Re: [OpenSIPS-Users] Load Balancer
Hi Bogdan, thanks for the reply. The issue was that the Opensips box could not contact the FS box using the FS external IP address. I have solved the issue by making some changes to the firewall. I have added a loop back (?) so that the Opensips box can contact the FS box using it's external IP address, and all works fine now. I didn't post an update just incase someone had an alternative/better solution. thanks Matt On 29 June 2015 at 10:34, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Matt, There is something bogus in your setup. If FS is advertising the public IP, why is it not able to receive traffic on that IP (the ACK from OpenSIPS). You need to make a consistent setup in there in terms of used and routable IPs. Now, the 200 OK (to INVITE) is sent by FS via the external or internal IP ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 24.06.2015 17:40, Matt Broad wrote: Hi, I have the following setup (the IP addresses are made up but should give an indication): 1 x opensips server with load balancer module - IP 192.168.0.1 2 x freeswitch servers - IP 192.168.0.2 192.168.0.3 All 3 servers have seperate external IP address routing to their internal IP via our firewall: 217.0.0.1 routed to 192.168.0.1 (Opensips) 217.0.0.2 routed to 192.168.0.2 (FS1) 217.0.0.3 routed to 192.168.0.3 (FS2) I have the load_balancer table with the following details: id, | group_id, | dst_uri,| resources, | probe_mode, | description '1', | '1', | 'sip:192.168.0.2:5080', | 'pstn=10', | '2', | 'FS1' '2', | '1', | 'sip:192.168.0.3:5080', | 'vm=1', | '2', | 'FS2' Now the initial invite goes through the usual steps, as shown below, but the issue arises when the ACK comes back from the SIP provider. Opensips is trying to forward the packet to the Freeswitch external IP address rather than the internal IP. This is due, I think, to the Contact details in the 200 response from Freeswitch containing the Freeswitch external IP. Is there a way of having Opensips take the external IP and route to the internal IP instead? Any help/suggestions would be much appreciated. No. Time Source Destination Protocol Length Info 42 5.827674 SIP Provider OpenSips Internal SIP/SDP 873 Request: INVITE sip:test@Opensips External | 43 5.828043 OpenSips Internal SIP Provider SIP 397 Status: 100 Giving a try | 44 5.828159 OpenSips Internal Freeswitch Internal SIP/SDP 1039 Request: INVITE sip:test@Opensips External | 45 5.828605 Freeswitch Internal OpenSips Internal SIP 588 Status: 100 Trying | 46 5.832171 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 54 5.833451 OpenSips Internal SIP Provider SIP/SDP 1132 Status: 200 OK | 55 5.836739 SIP Provider OpenSips Internal SIP 522 Request: ACK sip:test@Freeswitch External:5080;transport=udp | 61 5.838075 OpenSips Internal Freeswitch External SIP 545 Request: ACK sip:test@Freeswitch External:5080;transport=udp | 70 6.33254 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 71 6.33265 OpenSips Internal SIP Provider SIP/SDP 1132 Status: 200 OK | 80 7.332536 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 81 7.332666 OpenSips Internal SIP Provider SIP/SDP 1132 Status: 200 OK | 86 9.332576 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 87 9.332712 OpenSips Internal SIP Provider SIP/SDP 1132 Status: 200 OK | 136 13.331858 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 152 16.938621 SIP Provider OpenSips Internal SIP 553 Request: BYE sip:test@Freeswitch External:5080;transport=udp | 158 16.939786 OpenSips Internal Freeswitch External SIP 577 Request: BYE sip:test@Freeswitch External:5080;transport=udp | 160 17.33247 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 161 17.37482 OpenSips Internal Freeswitch External SIP 577 Request: BYE sip:test@Freeswitch External:5080;transport=udp | 163 17.937941 SIP Provider OpenSips Internal SIP 553 Request: BYE sip:test@Freeswitch External:5080;transport=udp | 166 18.375894 OpenSips Internal Freeswitch External SIP 577 Request: BYE sip:test@Freeswitch External:5080;transport=udp | 170 18.776373 OpenSips Internal SIP Provider SIP 415 Status: 408 Request Timeout | 178 21.332616 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 222 25.333161 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 257 29.333884 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 283 33.334482 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 306 37.335049 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | ___
Re: [OpenSIPS-Users] Protocol header changing with OpenSIPS
Hi, Just keep in mind OpenSIPS is taking care of SIP signaling only, it is not doing anything media/RTP related. Still you can alter the SDP (codecs, order, rtc), but not the RTP itself. For that, as Peter mentioned, you need separate tool. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 28.06.2015 13:32, H Yavari wrote: Hi, It is possible to change the protocol header of RTP before sending and change it after receiving? It is possible to compress RTP packets with OpenSIPS? Any one can give me some point about this solutions? Regards, H.Yavari ___ 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] topology_hiding function and loose_route etc
Hi John, So you have a SIP capture (full call) to show the re-INVITE problem ? Otherwise it is hard to figure out what is going on there. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 15.06.2015 22:08, John Nash wrote: Hello Bogdan, Thank you I will just ignore them. I have one more related issue. I am using uac_replace_from in auto mode along with topology_hiding. In a case when UA sends opensips a REInvite , my end carrier seem to completely ignore the Reinvite. I noticed that From URI in original Invite is different from the one sent in Reinvite (Only change is caller ID) Is there something I should know when mixing topology_hiding function and uac_replace_from? Regards John On Mon, Jun 15, 2015 at 9:57 PM, Bogdan-Andrei Iancu bog...@opensips.org mailto:bog...@opensips.org wrote: Hi John, The module complains of receiving in Early state a reply without tag param in TO header - something like that is bogus. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 15.06.2015 12:32, John Nash wrote: I have modified my proxy config to support topology_hiding function of dialog module. But I see lot of dialog related errors like .. ERROR:dialog:push_reply_in_dialog: [487] reply in dlg state [2]: missing TAG param in TO hdr ERROR:dialog:w_validate_dialog: null dialog I am just wondering if my configuration is correct. How functions like loose_route(); match_dialog();, Validate_dialog(), fix_route_dialog should be used in production environment to cover all cases. From documentation I find code snippets explaining application for each function but how they all work together in topology_hiding function scenario? PS: I can send my config in case someone needs to have a look. John ___ Users mailing list Users@lists.opensips.org mailto: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] isup strip and append
Hi, I am trying to perform a 'Transparent' Transit of ISUP Messages (rfc3398) using OpenSIPS textops module.The goal is on receiving an INVINTE with ISUP encapsulated, I need to forward/load balance it to asterisk (successfully done) and then append the ISUP data to the new call generated by asterisk (forwarded call) through OpenSIPS.Is this the correct way of doing so? If not, what would be the best method? Thanks,Andrei ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Load Balancer
Hi Matt, There is something bogus in your setup. If FS is advertising the public IP, why is it not able to receive traffic on that IP (the ACK from OpenSIPS). You need to make a consistent setup in there in terms of used and routable IPs. Now, the 200 OK (to INVITE) is sent by FS via the external or internal IP ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.06.2015 17:40, Matt Broad wrote: Hi, I have the following setup (the IP addresses are made up but should give an indication): 1 x opensips server with load balancer module - IP 192.168.0.1 2 x freeswitch servers - IP 192.168.0.2 192.168.0.3 All 3 servers have seperate external IP address routing to their internal IP via our firewall: 217.0.0.1 routed to 192.168.0.1 (Opensips) 217.0.0.2 routed to 192.168.0.2 (FS1) 217.0.0.3 routed to 192.168.0.3 (FS2) I have the load_balancer table with the following details: id, | group_id, | dst_uri, | resources, | probe_mode, | description '1', | '1', | 'sip:192.168.0.2:5080 http://192.168.0.2:5080', | 'pstn=10', | '2', | 'FS1' '2', | '1', | 'sip:192.168.0.3:5080 http://192.168.0.3:5080', | 'vm=1', | '2', | 'FS2' Now the initial invite goes through the usual steps, as shown below, but the issue arises when the ACK comes back from the SIP provider. Opensips is trying to forward the packet to the Freeswitch external IP address rather than the internal IP. This is due, I think, to the Contact details in the 200 response from Freeswitch containing the Freeswitch external IP. Is there a way of having Opensips take the external IP and route to the internal IP instead? Any help/suggestions would be much appreciated. No. Time Source Destination ProtocolLength Info 42 5.827674 SIP Provider OpenSips Internal SIP/SDP 873 Request: INVITE sip:test@Opensips External | 43 5.828043 OpenSips Internal SIP ProviderSIP 397 Status: 100 Giving a try | 44 5.828159 OpenSips Internal Freeswitch Internal SIP/SDP 1039 Request: INVITE sip:test@Opensips External | 45 5.828605 Freeswitch Internal OpenSips Internal SIP 588 Status: 100 Trying | 46 5.832171 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 54 5.833451 OpenSips Internal SIP ProviderSIP/SDP 1132Status: 200 OK | 55 5.836739 SIP Provider OpenSips Internal SIP 522 Request: ACK sip:test@Freeswitch External:5080;transport=udp | 61 5.838075 OpenSips Internal Freeswitch External SIP 545 Request: ACK sip:test@Freeswitch External:5080;transport=udp | 70 6.33254 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 71 6.33265 OpenSips Internal SIP ProviderSIP/SDP 1132Status: 200 OK | 80 7.332536 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 81 7.332666 OpenSips Internal SIP ProviderSIP/SDP 1132Status: 200 OK | 86 9.332576 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 87 9.332712 OpenSips Internal SIP ProviderSIP/SDP 1132Status: 200 OK | 136 13.331858 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 152 16.938621 SIP Provider OpenSips Internal SIP 553 Request: BYE sip:test@Freeswitch External:5080;transport=udp | 158 16.939786 OpenSips Internal Freeswitch External SIP 577 Request: BYE sip:test@Freeswitch External:5080;transport=udp | 160 17.33247 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 161 17.37482 OpenSips Internal Freeswitch External SIP 577 Request: BYE sip:test@Freeswitch External:5080;transport=udp | 163 17.937941 SIP Provider OpenSips Internal SIP 553 Request: BYE sip:test@Freeswitch External:5080;transport=udp | 166 18.375894 OpenSips Internal Freeswitch External SIP 577 Request: BYE sip:test@Freeswitch External:5080;transport=udp | 170 18.776373 OpenSips Internal SIP ProviderSIP 415 Status: 408 Request Timeout | 178 21.332616 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 222 25.333161 Freeswitch Internal OpenSips Internal SIP/SDP 1221 Status: 200 OK | 257 29.333884 Freeswitch Internal OpenSips Internal
[OpenSIPS-Users] OpenSIP 2.1 WebRTC missing Received header in usrloc
I am following this document: http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 My sipML5 client successfully register but somehow its not calling each other. I check AOS and it looks strange, where is the received: header to contact client? AOR:: 1...@sip.example.com Contact:: sip:1001@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws Q= Expires:: 191 Callid:: 589a0c22-f016-8ac6-8721-3d535c0dd836 Cseq:: 5318 User-agent:: IM-client/OMA1.0 sipML5-v1.2015.03.18 State:: CS_NEW Flags:: 0 Cflags:: Socket:: udp:182.XX.XX.164:5060 Methods:: 4294967295 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Post Dial delay issue
So I guess I am trying to get that going again. I am currently running 1.11
opensips and here is what my code:
loadmodule tm.so
modparam(tm, fr_timer, 10)
modparam(tm, fr_inv_timer, 10)
modparam(tm, restart_fr_on_each_reply, 0)
modparam(tm, onreply_avp_mode, 1)
...
route{
...
}
onreply_route {
if(t_check_status((180)|(183))) {
$T_fr_inv_timeout = 60;
}
}
So, what I am hoping to accomplish is that after we send INVITE to far end
carrier, if:
1. They do not response to INVITE within 10 second, opensips will give up
which cause Asterisk try next carrier.
2. If carrier response (e.g. 180/183) within 10 second, then we will change
the fr_inv_timeout timer to 60 seconds which we will let user wait 60
seconds until ring time out.
But it seems the above code does not work. I've try it and it keep on
cutting the call off at 10 second.
The only way I got the code to work is simply put
route{
$T_fr_inv_timeout = 60;
...
}
then the timer will change to 60 seconds. But then it is useless because I
want only change the timer to 60 seconds once I get a 180/183. The reason
is that carrier have their own LCR that have to go through which sometime
takes more then 10 seconds to get back 180/183 to us and thus kill our PDD
also. So in the case, I want to be able to skip to next carrier when PDD
gets too long.
I also tried (which I know probably will not work):
route{
if(t_check_status((180)|(183))) {
$T_fr_inv_timeout = 60;
}
...
}
which did not work either.
So it seems t_check_status(180) simply does not work? Or did I misplace
the code?
Thank you!
--
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Re: [OpenSIPS-Users] topology_hiding function and loose_route etc
Hello Bogdan, This issue is solved. It turned out to be my config issue. I was stripping some headers in initial Invite while in case of Reinvite I passed them as it is and that broke something at gateway. But I am facing a real issue which i cannot get past. I sent another mail by subject Ack without To tag. Once convenient please check that. John On Mon, Jun 29, 2015 at 3:07 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi John, So you have a SIP capture (full call) to show the re-INVITE problem ? Otherwise it is hard to figure out what is going on there. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 15.06.2015 22:08, John Nash wrote: Hello Bogdan, Thank you I will just ignore them. I have one more related issue. I am using uac_replace_from in auto mode along with topology_hiding. In a case when UA sends opensips a REInvite , my end carrier seem to completely ignore the Reinvite. I noticed that From URI in original Invite is different from the one sent in Reinvite (Only change is caller ID) Is there something I should know when mixing topology_hiding function and uac_replace_from? Regards John On Mon, Jun 15, 2015 at 9:57 PM, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi John, The module complains of receiving in Early state a reply without tag param in TO header - something like that is bogus. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 15.06.2015 12:32, John Nash wrote: I have modified my proxy config to support topology_hiding function of dialog module. But I see lot of dialog related errors like .. ERROR:dialog:push_reply_in_dialog: [487] reply in dlg state [2]: missing TAG param in TO hdr ERROR:dialog:w_validate_dialog: null dialog I am just wondering if my configuration is correct. How functions like loose_route(); match_dialog();, Validate_dialog(), fix_route_dialog should be used in production environment to cover all cases. From documentation I find code snippets explaining application for each function but how they all work together in topology_hiding function scenario? PS: I can send my config in case someone needs to have a look. John ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
