sr-users October 2008

sr-users@lists.kamailio.org

93 participants
169 discussions

by Daniel-Constantin Mierla

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

16 years

[Kamailio-Users] restrict calls from gateway to some extensions

by muhammad akl

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 ?

16 years

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

by Ricky Gutierrez

________________________________ 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

16 years

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

by Ricky Gutierrez

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......."

16 years

[Serusers] SER/FIFO not accessible

by Shruti Sanadhya

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!!

16 years

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

by luzango mfupe

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......."

16 years

[Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

by Ricky Gutierrez

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

16 years

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

by Ricky Gutierrez

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,

16 years

[Kamailio-Users] Fw: version cdrtool mediaproxy 1.9.1

by Ricky Gutierrez

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

16 years

[Kamailio-Users] Dimensioning a telephony system based on openser!

by Nuno Marques

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

16 years

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sr-users October 2008