> <--> UAC3 PublicIP Proxy <--> NAT <--> WAN UAC2 Private IP <--> <--> UAC4 PublicIP I'm looking for the correct configuration of record route for this scenario. I looked for examples with proxy behind the NAT and I have always found a config section for the record route like this # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("<MY OPENSER PBULIC IP>:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; In this way the proxy uses "record_route_preset" when the caller is private (client_nat_test("3")) and "record_route" when the caller is public, without consider the callee. Consequently the proxy puts in the record route the public IP when the caller is private and the private IP address when the caller is public. I can't understand how it can work: when a client on an public ip calls a client on an public ip, they find in record route the private ip and they will try to send the SIP method (ACK,BYE, reINVITE ecc) to a private IP, so this messages will be lost. This is the beahaviour that I'm observing in my test. Maybe this configuration isn't adapted for my scenario or I miss something else (I have already added advertised_address/port) or I haven't correctly understand the instructions or the Record Route's usage. ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 1

[Kamailio-Users] UUID column in usr_preferences - How unique does it need to be?
by Graham Wooden 22 Oct '08

22 Oct '08

Hi, By looking at the indexes set for the table usr_preferences, it appears that UUID is going to be unique against the attribute column. So if user A has a attribute of custom_parameter_Z and user B has the same attribute, then both UUIDs need to be different, right? UUID can really be anything right? Its set to VARCHAR(64), so heck - a MD5 checksum against say a epoch timestamp would be fine. I just want to confirm how unique each row (in respects to the UUID) it needs to be. Thanks, -graham

2 2

[Kamailio-Users] Registrar performance
by Pascal Maugeri 22 Oct '08

22 Oct '08

Hi I would like to have some rough estimate of the registrar performance (with DB). If there is no figure available, does it make sense to use call performance benchmark like this one http://www.transnexus.com/White%20Papers/OpenSER_RTPproxy_test.htm and extrapolate to registrar performances ? eg. can I say that the SIP REGISTRAR is able to handle >600 registrations per second in the case of TransNexus testbed ? Regards, Pascal

3 3

[Kamailio-Users] Pike module is not working!
by Arif-Uz-Zaman 21 Oct '08

21 Oct '08

Hi, I've configured OpenSER with Pike module but it's not working as expected (doesn't block IP according to the configured time). J Here is a simple configuration by considering the following: ---------------------------------------------------------------------------- --------------- modparam("pike", "sampling_time_unit", 60) #1 minute modparam("pike", "reqs_density_per_unit", 200) #200 requests modparam("pike", "remove_latency", 7200) #Block IP for 2 hours if (!pike_check_req()) xlog("L_CRIT","#########TOO MANY HITS FROM $si##########\n"); exit; } ---------------------------------------------------------------------------- ----------------- If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Action: Block IP for 2 hours. I'm not sure whether I'm doing wrong or is there any problem with Pike module? Please help me with valuable feedback. Thanks in advance! ARIF

1 0

[Kamailio-Users] Pike module is not working!
by Arif-Uz-Zaman 21 Oct '08

21 Oct '08

Hi, I've configured OpenSER with Pike module but it's not working as expected (doesn't block IP according to the configured time). :-) Here is a simple configuration by considering the following: ---------------------------------------------------------------------------- --------------- modparam("pike", "sampling_time_unit", 60) #1 minute modparam("pike", "reqs_density_per_unit", 200) #200 requests modparam("pike", "remove_latency", 7200) #Block IP for 2 hours if (!pike_check_req()) xlog("L_CRIT","#########TOO MANY HITS FROM $si##########\n"); exit; } ---------------------------------------------------------------------------- ----------------- If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Action: Block IP for 2 hours. I'm not sure whether I'm doing wrong or is there any problem with Pike module? Please help me with valuable feedback. Thanks in advance! ARIF

1 0

[Kamailio-Users] Re write any SIP Header Value!
by arif.zaman 21 Oct '08

21 Oct '08

Hi, Is it possible to update any SIP Header (i.e. User-Agent) value before it saving to “location” database? If yes, please help me with sample config. For example: User-Agent value always save as "arif" in location database. Thanks in advance, ARIF -- View this message in context: http://www.nabble.com/Rewrite-any-SIP-Header-Value%21-tp20067253p20067253.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 7

[Kamailio-Users] (moved to kamailio)Why PUA_BLA generates a SUBSCRIBE "Event: dialog" that is never supported by OpenSIPS
by Klaus Darilion 21 Oct '08

21 Oct '08

Iñaki Baz Castillo wrote: > El Domingo, 19 de Octubre de 2008, Iñaki Baz Castillo escribió: >>> Hi, I using PUA_BLA and PUA_DIALOGINFO modules. >>> >>> I can't understand why when the proxy receives an INVITE it generates a >>> SUBSCRIBE to itselfs with: >>> Event: dialog >>> >>> Shouldn't it be: >>> Event: dialog;sla >>> ? > > Well, I've realized that it's because PUA_DIALOG info (Event: dialog). > > And more sorry! since it's just in Kamailio. The pua_dialoginfo modules sends SUBSCRIBE requests to itself? It can not send SUBSCRIBE - only PUBLISH. I guess the SUBSCRIBE is coming from pua_bla regards klaus

2 1

[Kamailio-Users] Remove function
by michel freiha 21 Oct '08

21 Oct '08

Dear All, I have the following function: if($rU =~ "^[+].*") { route(8); exit; } I need to remove the + sign from the dilaed number before sending the call to route(8) Regards

2 1

[Serusers] simple cluster setup with asterisk
by Uwe Kastens 21 Oct '08

21 Oct '08

Hello, Can anybody give me a hint how to start with a very simple cluster setup? I just want to route SIP to 2 different asterisk server. There is no need to care about NAT or call routing. I was wondering if its enough to use the dispatch module. Is it possible with the dispatch module to achieve that the system will reuse the information where a user is registered to route the calls to the same system? BR kiste -- kiste lat: 54.322684, lon: 10.13586

1 0

[Kamailio-Users] t_relay("0x02")
by jh＠tutpro.com 20 Oct '08

20 Oct '08

tm readme on t_relay flag "0x02" says: 0x02 - do not internally send a negative reply in case of forward failure (due internal error, bad RURI, bad message, etc). When a forward failure occurs, no SIP request is relayed and therefore no negative reply or timeout will show up on the failure_route (if one is set). It applies only when the transaction is created. By default one negative reply is sent. Useful if you want to implement a serial forking in case of failure. i don't quite understand the statement: It applies only when the transaction is created. what does "It" refer to? i did some experiments with new lcr implementation and if, for example, tcp connection to a gw cannot be established, then t_relay without "0x02" flag generates 477 reply to caller and, indeed, failure route is not executed. so i need test if t_relay("0x02") failed due to "send failed" and handle selection of next gw in the same route block where t_relay("0x02") was called. this is inconvenient, because there already exists a failure route block that does all that and now its code needs to be at least partially duplicated. or am i missing something and it is somehow possible to get a failure route block executed also on this kind of internally generated negative reply? -- juha

2 2

[Kamailio-Users] $rr and $rr are null
by jh＠tutpro.com 20 Oct '08

20 Oct '08

i have if (t_check_status("403")) { xlog("L_INFO", "DBG: status is 403\n, rr is <$rr> and rs is <$rs>\n"); }; and i get to syslog: Oct 18 19:51:37 localhost /usr/sbin/kamailio[18580]: DBG: status is 403 , rr is <<null>> and rs is <<null>> is this expected behavior? -- juha

2 4

Re: [Kamailio-Users] opensips db
by Alex Balashov 20 Oct '08

20 Oct '08

waleed hassn wrote: > On Mon, Oct 20, 2008 at 5:41 AM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> waleed hassn wrote: >>> On Mon, Oct 20, 2008 at 4:25 AM, Alex Balashov >>> <abalashov(a)evaristesys.com> wrote: >>>> waleed hassn wrote: >>>>> hi all >>>>> i am new in opensips , how can i set database and add users >>>> Which database, and what users? >>>> >>>> >>>> -- >>>> Alex Balashov >>>> Evariste Systems >>>> Web : http://www.evaristesys.com/ >>>> Tel : (+1) (678) 954-0670 >>>> Direct : (+1) (678) 954-0671 >>>> Mobile : (+1) (706) 338-8599 >>>> >>> sip users and mysql database >>> an error occur when i execute opensipsdb command >> 1) Learn to set up and administer MySQL. >> >> 2) What error? >> >> 3) It might be beneficial to post to the OpenSIPS-Users list. >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> Mobile : (+1) (706) 338-8599 >> > thanks , can you get me some useful links to begin First result on Google for "mysql tutorial": http://dev.mysql.com/doc/refman/5.0/en/tutorial.html -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

Re: [Kamailio-Users] opensips db
by Alex Balashov 20 Oct '08

20 Oct '08

waleed hassn wrote: > On Mon, Oct 20, 2008 at 4:25 AM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> waleed hassn wrote: >>> hi all >>> i am new in opensips , how can i set database and add users >> Which database, and what users? >> >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> Mobile : (+1) (706) 338-8599 >> > > sip users and mysql database > an error occur when i execute opensipsdb command 1) Learn to set up and administer MySQL. 2) What error? 3) It might be beneficial to post to the OpenSIPS-Users list. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

[Kamailio-Users] opensips db
by waleed hassn 20 Oct '08

20 Oct '08

hi all i am new in opensips , how can i set database and add users thanks

2 1

[Kamailio-Users] RLS and presence Server
by mahesh.peddi＠info-spectrum.com 20 Oct '08

20 Oct '08

Hello, I am able to see RLS properly working when RLS and presence functionality is in the same server. after that, I tried to integrate RLS as a separate application server. To forward the message from RLS to presence server, I made below changes in the config file: route[2] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("NOTIFY")) { rls_handle_notify(); t_release(); } if(is_method("SUBSCRIBE")) { $var(ret_code)= rls_handle_subscribe(); if($var(ret_code)== 10) #to forward to presence server t_relay("udp:<ip>:5070"); } if(is_method("PUBLISH")) { #to forward to presence server. t_relay("udp:<ip>:5070"); t_release(); } exit; } because of this configuration, i found two problems: 1. when i am forwarding subscribe message (from RLS to presence server), its not sending the subscribe for all the contacts. Here, i am expecting (sort of) output of rls_handle_subscribe() to be sent to the presence server. Which is not happening. 2. When presence server is sending notify message, it is sending directly to scscf..instead of rls.. Can anybody let me know, how can I get the desired functionality of: 1. UE --> RLS --> presence server (subscribe) 2. Presence server --> RLS --> UE (for notify). If there is any other way to forward message from RLS to presence, please let me know Regards, Mahesh Peddi

1 0

Re: [Kamailio-Users] Check if a user is calling from aregistered device and if not, deny the call!
by Daniel-Constantin Mierla 20 Oct '08

20 Oct '08

Hello, On 10/19/08 14:32, Arif-Uz-Zaman wrote: > Hi, > > Here is a tricky sample config to check if a user is calling from a > registered device and if not, deny the call: > > $var(temp) = $ruri; > $ruri = $fu; > if(!registered("location")) { > sl_send_reply("403","Forbidden auth ID"); > xlog("<ARIF> Not Found\n"); > exit; > } > $ruri = $var(temp); > in the trunk (so available with upcoming 1.5.0) you can do more strict checks. While you are sure an user is registered, you cannot detect if he is calling from registered device or a different one. You just know it has a valid contact in location database. Here is an example to check Contact address in INVITE is same as one in location table: http://openser.blogspot.com/2008/10/registrar-enhancements.html As you get access to all attributes stored for a location record, you can do many other tests, like source IP and port, etc. Cheers, Daniel > Thanks, > ARIF > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Juha Heinanen > Sent: Sunday, October 19, 2008 3:44 PM > To: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] Check if a user is calling from > aregistereddevice and if not, deny the call! > > Arif-Uz-Zaman writes: > > > I've already tested with the following config but no luck. Since the > > following config check only whether callee is registered or not. Not the > > caller. > > > > if(!registered("location")) { > > sl_send_reply("403","Forbidden auth ID"); > > exit; > > } > > indeed the function should take a pseudo variable that contains aor used > for checking. > > > Is there any other suggestions? > > save r-uri, store caller's aor to r-uri and call the above function. > then restore r-uri. > > -- juha > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] load/next_contacts question
by Klaus Darilion 20 Oct '08

20 Oct '08

Hi Juha and others! I wonder if I understand load/next_contacts(). If I understand it right it can be used for example to make parallel/serial forking after an ENUM lookup (multiple NAPTRs). I think this is the correct usage: if (enum_query()) { if (load_contacts()) { if (next_contacts()) { t_on_failure("1"); t_relay(); } else { sl_reply("500","error adding contacts"); } } else { sl_reply("500","error fetching destination set"); } exit; } But I wonder what if all the branches after enum_query() have the same q value. According to the README: load_contact: If all contacts in the destination set have the same qvalue, load_contacts() does not do anything thus minimizing performance impact of sequential forking capability when it is not needed. Returns 1 if loading of contacts succeeded or there was nothing to do. Thus, next_contacts() will be called although there are no AVPs to fetch. Do I miss something here? regards klaus

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Re: [Kamailio-Users] Check if a user is calling from a registereddevice and if not, deny the call!
by jh＠tutpro.com 20 Oct '08

20 Oct '08

Arif-Uz-Zaman writes: > I've already tested with the following config but no luck. Since the > following config check only whether callee is registered or not. Not the > caller. > > if(!registered("location")) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } indeed the function should take a pseudo variable that contains aor used for checking. > Is there any other suggestions? save r-uri, store caller's aor to r-uri and call the above function. then restore r-uri. -- juha

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