Re: [OpenSIPS-Users] opensips and NAT
Hi, Should I try something like nat_uac_test(diff-ip-src-contact || private-contact || diff-ip-src-via || diff-port-src-via)? Any ideas? Regards, Ronald December 15, 2023 at 4:38 PM, r...@rvgeerligs.nl wrote: > > Hi > > I use opensips 3.4 and I have NAT problems with 2 devices behind the same NAT > (called party hears nothing). > The A party is softphone on iPhone (linphone) the B (called)party is > Polycom310. The other way around works (Polycom calls linphone). > > Actually I tested this in two different NAT locations. > The first location repeatedly works. The second location gives the problem. > Both locations have changing public IP addresses and DHCP on 192.168 network. > Both locations have one FX number assigned to them. No SIP ALG active. > > I use nat_uac_test(diff-ip-src-contact). > > There is a table: > 1.5.5. nat_uac_test(flags) > Tries to guess if client's request originated behind a nat. The parameter > determines what heuristics is used. > > Meaning of the flags (string) parameter is as follows: > > private-contact - (old 1 flag) Contact header field is searched for > occurrence of RFC1918 / RFC6598 addresses. > > diff-ip-src-via - (old 2 flag) the "received" test is used: address in Via is > compared against source IP address of signaling > > private-via - (old 4 flag) Top Most VIA is searched for occurrence of RFC1918 > / RFC6598 addresses > > private-sdp - (old 8 flag) SDP is searched for occurrence of RFC1918 / > RFC6598 addresses > > diff-port-src-via - (old 16 flag) test if the source port is different from > the port in Via > > diff-ip-src-contact - (old 32 flag) address in Contact is compared against > source IP address of signaling. > > diff-port-src-contact - (old 64 flag) Port in Contact is compared against > source port of signaling > > carrier-grade-nat - (old 128 flag) also include RFC 6333 addresses in the > checks for ct, via and sdp flags. > > A CSV of the above flags can be provided, the test returns true if any of the > tests identified a NAT. > > Currently I use old flag 32. > > I read that using the equivalent of 19 might help but I dont see that in the > table. > > Any advice is appreciated. > > Regards, > > Ronald Geerligs >___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips and NAT
Hi I use opensips 3.4 and I have NAT problems with 2 devices behind the same NAT (called party hears nothing). The A party is softphone on iPhone (linphone) the B (called)party is Polycom310. The other way around works (Polycom calls linphone). Actually I tested this in two different NAT locations. The first location repeatedly works. The second location gives the problem. Both locations have changing public IP addresses and DHCP on 192.168 network. Both locations have one FX number assigned to them. No SIP ALG active. I use nat_uac_test(diff-ip-src-contact). There is a table: 1.5.5. nat_uac_test(flags) Tries to guess if client's request originated behind a nat. The parameter determines what heuristics is used. Meaning of the flags (string) parameter is as follows: private-contact - (old 1 flag) Contact header field is searched for occurrence of RFC1918 / RFC6598 addresses. diff-ip-src-via - (old 2 flag) the "received" test is used: address in Via is compared against source IP address of signaling private-via - (old 4 flag) Top Most VIA is searched for occurrence of RFC1918 / RFC6598 addresses private-sdp - (old 8 flag) SDP is searched for occurrence of RFC1918 / RFC6598 addresses diff-port-src-via - (old 16 flag) test if the source port is different from the port in Via diff-ip-src-contact - (old 32 flag) address in Contact is compared against source IP address of signaling. diff-port-src-contact - (old 64 flag) Port in Contact is compared against source port of signaling carrier-grade-nat - (old 128 flag) also include RFC 6333 addresses in the checks for ct, via and sdp flags. A CSV of the above flags can be provided, the test returns true if any of the tests identified a NAT. Currently I use old flag 32. I read that using the equivalent of 19 might help but I dont see that in the table. Any advice is appreciated. Regards, Ronald Geerligs___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Thanks for the responses. They helped me exclude some things. I've managed to make progress and pinned down the lack of audio to a misconfiguration of Mediaproxy. Two-way audio through double-nat / firewall is working but goes silent after about 60 seconds connected and Asterisk kills the connection 31 seconds later due to lack of RTP activity for the last 31 seconds On Thu, 14 Jan 2021 at 12:00, David Villasmil < david.villasmil.w...@gmail.com> wrote: > Check out what IPs are offered in the SDPs in asterisk. Make sure they’re > both public IPs. > If you only have 1 asterisk, forwarding the rtp port range configured in > asterisk from the firewall to asterisk should do it. > > > On Thu, 14 Jan 2021 at 08:23, Mark Allen wrote: > >> Thanks Adrian >> >> The firewall has SIP-ALG disabled and just forwards ports from externally >> to where they need to be internally - so ports 5060 and 1 - 65535 of >> 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box) >> >> On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu >> wrote: >> >>> Google search for SIP ALG problem to see if this is relevant for your >>> case. >>> >>> Regards, >>> Adrian >>> >>> >>> On 13 Jan 2021, at 13:08, Mark Allen wrote: >>> >>> Hi all - I've been banging my head against this but not succeeding. >>> >>> Our setup... >>> >>> UAC 192.168.x.x >>> | >>> Router5.x.x.x >>> | >>> (internet) >>> | >>> Firewall 46.x.x.x maps >>> | directly to >>> OpenSIPS 192.168.x.x Mid-registrar >>> | >>> Asterisk 192.168.x.x >>> >>> >>> Current situation: >>> - UAC can register on Asterisk via OpenSIPS >>> - UAC can call destination registered on Asterisk on local n/w to >>> Asterisk box >>> - Destination extension rings and can pick up call >>> - There is no audio either way & call drops after about 30 secs >>> (Asterisk kills call with "Requested channel not available" because not >>> RTP traffic is reaching destination) >>> >>> I have tried passing audio through Mediaproxy on OpenSIPS box but with >>> no success. Using Wireshark I can see RTP traffic initiated at both ends, >>> but it doesn't reach the other end either way. >>> >>> Is there some definitive guide to setting this up correctly or are there >>> specific steps that I need to follow? >>> >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -- > Regards, > > David Villasmil > email: david.villasmil.w...@gmail.com > phone: +34669448337 > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Check out what IPs are offered in the SDPs in asterisk. Make sure they’re both public IPs. If you only have 1 asterisk, forwarding the rtp port range configured in asterisk from the firewall to asterisk should do it. On Thu, 14 Jan 2021 at 08:23, Mark Allen wrote: > Thanks Adrian > > The firewall has SIP-ALG disabled and just forwards ports from externally > to where they need to be internally - so ports 5060 and 1 - 65535 of > 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box) > > On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu wrote: > >> Google search for SIP ALG problem to see if this is relevant for your >> case. >> >> Regards, >> Adrian >> >> >> On 13 Jan 2021, at 13:08, Mark Allen wrote: >> >> Hi all - I've been banging my head against this but not succeeding. >> >> Our setup... >> >> UAC 192.168.x.x >> | >> Router5.x.x.x >> | >> (internet) >> | >> Firewall 46.x.x.x maps >> | directly to >> OpenSIPS 192.168.x.x Mid-registrar >> | >> Asterisk 192.168.x.x >> >> >> Current situation: >> - UAC can register on Asterisk via OpenSIPS >> - UAC can call destination registered on Asterisk on local n/w to >> Asterisk box >> - Destination extension rings and can pick up call >> - There is no audio either way & call drops after about 30 secs (Asterisk >> kills call with "Requested channel not available" because not RTP >> traffic is reaching destination) >> >> I have tried passing audio through Mediaproxy on OpenSIPS box but with no >> success. Using Wireshark I can see RTP traffic initiated at both ends, but >> it doesn't reach the other end either way. >> >> Is there some definitive guide to setting this up correctly or are there >> specific steps that I need to follow? >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.w...@gmail.com phone: +34669448337 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Thanks Adrian The firewall has SIP-ALG disabled and just forwards ports from externally to where they need to be internally - so ports 5060 and 1 - 65535 of 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box) On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu wrote: > Google search for SIP ALG problem to see if this is relevant for your case. > > Regards, > Adrian > > > On 13 Jan 2021, at 13:08, Mark Allen wrote: > > Hi all - I've been banging my head against this but not succeeding. > > Our setup... > > UAC 192.168.x.x > | > Router5.x.x.x > | > (internet) > | > Firewall 46.x.x.x maps > | directly to > OpenSIPS 192.168.x.x Mid-registrar > | > Asterisk 192.168.x.x > > > Current situation: > - UAC can register on Asterisk via OpenSIPS > - UAC can call destination registered on Asterisk on local n/w to Asterisk > box > - Destination extension rings and can pick up call > - There is no audio either way & call drops after about 30 secs (Asterisk > kills call with "Requested channel not available" because not RTP traffic > is reaching destination) > > I have tried passing audio through Mediaproxy on OpenSIPS box but with no > success. Using Wireshark I can see RTP traffic initiated at both ends, but > it doesn't reach the other end either way. > > Is there some definitive guide to setting this up correctly or are there > specific steps that I need to follow? > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Google search for SIP ALG problem to see if this is relevant for your case. Regards, Adrian > On 13 Jan 2021, at 13:08, Mark Allen wrote: > > Hi all - I've been banging my head against this but not succeeding. > > Our setup... > > UAC 192.168.x.x > | > Router5.x.x.x > | > (internet) > | > Firewall 46.x.x.x maps > | directly to > OpenSIPS 192.168.x.x Mid-registrar > | > Asterisk 192.168.x.x > > > Current situation: > - UAC can register on Asterisk via OpenSIPS > - UAC can call destination registered on Asterisk on local n/w to Asterisk box > - Destination extension rings and can pick up call > - There is no audio either way & call drops after about 30 secs (Asterisk > kills call with "Requested channel not available" because not RTP traffic is > reaching destination) > > I have tried passing audio through Mediaproxy on OpenSIPS box but with no > success. Using Wireshark I can see RTP traffic initiated at both ends, but it > doesn't reach the other end either way. > > Is there some definitive guide to setting this up correctly or are there > specific steps that I need to follow? > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Firewall is not sip aware, rtprelay via box in dmz Outlook voor iOS<https://aka.ms/o0ukef> downloaden Van: Users namens Mark Allen Verzonden: Wednesday, January 13, 2021 5:08:27 PM Aan: OpenSIPS users mailling list Onderwerp: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues Hi all - I've been banging my head against this but not succeeding. Our setup... UAC 192.168.x.x | Router5.x.x.x | (internet) | Firewall 46.x.x.x maps | directly to OpenSIPS 192.168.x.x Mid-registrar | Asterisk 192.168.x.x Current situation: - UAC can register on Asterisk via OpenSIPS - UAC can call destination registered on Asterisk on local n/w to Asterisk box - Destination extension rings and can pick up call - There is no audio either way & call drops after about 30 secs (Asterisk kills call with "Requested channel not available" because not RTP traffic is reaching destination) I have tried passing audio through Mediaproxy on OpenSIPS box but with no success. Using Wireshark I can see RTP traffic initiated at both ends, but it doesn't reach the other end either way. Is there some definitive guide to setting this up correctly or are there specific steps that I need to follow? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Hi all - I've been banging my head against this but not succeeding. Our setup... UAC 192.168.x.x | Router5.x.x.x | (internet) | Firewall 46.x.x.x maps | directly to OpenSIPS 192.168.x.x Mid-registrar | Asterisk 192.168.x.x Current situation: - UAC can register on Asterisk via OpenSIPS - UAC can call destination registered on Asterisk on local n/w to Asterisk box - Destination extension rings and can pick up call - There is no audio either way & call drops after about 30 secs (Asterisk kills call with "Requested channel not available" because not RTP traffic is reaching destination) I have tried passing audio through Mediaproxy on OpenSIPS box but with no success. Using Wireshark I can see RTP traffic initiated at both ends, but it doesn't reach the other end either way. Is there some definitive guide to setting this up correctly or are there specific steps that I need to follow? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS behing NAT as TLS/UDP Proxy
Hi All, Can anyone send a sample configuration of OpenSIPS which behind NAT and doing connects to clients on TLS and forwards to Softswitch on UDP? -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10 NAT
I didnt troxlinux, I forced all my traffic through rtpproxy Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/23 troxlinux xserverli...@gmail.com Hi, I have the same problem, did you solve? 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com I think I found the problem .. Looking at my SIP Messages, the VIA and the Contact headers doesnt have my INVALID IP, it shows me my VALID IP. But I dont know how to set that, im doing the fixes for nated contacts. Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com I'm running out of ideas .. My rtpproxy is fine Oct 4 09:10:02 opensips /sbin/opensips[5019]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5020]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5018]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5022]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5023]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5017]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5016]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled and here is my opensips.cfg, ### 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 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 /* comment the next line to enable the auto discovery of local aliases based on revers DNS on IPs */ auto_aliases=no listen=udp:###.###.###.###:5060 disable_tcp=yes db_default_url=mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips ### Modules Section #set module path mpath=/lib64/opensips/modules/ SIGNALING module loadmodule signaling.so StateLess module loadmodule sl.so Transaction Module loadmodule tm.so modparam(tm, fr_timer, 5) modparam(tm, fr_inv_timer, 30) modparam(tm, restart_fr_on_each_reply, 0) modparam(tm, onreply_avp_mode, 1) Record Route Module loadmodule rr.so /* do not append from tag to the RR (no need for this script) */ modparam(rr, append_fromtag, 0) MAX ForWarD module loadmodule maxfwd.so SIP MSG OPerationS module loadmodule sipmsgops.so FIFO Management Interface loadmodule mi_fifo.so modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0666) URI module loadmodule uri.so modparam(uri, use_uri_table, 0) MYSQL module loadmodule db_mysql.so USeR LOCation module loadmodule usrloc.so modparam(usrloc, nat_bflag, NAT) modparam(usrloc, db_mode, 2) modparam(usrloc, db_url, mysql://:@##/opensips) REGISTRAR module loadmodule registrar.so modparam(registrar, tcp_persistent_flag, TCP_PERSISTENT) modparam(registrar, received_avp, $avp(received_nh)) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam(registrar, max_contacts, 10) ACCounting module loadmodule acc.so /* what special events should be accounted ? */ modparam(acc, early_media, 0) modparam(acc, report_cancels, 0) /* by default we 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) modparam(acc, failed_transaction_flag, ACC_FAILED) /* account triggers (flags) */ modparam(acc, db_flag, ACC_DO) modparam(acc, db_missed_flag, ACC_MISSED) modparam(acc, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) AUTHentication modules loadmodule auth.so loadmodule auth_db.so modparam(auth_db, calculate_ha1, yes) modparam(auth_db, password_column, password) modparam(auth_db, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) modparam(auth_db, load_credentials, ) ALIAS module loadmodule alias_db.so modparam(alias_db, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) DOMAIN module loadmodule domain.so modparam(domain, db_url,
Re: [OpenSIPS-Users] Opensips 1.10 NAT
You can set a flag on usr_preferences to force the nat to that customer. 2013/10/23 troxlinux xserverli...@gmail.com Hi, I have the same problem, did you solve? 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com I think I found the problem .. Looking at my SIP Messages, the VIA and the Contact headers doesnt have my INVALID IP, it shows me my VALID IP. But I dont know how to set that, im doing the fixes for nated contacts. Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com I'm running out of ideas .. My rtpproxy is fine Oct 4 09:10:02 opensips /sbin/opensips[5019]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5020]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5018]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5022]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5023]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5017]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5016]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled and here is my opensips.cfg, ### 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 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 /* comment the next line to enable the auto discovery of local aliases based on revers DNS on IPs */ auto_aliases=no listen=udp:###.###.###.###:5060 disable_tcp=yes db_default_url=mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips ### Modules Section #set module path mpath=/lib64/opensips/modules/ SIGNALING module loadmodule signaling.so StateLess module loadmodule sl.so Transaction Module loadmodule tm.so modparam(tm, fr_timer, 5) modparam(tm, fr_inv_timer, 30) modparam(tm, restart_fr_on_each_reply, 0) modparam(tm, onreply_avp_mode, 1) Record Route Module loadmodule rr.so /* do not append from tag to the RR (no need for this script) */ modparam(rr, append_fromtag, 0) MAX ForWarD module loadmodule maxfwd.so SIP MSG OPerationS module loadmodule sipmsgops.so FIFO Management Interface loadmodule mi_fifo.so modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0666) URI module loadmodule uri.so modparam(uri, use_uri_table, 0) MYSQL module loadmodule db_mysql.so USeR LOCation module loadmodule usrloc.so modparam(usrloc, nat_bflag, NAT) modparam(usrloc, db_mode, 2) modparam(usrloc, db_url, mysql://:@##/opensips) REGISTRAR module loadmodule registrar.so modparam(registrar, tcp_persistent_flag, TCP_PERSISTENT) modparam(registrar, received_avp, $avp(received_nh)) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam(registrar, max_contacts, 10) ACCounting module loadmodule acc.so /* what special events should be accounted ? */ modparam(acc, early_media, 0) modparam(acc, report_cancels, 0) /* by default we 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) modparam(acc, failed_transaction_flag, ACC_FAILED) /* account triggers (flags) */ modparam(acc, db_flag, ACC_DO) modparam(acc, db_missed_flag, ACC_MISSED) modparam(acc, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) AUTHentication modules loadmodule auth.so loadmodule auth_db.so modparam(auth_db, calculate_ha1, yes) modparam(auth_db, password_column, password) modparam(auth_db, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) modparam(auth_db, load_credentials, ) ALIAS module loadmodule alias_db.so modparam(alias_db, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) DOMAIN module loadmodule domain.so modparam(domain, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) modparam(domain, db_mode,
Re: [OpenSIPS-Users] Opensips 1.10 NAT
Hi, I have the same problem, did you solve? 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com I think I found the problem .. Looking at my SIP Messages, the VIA and the Contact headers doesnt have my INVALID IP, it shows me my VALID IP. But I dont know how to set that, im doing the fixes for nated contacts. Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com I'm running out of ideas .. My rtpproxy is fine Oct 4 09:10:02 opensips /sbin/opensips[5019]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5020]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5018]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5022]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5023]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5017]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5016]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled and here is my opensips.cfg, ### 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 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 /* comment the next line to enable the auto discovery of local aliases based on revers DNS on IPs */ auto_aliases=no listen=udp:###.###.###.###:5060 disable_tcp=yes db_default_url=mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips ### Modules Section #set module path mpath=/lib64/opensips/modules/ SIGNALING module loadmodule signaling.so StateLess module loadmodule sl.so Transaction Module loadmodule tm.so modparam(tm, fr_timer, 5) modparam(tm, fr_inv_timer, 30) modparam(tm, restart_fr_on_each_reply, 0) modparam(tm, onreply_avp_mode, 1) Record Route Module loadmodule rr.so /* do not append from tag to the RR (no need for this script) */ modparam(rr, append_fromtag, 0) MAX ForWarD module loadmodule maxfwd.so SIP MSG OPerationS module loadmodule sipmsgops.so FIFO Management Interface loadmodule mi_fifo.so modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0666) URI module loadmodule uri.so modparam(uri, use_uri_table, 0) MYSQL module loadmodule db_mysql.so USeR LOCation module loadmodule usrloc.so modparam(usrloc, nat_bflag, NAT) modparam(usrloc, db_mode, 2) modparam(usrloc, db_url, mysql://:@##/opensips) REGISTRAR module loadmodule registrar.so modparam(registrar, tcp_persistent_flag, TCP_PERSISTENT) modparam(registrar, received_avp, $avp(received_nh)) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam(registrar, max_contacts, 10) ACCounting module loadmodule acc.so /* what special events should be accounted ? */ modparam(acc, early_media, 0) modparam(acc, report_cancels, 0) /* by default we 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) modparam(acc, failed_transaction_flag, ACC_FAILED) /* account triggers (flags) */ modparam(acc, db_flag, ACC_DO) modparam(acc, db_missed_flag, ACC_MISSED) modparam(acc, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) AUTHentication modules loadmodule auth.so loadmodule auth_db.so modparam(auth_db, calculate_ha1, yes) modparam(auth_db, password_column, password) modparam(auth_db, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) modparam(auth_db, load_credentials, ) ALIAS module loadmodule alias_db.so modparam(alias_db, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) DOMAIN module loadmodule domain.so modparam(domain, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) modparam(domain, db_mode, 1) # Use caching modparam(auth_db|usrloc|uri, use_domain, 1) DIALOG module loadmodule dialog.so
Re: [OpenSIPS-Users] Opensips 1.10 NAT
You can set a flag on usr_preferences to force the nat to that customer so you can manage this on your dialplan if the user cannot be recognized over nat help you can force. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10 NAT
Thnk mike, an example where I can watch this? 2013/11/5 Mike Tesliuk m...@ultra.net.br You can set a flag on usr_preferences to force the nat to that customer so you can manage this on your dialplan if the user cannot be recognized over nat help you can force. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- rickygm http://gnuforever.homelinux.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10 NAT
You can use the avpops module avp_db_load($fu,$avp(678)); This will load the preferences from the user on avp(678) , so you check the value and force the use of rtpproxy/mediaproxy as you do when the user is behind proxy. you can use the avp_db_query also avp_db_query(select value from usr_preferences where username='$fu' and atrribute = 'USE_NAT', $avp(678)); in this case you should have an information for this username ($fu) using an attribute USE_NAT , you can set the value to 1 or 0 , and you do the rest on you dialplan 2013/11/5 troxlinux xserverli...@gmail.com Thnk mike, an example where I can watch this? 2013/11/5 Mike Tesliuk m...@ultra.net.br You can set a flag on usr_preferences to force the nat to that customer so you can manage this on your dialplan if the user cannot be recognized over nat help you can force. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- rickygm http://gnuforever.homelinux.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10 NAT
I'm running out of ideas .. My rtpproxy is fine Oct 4 09:10:02 opensips /sbin/opensips[5019]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5020]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5018]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5022]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5023]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5017]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5016]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled and here is my opensips.cfg, ### 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 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 /* comment the next line to enable the auto discovery of local aliases based on revers DNS on IPs */ auto_aliases=no listen=udp:###.###.###.###:5060 disable_tcp=yes db_default_url=mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips ### Modules Section #set module path mpath=/lib64/opensips/modules/ SIGNALING module loadmodule signaling.so StateLess module loadmodule sl.so Transaction Module loadmodule tm.so modparam(tm, fr_timer, 5) modparam(tm, fr_inv_timer, 30) modparam(tm, restart_fr_on_each_reply, 0) modparam(tm, onreply_avp_mode, 1) Record Route Module loadmodule rr.so /* do not append from tag to the RR (no need for this script) */ modparam(rr, append_fromtag, 0) MAX ForWarD module loadmodule maxfwd.so SIP MSG OPerationS module loadmodule sipmsgops.so FIFO Management Interface loadmodule mi_fifo.so modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0666) URI module loadmodule uri.so modparam(uri, use_uri_table, 0) MYSQL module loadmodule db_mysql.so USeR LOCation module loadmodule usrloc.so modparam(usrloc, nat_bflag, NAT) modparam(usrloc, db_mode, 2) modparam(usrloc, db_url, mysql://:@##/opensips) REGISTRAR module loadmodule registrar.so modparam(registrar, tcp_persistent_flag, TCP_PERSISTENT) modparam(registrar, received_avp, $avp(received_nh)) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam(registrar, max_contacts, 10) ACCounting module loadmodule acc.so /* what special events should be accounted ? */ modparam(acc, early_media, 0) modparam(acc, report_cancels, 0) /* by default we 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) modparam(acc, failed_transaction_flag, ACC_FAILED) /* account triggers (flags) */ modparam(acc, db_flag, ACC_DO) modparam(acc, db_missed_flag, ACC_MISSED) modparam(acc, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) AUTHentication modules loadmodule auth.so loadmodule auth_db.so modparam(auth_db, calculate_ha1, yes) modparam(auth_db, password_column, password) modparam(auth_db, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) modparam(auth_db, load_credentials, ) ALIAS module loadmodule alias_db.so modparam(alias_db, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) DOMAIN module loadmodule domain.so modparam(domain, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) modparam(domain, db_mode, 1) # Use caching modparam(auth_db|usrloc|uri, use_domain, 1) DIALOG module loadmodule dialog.so modparam(dialog, dlg_match_mode, 1) modparam(dialog, default_timeout, 21600) # 6 hours timeout modparam(dialog, db_mode, 2) modparam(dialog, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) NAT modules loadmodule nathelper.so modparam(nathelper, natping_interval, 10) modparam(nathelper, ping_nated_only, 1) modparam(nathelper, received_avp, $avp(received_nh)) modparam(nathelper, natping_socket, ###.###.###.###:5060) modparam(nathelper, sipping_from, sip:pinger@###.###.###.###) modparam(nathelper, sipping_method, OPTIONS) loadmodule rtpproxy.so modparam(rtpproxy,
Re: [OpenSIPS-Users] Opensips 1.10 NAT
I think I found the problem .. Looking at my SIP Messages, the VIA and the Contact headers doesnt have my INVALID IP, it shows me my VALID IP. But I dont know how to set that, im doing the fixes for nated contacts. Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com I'm running out of ideas .. My rtpproxy is fine Oct 4 09:10:02 opensips /sbin/opensips[5019]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5020]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5018]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5022]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5023]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5017]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled Oct 4 09:10:02 opensips /sbin/opensips[5016]: INFO:rtpproxy:rtpp_test: rtp proxy udp:127.0.0.1:7890 found, support for it enabled and here is my opensips.cfg, ### 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 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 /* comment the next line to enable the auto discovery of local aliases based on revers DNS on IPs */ auto_aliases=no listen=udp:###.###.###.###:5060 disable_tcp=yes db_default_url=mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips ### Modules Section #set module path mpath=/lib64/opensips/modules/ SIGNALING module loadmodule signaling.so StateLess module loadmodule sl.so Transaction Module loadmodule tm.so modparam(tm, fr_timer, 5) modparam(tm, fr_inv_timer, 30) modparam(tm, restart_fr_on_each_reply, 0) modparam(tm, onreply_avp_mode, 1) Record Route Module loadmodule rr.so /* do not append from tag to the RR (no need for this script) */ modparam(rr, append_fromtag, 0) MAX ForWarD module loadmodule maxfwd.so SIP MSG OPerationS module loadmodule sipmsgops.so FIFO Management Interface loadmodule mi_fifo.so modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0666) URI module loadmodule uri.so modparam(uri, use_uri_table, 0) MYSQL module loadmodule db_mysql.so USeR LOCation module loadmodule usrloc.so modparam(usrloc, nat_bflag, NAT) modparam(usrloc, db_mode, 2) modparam(usrloc, db_url, mysql://:@##/opensips) REGISTRAR module loadmodule registrar.so modparam(registrar, tcp_persistent_flag, TCP_PERSISTENT) modparam(registrar, received_avp, $avp(received_nh)) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam(registrar, max_contacts, 10) ACCounting module loadmodule acc.so /* what special events should be accounted ? */ modparam(acc, early_media, 0) modparam(acc, report_cancels, 0) /* by default we 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) modparam(acc, failed_transaction_flag, ACC_FAILED) /* account triggers (flags) */ modparam(acc, db_flag, ACC_DO) modparam(acc, db_missed_flag, ACC_MISSED) modparam(acc, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) AUTHentication modules loadmodule auth.so loadmodule auth_db.so modparam(auth_db, calculate_ha1, yes) modparam(auth_db, password_column, password) modparam(auth_db, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) modparam(auth_db, load_credentials, ) ALIAS module loadmodule alias_db.so modparam(alias_db, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) DOMAIN module loadmodule domain.so modparam(domain, db_url, mysql://opensips:opensipsrw@localhost /opensipsmysql://###:###@###/opensips) modparam(domain, db_mode, 1) # Use caching modparam(auth_db|usrloc|uri, use_domain, 1) DIALOG module loadmodule dialog.so modparam(dialog, dlg_match_mode, 1) modparam(dialog, default_timeout, 21600) # 6 hours timeout modparam(dialog, db_mode,
Re: [OpenSIPS-Users] Opensips 1.10 NAT
Yes I did Mike, and my SIP messages are ok. I will take a look at your tutorial. tks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/3 Mike Tesliuk m...@ultra.net.br Did you try to made some debug rodrigo ? maybe some rule is missing on your route script i made a tutorial over version 1.9 that you can check [portugues] http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy [english] http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English 2013/10/3 Rodrigo Ferreira rsferreir...@gmail.com Hi guys, After a long time without using Opensips (almost a year) I tried to install the opensips 1.10 and everything went well BUT when I make a call, there's no audio, I know that is something because of NAT, but I have the nathelper and rtpproxy configuration on my opensips.cfg. There's anything else that I could take a look at? Thanks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10 NAT
That howto is just a sample (with a lot of comments) to better understand of nat configuration (over my understand offcourse), so, you can check and compare with your configuration to identify about something missing 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com Yes I did Mike, and my SIP messages are ok. I will take a look at your tutorial. tks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/3 Mike Tesliuk m...@ultra.net.br Did you try to made some debug rodrigo ? maybe some rule is missing on your route script i made a tutorial over version 1.9 that you can check [portugues] http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy [english] http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English 2013/10/3 Rodrigo Ferreira rsferreir...@gmail.com Hi guys, After a long time without using Opensips (almost a year) I tried to install the opensips 1.10 and everything went well BUT when I make a call, there's no audio, I know that is something because of NAT, but I have the nathelper and rtpproxy configuration on my opensips.cfg. There's anything else that I could take a look at? Thanks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10 NAT
I did that Mike .. my nat_uac_client isnt passing in any verification ... I did this .. if ( nat_uac_test(1) ) xlog(UAC TEST = 1); if ( nat_uac_test(2) ) xlog(UAC TEST = 2); if ( nat_uac_test(4) ) xlog(UAC TEST = 4); if ( nat_uac_test(8) ) xlog(UAC TEST = 8); if ( nat_uac_test(16) ) xlog(UAC TEST = 16); if ( nat_uac_test(32) ) xlog(UAC TEST = 32); if ( nat_uac_test(64) ) xlog(UAC TEST = 64); in the beginning of the script, to see what is happening to my NAT, and i got nothing. Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/4 Mike Tesliuk m...@ultra.net.br That howto is just a sample (with a lot of comments) to better understand of nat configuration (over my understand offcourse), so, you can check and compare with your configuration to identify about something missing 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com Yes I did Mike, and my SIP messages are ok. I will take a look at your tutorial. tks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/3 Mike Tesliuk m...@ultra.net.br Did you try to made some debug rodrigo ? maybe some rule is missing on your route script i made a tutorial over version 1.9 that you can check [portugues] http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy [english] http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English 2013/10/3 Rodrigo Ferreira rsferreir...@gmail.com Hi guys, After a long time without using Opensips (almost a year) I tried to install the opensips 1.10 and everything went well BUT when I make a call, there's no audio, I know that is something because of NAT, but I have the nathelper and rtpproxy configuration on my opensips.cfg. There's anything else that I could take a look at? Thanks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10 NAT
well, probably you softphone/ip phone, is using some kind of stun or other kind of nat features, so, nothing come to be detected, this can happen, so, if you will be ever using nat, you can force the rtpproxy without nat detection, this will solve your problem, if you read the documentation ( http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html#id292854 ) you can see that this test are made over rfc1918 or different ip address from via and signalling The problem is probably the fact that when the call is stablished, the media cannot traverse, you have the correct ip information on sdp but the router does not permit the session to be opened, so, do a test forcing the use of rtpproxy without the nat detection, just force all trafic throught rtpproxy 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com I did that Mike .. my nat_uac_client isnt passing in any verification ... I did this .. if ( nat_uac_test(1) ) xlog(UAC TEST = 1); if ( nat_uac_test(2) ) xlog(UAC TEST = 2); if ( nat_uac_test(4) ) xlog(UAC TEST = 4); if ( nat_uac_test(8) ) xlog(UAC TEST = 8); if ( nat_uac_test(16) ) xlog(UAC TEST = 16); if ( nat_uac_test(32) ) xlog(UAC TEST = 32); if ( nat_uac_test(64) ) xlog(UAC TEST = 64); in the beginning of the script, to see what is happening to my NAT, and i got nothing. Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/4 Mike Tesliuk m...@ultra.net.br That howto is just a sample (with a lot of comments) to better understand of nat configuration (over my understand offcourse), so, you can check and compare with your configuration to identify about something missing 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com Yes I did Mike, and my SIP messages are ok. I will take a look at your tutorial. tks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/3 Mike Tesliuk m...@ultra.net.br Did you try to made some debug rodrigo ? maybe some rule is missing on your route script i made a tutorial over version 1.9 that you can check [portugues] http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy [english] http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English 2013/10/3 Rodrigo Ferreira rsferreir...@gmail.com Hi guys, After a long time without using Opensips (almost a year) I tried to install the opensips 1.10 and everything went well BUT when I make a call, there's no audio, I know that is something because of NAT, but I have the nathelper and rtpproxy configuration on my opensips.cfg. There's anything else that I could take a look at? Thanks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10 NAT
Forcing the traffic through RTPPROXY worked, but why isnt working the nat_uac_test? Kinda weird Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/4 Mike Tesliuk m...@ultra.net.br well, probably you softphone/ip phone, is using some kind of stun or other kind of nat features, so, nothing come to be detected, this can happen, so, if you will be ever using nat, you can force the rtpproxy without nat detection, this will solve your problem, if you read the documentation ( http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html#id292854 ) you can see that this test are made over rfc1918 or different ip address from via and signalling The problem is probably the fact that when the call is stablished, the media cannot traverse, you have the correct ip information on sdp but the router does not permit the session to be opened, so, do a test forcing the use of rtpproxy without the nat detection, just force all trafic throught rtpproxy 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com I did that Mike .. my nat_uac_client isnt passing in any verification ... I did this .. if ( nat_uac_test(1) ) xlog(UAC TEST = 1); if ( nat_uac_test(2) ) xlog(UAC TEST = 2); if ( nat_uac_test(4) ) xlog(UAC TEST = 4); if ( nat_uac_test(8) ) xlog(UAC TEST = 8); if ( nat_uac_test(16) ) xlog(UAC TEST = 16); if ( nat_uac_test(32) ) xlog(UAC TEST = 32); if ( nat_uac_test(64) ) xlog(UAC TEST = 64); in the beginning of the script, to see what is happening to my NAT, and i got nothing. Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/4 Mike Tesliuk m...@ultra.net.br That howto is just a sample (with a lot of comments) to better understand of nat configuration (over my understand offcourse), so, you can check and compare with your configuration to identify about something missing 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com Yes I did Mike, and my SIP messages are ok. I will take a look at your tutorial. tks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/3 Mike Tesliuk m...@ultra.net.br Did you try to made some debug rodrigo ? maybe some rule is missing on your route script i made a tutorial over version 1.9 that you can check [portugues] http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy [english] http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English 2013/10/3 Rodrigo Ferreira rsferreir...@gmail.com Hi guys, After a long time without using Opensips (almost a year) I tried to install the opensips 1.10 and everything went well BUT when I make a call, there's no audio, I know that is something because of NAT, but I have the nathelper and rtpproxy configuration on my opensips.cfg. There's anything else that I could take a look at? Thanks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10 NAT
probably if the UA's are on the same network, when you send the package the package is going with the external ip on the SDP , when the call is stablished probably you router is not allowing to open the second lag because the UA's are trying to stablish from inside using the outside ip addressl, so when you go through rtpproxy this not happen both sides use the opensips (rtpproxy) ip address to sdp. If my logic is not correct please somebody let me know. 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com Forcing the traffic through RTPPROXY worked, but why isnt working the nat_uac_test? Kinda weird Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/4 Mike Tesliuk m...@ultra.net.br well, probably you softphone/ip phone, is using some kind of stun or other kind of nat features, so, nothing come to be detected, this can happen, so, if you will be ever using nat, you can force the rtpproxy without nat detection, this will solve your problem, if you read the documentation ( http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html#id292854) you can see that this test are made over rfc1918 or different ip address from via and signalling The problem is probably the fact that when the call is stablished, the media cannot traverse, you have the correct ip information on sdp but the router does not permit the session to be opened, so, do a test forcing the use of rtpproxy without the nat detection, just force all trafic throught rtpproxy 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com I did that Mike .. my nat_uac_client isnt passing in any verification ... I did this .. if ( nat_uac_test(1) ) xlog(UAC TEST = 1); if ( nat_uac_test(2) ) xlog(UAC TEST = 2); if ( nat_uac_test(4) ) xlog(UAC TEST = 4); if ( nat_uac_test(8) ) xlog(UAC TEST = 8); if ( nat_uac_test(16) ) xlog(UAC TEST = 16); if ( nat_uac_test(32) ) xlog(UAC TEST = 32); if ( nat_uac_test(64) ) xlog(UAC TEST = 64); in the beginning of the script, to see what is happening to my NAT, and i got nothing. Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/4 Mike Tesliuk m...@ultra.net.br That howto is just a sample (with a lot of comments) to better understand of nat configuration (over my understand offcourse), so, you can check and compare with your configuration to identify about something missing 2013/10/4 Rodrigo Ferreira rsferreir...@gmail.com Yes I did Mike, and my SIP messages are ok. I will take a look at your tutorial. tks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 2013/10/3 Mike Tesliuk m...@ultra.net.br Did you try to made some debug rodrigo ? maybe some rule is missing on your route script i made a tutorial over version 1.9 that you can check [portugues] http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy [english] http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English 2013/10/3 Rodrigo Ferreira rsferreir...@gmail.com Hi guys, After a long time without using Opensips (almost a year) I tried to install the opensips 1.10 and everything went well BUT when I make a call, there's no audio, I know that is something because of NAT, but I have the nathelper and rtpproxy configuration on my opensips.cfg. There's anything else that I could take a look at? Thanks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips 1.10 NAT
Hi guys, After a long time without using Opensips (almost a year) I tried to install the opensips 1.10 and everything went well BUT when I make a call, there's no audio, I know that is something because of NAT, but I have the nathelper and rtpproxy configuration on my opensips.cfg. There's anything else that I could take a look at? Thanks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.10 NAT
Did you try to made some debug rodrigo ? maybe some rule is missing on your route script i made a tutorial over version 1.9 that you can check [portugues] http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy [english] http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English 2013/10/3 Rodrigo Ferreira rsferreir...@gmail.com Hi guys, After a long time without using Opensips (almost a year) I tried to install the opensips 1.10 and everything went well BUT when I make a call, there's no audio, I know that is something because of NAT, but I have the nathelper and rtpproxy configuration on my opensips.cfg. There's anything else that I could take a look at? Thanks Atenciosamente. Eng.° Rodrigo Ferreira ITIL v3 Certified http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 1.7 + NAT + rtpproxy
Have you installed and started rtpproxy? if not just scroll through this
website http://www.rtpproxy.org/.
Regards,
Qasim
On Fri, Aug 3, 2012 at 2:27 AM, Ashish Kundu kash...@gmail.com wrote:
Opensips is a great product, but I have been having problem in configuring
the nat traversal + rtpproxy with opensips and have spent about a week on
this. I am a novice in this... when opensips runs with the following
opensips.cfg relevant portions -- it raises the following rtpproxy problem:
ERROR:rtpproxy:select_rtpp_node: script error -no valid set selected
ERROR:rtpproxy:force_rtp_proxy: no available proxies
# nat_traversal params -
modparam(nat_traversal, keepalive_interval, 30)
modparam(nat_traversal, keepalive_method, OPTIONS)
modparam(nat_traversal, keepalive_from, sip:keepalive@a.b.c.d)
modparam(nat_traversal, keepalive_state_file,
/var/run/opensips/keepalive_state)
#ak# --- rtpproxy -
# single rtproxy with specific weight
modparam(rtpproxy, rtpproxy_sock, udp:localhost:)
modparam(rtpproxy, nortpproxy_str, a=sdpmangled:yes\r\n)
modparam(rtpproxy, db_url, mysql://opensips:opensipsrw@localhost
/opensips)
#modparam(rtpproxy, db_table, nh_rtpp)
modparam(rtpproxy, rtpp_socket_col, rtpproxy_sock)
### Routing Logic
# main request routing logic
route{
nat_traversal info
force_rport();
if (client_nat_test(7)) {
fix_contact();
setflag(5);
}
if ((method==REGISTER || method==SUBSCRIBE ||
(method==INVITE !has_totag()))
client_nat_test(7))
{
nat_keepalive();
}
nat_traversal info ends
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
##ak#
if ((is_method(INVITE)) has_totag()) {
#(has_body(application/sdp))) {
engage_rtp_proxy();
}
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction
fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless,
do RR for
# re-INVITEs alos, as some buggy clients
do change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(1);
} else {
/* uncomment the following lines if you want to
enable presence */
##if (is_method(SUBSCRIBE) $rd == your.server.ip.address) {
## # in-dialog subscribe requests
## route(2);
## exit;
##}
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
t_relay();
exit;
} else {
# ACK without matching transaction
-
# ignore and discard
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
#initial requests
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
#unforce_rtpproxy();
exit;
}
t_check_trans();
# preloaded route checking
if (loose_route()) {
xlog(L_ERR,
Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}
# record routing
if (!is_method(REGISTER|MESSAGE))
record_route();
# account only INVITEs
if (is_method(INVITE)) {
setflag(1); # do accounting
}
if (!uri==myself)
## replace with following line if multi-domain support is used
##if
[OpenSIPS-Users] OpenSIPS 1.7 + NAT + rtpproxy
Opensips is a great product, but I have been having problem in configuring
the nat traversal + rtpproxy with opensips and have spent about a week on
this. I am a novice in this... when opensips runs with the following
opensips.cfg relevant portions -- it raises the following rtpproxy problem:
ERROR:rtpproxy:select_rtpp_node: script error -no valid set selected
ERROR:rtpproxy:force_rtp_proxy: no available proxies
# nat_traversal params -
modparam(nat_traversal, keepalive_interval, 30)
modparam(nat_traversal, keepalive_method, OPTIONS)
modparam(nat_traversal, keepalive_from, sip:keepalive@a.b.c.d)
modparam(nat_traversal, keepalive_state_file,
/var/run/opensips/keepalive_state)
#ak# --- rtpproxy -
# single rtproxy with specific weight
modparam(rtpproxy, rtpproxy_sock, udp:localhost:)
modparam(rtpproxy, nortpproxy_str, a=sdpmangled:yes\r\n)
modparam(rtpproxy, db_url, mysql://opensips:opensipsrw@localhost
/opensips)
#modparam(rtpproxy, db_table, nh_rtpp)
modparam(rtpproxy, rtpp_socket_col, rtpproxy_sock)
### Routing Logic
# main request routing logic
route{
nat_traversal info
force_rport();
if (client_nat_test(7)) {
fix_contact();
setflag(5);
}
if ((method==REGISTER || method==SUBSCRIBE ||
(method==INVITE !has_totag())) client_nat_test(7))
{
nat_keepalive();
}
nat_traversal info ends
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
##ak#
if ((is_method(INVITE)) has_totag()) {
#(has_body(application/sdp))) {
engage_rtp_proxy();
}
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction
fails
} else if (is_method(INVITE)) {
# even if in most of the cases is useless,
do RR for
# re-INVITEs alos, as some buggy clients do
change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by
loose_route()
# in $du (destination URI).
route(1);
} else {
/* uncomment the following lines if you want to
enable presence */
##if (is_method(SUBSCRIBE) $rd == your.server.ip.address) {
## # in-dialog subscribe requests
## route(2);
## exit;
##}
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
# non loose-route, but stateful
ACK; must be an ACK after
# a 487 or e.g. 404 from upstream
server
t_relay();
exit;
} else {
# ACK without matching transaction
-
# ignore and discard
exit;
}
}
sl_send_reply(404,Not here);
}
exit;
}
#initial requests
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans())
t_relay();
#unforce_rtpproxy();
exit;
}
t_check_trans();
# preloaded route checking
if (loose_route()) {
xlog(L_ERR,
Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci]);
if (!is_method(ACK))
sl_send_reply(403,Preload Route denied);
exit;
}
# record routing
if (!is_method(REGISTER|MESSAGE))
record_route();
# account only INVITEs
if (is_method(INVITE)) {
setflag(1); # do accounting
}
if (!uri==myself)
## replace with following line if multi-domain support is used
##if (!is_uri_host_local())
{
append_hf(P-hint: outbound\r\n);
# if you have some interdomain connections via TLS
##if($rd==tls_domain1.net) {
## t_relay(tls:domain1.net);
## exit;
##} else if($rd==tls_domain2.net) {
[OpenSIPS-Users] OpenSIPS + OpenIMS + NAT issues
Hello all, I have some problems with OpenSIPS and OpenIMS due to NAT configuration. My setup is a follow: UA -- HGW (embedded both OpenSIPS and NAT stuff) -- P-CSCF (OpenIMS) I used different UA (mainly Twinkle and X-Lite). The HGW (Home GateWay) is running under OpenWRT on which I compile and install OpenSIPS. The P-CSCF, S-CSCF and I-CSCF are all running on the same PC (standard configuration from OpenIMS installation). My OpenSIPS is just used to manage local message (perform some security check) and manage the NAT configuration of the HGW. My problem come from the fact that the P-CSCF (and subsequently the S-CSCF) is registered my UA with its private @IP address and not the public @IP address of the HGW. So, each time I sent a SIP message to the IMS Core, the P-CSCF reject my messages with a 403 Forbidden. You must registered first in the P-CSCF. This come from that the P-CSCF check who is sending the SIP message based on the source @IP. In my case, the source @IP address is the public one (i.e. the HGW public one). However, this public @IP address is not know by the P-CSCF i.e. it doesn't correspond to a registered UA. So, Outgoing call are not working. Fortunately, Incoming call (i.e. from a UA which is directly connected to the IMS Core) are working well. I try several configuration using nathelper module, but I just got a negative reply from the S-CSCF instead of the P-CSCF (I.e. I pass the P-CSCF check by using force_rport in register and invite message)). I fact, the problem come from the fix_nated_contact() and fix_nated_register() function which don't do the job I want. They rewrite the contact field with the source IP and Port of the original message i.e. the @IP address and port of the UA. So, what I'm looking for, is a way to hide the private @IP address and the possibility to rewrite the Contact field with the public @IP of the HGW in order for the P-CSCF thinks that the UA is registered with the public @IP address and not the private one. Is it possible and how ? Thanks a lot for your help. Olivier PS: Here it is my opensips configuration: # --- global configuration parameters debug=3 # debug level (cmd line: -dd) log_stderror=yes # (cmd line: -E) log_facility=LOG_LOCAL1 fork=yes sip_warning=0 check_via=no# (cmd. line: -v) #dns=yes # (cmd. line: -r) dns=no # (cmd. line: -r) #rev_dns=yes # (cmd. line: -R) rev_dns=no # (cmd. line: -R) disable_tcp=yes disable_dns_blacklist=yes disable_dns_failover=yes listen=udp:192.168.1.1:5060 listen=udp:217.70.81.211:5060 children=1 auto_aliases=no alias=zpna.systerminal.eu:5060 # -- module loading -- mpath=/usr/lib/opensips/modules loadmodule db_text.so loadmodule sl.so loadmodule tm.so loadmodule rr.so loadmodule xlog.so loadmodule mi_fifo.so loadmodule maxfwd.so loadmodule uac.so loadmodule usrloc.so loadmodule registrar.so loadmodule auth.so loadmodule auth_db.so loadmodule alias_db.so loadmodule uri.so loadmodule uri_db.so loadmodule domain.so loadmodule nathelper.so loadmodule textops.so loadmodule avpops.so loadmodule permissions.so loadmodule presence.so loadmodule presence_xml.so loadmodule pua.so loadmodule rls.so loadmodule xcap_client.so # - setting module-specific parameters --- # -- multi-modules params -- modparam(usrloc|permissions|auth_db|uri_db|domain|presence|presence_xml|rls|pua|xcap_client|alias_db, db_url, text:///etc/opensips/opensipsdb) modparam(auth_db|alias_db|uri_db|usrloc, use_domain, 1) # -- mi_fifo params -- modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) modparam(mi_fifo, fifo_mode, 0666) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam(rr, enable_full_lr, 1) # -- nathelper -- modparam(nathelper, rtpproxy_sock, unix:/var/run/rtpproxy.sock) modparam(nathelper, natping_interval, 60) modparam(nathelper, ping_nated_only, 1) modparam(nathelper, received_avp, $avp(i:9)) # -- timer params -- modparam(tm, fr_timer, 5) modparam(tm, fr_inv_timer, 100) modparam(tm, wt_timer, 10) # -- usrloc params -- modparam(usrloc, db_mode, 1) modparam(usrloc, timer_interval, 10) modparam(usrloc, nat_bflag, 6) modparam(usrloc, desc_time_order, 1) # -- auth params -- modparam(auth, nonce_expire, 300) modparam(auth, realm_prefix, sip.) # modparam(auth, rpid_avp, $avp(rpid)) # -- auth_db params -- modparam(auth_db, password_column, password) modparam(auth_db, calculate_ha1, 1) # -- registrar params -- modparam(registrar, max_contacts, 2) modparam(registrar, received_avp, $avp(i:9)) modparam(registrar, sock_flag, 12) modparam(registrar, sock_hdr_name, Local-Sock) modparam(registrar, max_expires, 3600) # -- permissions params -- modparam(permissions, db_mode, 1) modparam(permissions, trusted_table, trusted) # -- presence params -- modparam(presence, server_address, sip:192.168.1.1:5060)
