[OpenSIPS-Users] Basic doubt of sip routing
Hello community, I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a domain that he doesn't now? i mean...user A is registered in proxy AA, if A wants to call to another user in another domain (not registered in the Proxy AA) how does this proxy should handle the call? how does he now where to send this call? Any clarification would be appreciated :) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Basic doubt of sip routing
The proxy is using DNS to lookup the destination server. Google for RFC 3263 Adrian On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote: Hello community, I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a domain that he doesn't now? i mean...user A is registered in proxy AA, if A wants to call to another user in another domain (not registered in the Proxy AA) how does this proxy should handle the call? how does he now where to send this call? Any clarification would be appreciated :) ___ 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] Change in SDP/RTP routing with Opensips 1.4 and 1.6
Good to hear that! Cheers, Henk On 11-02-11 02:15, Chris Stone wrote: Well, looks like it WAS the ip_nat_sip and related kernel modules, but not just on the Opensips server, also on the Asterisk server. I unloaded all of the modules on the backend Asterisk server too and tried a test call again and this time it worked just fine. Appreciate all the help with this Henk and Ovidiu! Chris ___ 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] CANCELs with no transaction
Hi All,
Need help with a nagging issue:
UA-Opensips 1-Opensips 2-PSTN
UA sends an invite on Opensips 1, and is routed via do_routing() to Opensips
2, Opensips 2 uses do_routing to get to the PSTN, call starts ringing.
UA cancels call before answer, but now t_check_trans fails and the CANCEL is
not passed onto the PSTN, with the result that the call rings forever and
can only be terminated by the remote answering and dropping the call or
through a timeout.
The scripts on Opensips 1 Opensips 2 is virtuall identical:
How do I get the CANCEL to the PSTN ?
route{
.
if (is_method(CANCEL) ) {
route(5); # drop media proxy
if (t_check_trans()){ # this always fails after a do_routing()
xlog(L_INFO,CANCEL
Transaction[$fd/$fu/$rd/$ru/$si/]\n);
t_relay();
exit;
};
exit;
}
}
route[4] {
xlog(L_INFO,Route4 [$fd/$fu/$rd/$ru/$si/]\n);
$avp(i:102)=1; # Default dr-group
route(10); # Do custom stuff
t_on_failure(4);
if (do_routing($avp(i:102))){
xlog(L_INFO,Route4 Route to Dyna Group:
$avp(i:102)[$fd/$fu/$rd/$ru/$si/]\n);
t_newtran();
route(1);
exit;
};
xlog(L_INFO,Route4 No Route to Host[$fd/$fu/$rd/$ru/$si/]\n);
sl_reply_error();
exit;
}
Regards
http://www.greydotelecom.net/bcard/jnysschen.htm Juri Nysschen
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Re: [OpenSIPS-Users] Merged Request
You're right my friend...the problem is with the softphones and the OS of the machine...it's ok now! Thanks! 2011/2/10 Anca Vamanu a...@opensips.org Hi Toyima, That when you un-Register and the phone sends expires=0 you get that reply with contact and expires is correct, because of what you already had in database. There were two contacts and only one was deleted. In the reply all the registered contacts are retrieved. The question is how you got with the two contacts - and probably it was because you closed the client and did not unregister. You can test this by looking that there is no contact, open up the client and close it. Also run a message trace from the beginning to see clearly what the client sends. Regards, -- Anca Vamanu OpenSIPS Developer On 02/09/2011 12:30 PM, Toyima Dias wrote: I've seen something interesting here, When the zoiper softphone send the REGISTER with expires=0 (normal behavior as i'm restarting the phone), Opensips answers with the following: (take a look at the trace) # U 2011/02/09 12:22:19.307852 172.30.140.47:5060 - 172.30.140.57:5060 REGISTER sip:172.30.140.57;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.30.140.47:5060 ;branch=z9hG4bK-d8754z-15075d40ba64fa00-1---d8754z- Max-Forwards: 70 Contact: sip:1001@172.30.140.47:5060;rinstance=bd724156614686a6;transport=UDP;* expires=0 *To: 1001sip:1001@172.30.140.57 sip%3A1001@172.30.140.57 ;transport=UDP From: 1001sip:1001@172.30.140.57 sip%3A1001@172.30.140.57 ;transport=UDP;tag=8e1f5910 Call-ID: NTAwOGYyNDVhNzY4NjNhMjY0NTZlNTcwN2VjN2RhYWM. CSeq: 4 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper rev.5324 Authorization: Digest username=1001,realm=172.30.140.57,nonce=4d527909001c0a15fa778702ef3d4d1139dfda7a275e,uri= sip:172.30.140.57;transport=UDP ,response=859420dd3fefcccbf4d2727ff4db611d,algorithm=MD5 Allow-Events: presence Content-Length: 0 # U 2011/02/09 12:22:19.308281 172.30.140.57:5060 - 172.30.140.47:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.140.47:5060 ;branch=z9hG4bK-d8754z-15075d40ba64fa00-1---d8754z- To: 1001sip:1001@172.30.140.57 sip%3A1001@172.30.140.57 ;transport=UDP;tag=c1aca2eceea8b9ed63a816bcd8cf10b1.e871 From: 1001sip:1001@172.30.140.57 sip%3A1001@172.30.140.57 ;transport=UDP;tag=8e1f5910 Call-ID: NTAwOGYyNDVhNzY4NjNhMjY0NTZlNTcwN2VjN2RhYWM. CSeq: 4 REGISTER Contact: sip:1001@172.30.140.47:5060;rinstance=a03689bfb7cef683;transport=UDP;* expires=2149 *Server: OpenSIPS (1.6.3-notls (i386/linux)) Content-Length: 0 As you can see, Opensips answers with a expires of 2149; that's why opensips keep this registration untill the expiration time reachs 0, any ideas why opensips answer with this value? Thanks! 2011/2/9 Toyima Dias toyim...@gmail.com Hello, I've a doubt about a little problem in my opensips server, right now i just have 2 softphones registered with my opensips server, every time i restart the phones opensips creates the following: OpenSIPS:/usr/src/opensips-1.6.3-tls#opensipsctl ul show Domain:: location table=512 records=2 AOR:: 1000 Contact:: sip:1000@172.30.140.47:26612;rinstance=4975490f64787658 Q= Expires:: 3565 Callid:: OWE3Y2NmYTI3MWNjNzRjOTkxNDU0YTQ1ZTMxM2RhNTU. Cseq:: 2 User-agent:: X-Lite release 1011s stamp 41150 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:172.30.140.57:5060 Methods:: 5951 AOR:: 1001 Contact:: sip:1001@172.30.140.47:5060;rinstance=a03689bfb7cef683;transport=UDP Q= Expires:: 2641 Callid:: NGQ4ZjBmMDhiMGIyNDc5MTA5NmExMDE1YzFhZjFlMjg. Cseq:: 2 User-agent:: Zoiper rev.5324 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:172.30.140.57:5060 Methods:: 5951 Contact:: sip:1001@172.30.140.47:5060;rinstance=333586886a503975;transport=UDP Q= Expires:: 3597 Callid:: NTI0Y2VlYWJjNzI3NDEyMjkzYTNkZTYzMTdhMGEwYmY. Cseq:: 2 User-agent:: Zoiper rev.5324 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:172.30.140.57:5060 Methods:: 5951 As you can see, user 1001 has created 2 registrations (don't now why, it should sent an expires=0 when the softphone was restarted...it might be a problem of the zoiper softphone?); the problem is when 1000 calls 1001, Opensips send INVITE to both registrations of
Re: [OpenSIPS-Users] Basic doubt of sip routing
Thanks Adrian...reading the RFC3263! Thanks! 2011/2/11 Adrian Georgescu a...@ag-projects.com The proxy is using DNS to lookup the destination server. Google for RFC 3263 Adrian On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote: Hello community, I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a domain that he doesn't now? i mean...user A is registered in proxy AA, if A wants to call to another user in another domain (not registered in the Proxy AA) how does this proxy should handle the call? how does he now where to send this call? Any clarification would be appreciated :) ___ 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] Basic doubt of sip routing
Adrian, i'm checking the rfc...but even i have a question...when UA sends an INVITE to it's proxy to a phone for example (obviously not registered on the proxy), the proxy will check the RURI of this invite and it will se the following: user A sends the invite to its proxy : INVITE sip:264512380973@172.30.140.57sip%3A264512380973@172.30.140.57;transport=UDP SIP/2.0 (172.30.140.57 is the IP of proxy A) Where does the DNS takes part? the domain is it's ip address...i'm quite confuse, any help would b e appreciated Thanks 2011/2/11 Toyima Dias toyim...@gmail.com Thanks Adrian...reading the RFC3263! Thanks! 2011/2/11 Adrian Georgescu a...@ag-projects.com The proxy is using DNS to lookup the destination server. Google for RFC 3263 Adrian On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote: Hello community, I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a domain that he doesn't now? i mean...user A is registered in proxy AA, if A wants to call to another user in another domain (not registered in the Proxy AA) how does this proxy should handle the call? how does he now where to send this call? Any clarification would be appreciated :) ___ 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] Basic doubt of sip routing
The SIP proxy lookups up the domain part, what appears after the @ sign before any parameters separated by ; if is an IP address like in your example you do not perform a DNS lookup you just send the packet there. In the request URI you must put the address of the remote end, not your own address. In your example user A calls user 264512380973 on the same Proxy and not a remote one. See the illustrated examples from: http://www.tech-invite.com/Ti-sip-ex3261.html Adrian On Feb 11, 2011, at 1:17 PM, Toyima Dias wrote: Adrian, i'm checking the rfc...but even i have a question...when UA sends an INVITE to it's proxy to a phone for example (obviously not registered on the proxy), the proxy will check the RURI of this invite and it will se the following: user A sends the invite to its proxy : INVITE sip:264512380973@172.30.140.57;transport=UDP SIP/2.0 (172.30.140.57 is the IP of proxy A) Where does the DNS takes part? the domain is it's ip address...i'm quite confuse, any help would b e appreciated Thanks 2011/2/11 Toyima Dias toyim...@gmail.com Thanks Adrian...reading the RFC3263! Thanks! 2011/2/11 Adrian Georgescu a...@ag-projects.com The proxy is using DNS to lookup the destination server. Google for RFC 3263 Adrian On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote: Hello community, I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a domain that he doesn't now? i mean...user A is registered in proxy AA, if A wants to call to another user in another domain (not registered in the Proxy AA) how does this proxy should handle the call? how does he now where to send this call? Any clarification would be appreciated :) ___ 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] Basic doubt of sip routing
Thanks Adrian... So...how does ALice now that bob is in the biloxi.com domain? per the rfc 3263 section 4 (client usage) the ua must use DNS to determine where to send a call...but i have a softphone righ now, and i'm trying to make a call like this: 234...@proxy2.com (inserted by me), not just puting the numbermaybe i'm saying something crazy, but i'm confuse! 2011/2/11 Adrian Georgescu a...@ag-projects.com The SIP proxy lookups up the domain part, what appears after the @ sign before any parameters separated by ; if is an IP address like in your example you do not perform a DNS lookup you just send the packet there. In the request URI you must put the address of the remote end, not your own address. In your example user A calls user 264512380973sip%3A264512380973@172.30.140.57 on the same Proxy and not a remote one. See the illustrated examples from: http://www.tech-invite.com/Ti-sip-ex3261.html Adrian On Feb 11, 2011, at 1:17 PM, Toyima Dias wrote: Adrian, i'm checking the rfc...but even i have a question...when UA sends an INVITE to it's proxy to a phone for example (obviously not registered on the proxy), the proxy will check the RURI of this invite and it will se the following: user A sends the invite to its proxy : INVITE sip:264512380973@172.30.140.57 sip%3A264512380973@172.30.140.57;transport=UDP SIP/2.0 (172.30.140.57 is the IP of proxy A) Where does the DNS takes part? the domain is it's ip address...i'm quite confuse, any help would b e appreciated Thanks 2011/2/11 Toyima Dias toyim...@gmail.com Thanks Adrian...reading the RFC3263! Thanks! 2011/2/11 Adrian Georgescu a...@ag-projects.com The proxy is using DNS to lookup the destination server. Google for RFC 3263 Adrian On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote: Hello community, I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a domain that he doesn't now? i mean...user A is registered in proxy AA, if A wants to call to another user in another domain (not registered in the Proxy AA) how does this proxy should handle the call? how does he now where to send this call? Any clarification would be appreciated :) ___ 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] Basic doubt of sip routing
SIP routing works exactly like email. How did you know to email this list? Adrian On Feb 11, 2011, at 1:42 PM, Toyima Dias wrote: Thanks Adrian... So...how does ALice now that bob is in the biloxi.com domain? per the rfc 3263 section 4 (client usage) the ua must use DNS to determine where to send a call...but i have a softphone righ now, and i'm trying to make a call like this: 234...@proxy2.com (inserted by me), not just puting the numbermaybe i'm saying something crazy, but i'm confuse! 2011/2/11 Adrian Georgescu a...@ag-projects.com The SIP proxy lookups up the domain part, what appears after the @ sign before any parameters separated by ; if is an IP address like in your example you do not perform a DNS lookup you just send the packet there. In the request URI you must put the address of the remote end, not your own address. In your example user A calls user 264512380973 on the same Proxy and not a remote one. See the illustrated examples from: http://www.tech-invite.com/Ti-sip-ex3261.html Adrian On Feb 11, 2011, at 1:17 PM, Toyima Dias wrote: Adrian, i'm checking the rfc...but even i have a question...when UA sends an INVITE to it's proxy to a phone for example (obviously not registered on the proxy), the proxy will check the RURI of this invite and it will se the following: user A sends the invite to its proxy : INVITE sip:264512380973@172.30.140.57;transport=UDP SIP/2.0 (172.30.140.57 is the IP of proxy A) Where does the DNS takes part? the domain is it's ip address...i'm quite confuse, any help would b e appreciated Thanks 2011/2/11 Toyima Dias toyim...@gmail.com Thanks Adrian...reading the RFC3263! Thanks! 2011/2/11 Adrian Georgescu a...@ag-projects.com The proxy is using DNS to lookup the destination server. Google for RFC 3263 Adrian On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote: Hello community, I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a domain that he doesn't now? i mean...user A is registered in proxy AA, if A wants to call to another user in another domain (not registered in the Proxy AA) how does this proxy should handle the call? how does he now where to send this call? Any clarification would be appreciated :) ___ 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] Basic doubt of sip routing
create...got it...that means that if i have a phone registered in proxy A, and i want to call userB, A has no idea where B resides, at all...how does A know the domain of B? he must put in the RURI of the invite userB@domain_of_b, right? how does A knows the domain of B? does A must press in the phone: usearB(a number)@DOMAIN_OF_B? sorry for many questions, keep reading the rfc... 2011/2/11 Adrian Georgescu a...@ag-projects.com SIP routing works exactly like email. How did you know to email this list? Adrian On Feb 11, 2011, at 1:42 PM, Toyima Dias wrote: Thanks Adrian... So...how does ALice now that bob is in the biloxi.com domain? per the rfc 3263 section 4 (client usage) the ua must use DNS to determine where to send a call...but i have a softphone righ now, and i'm trying to make a call like this: 234...@proxy2.com (inserted by me), not just puting the numbermaybe i'm saying something crazy, but i'm confuse! 2011/2/11 Adrian Georgescu a...@ag-projects.com The SIP proxy lookups up the domain part, what appears after the @ sign before any parameters separated by ; if is an IP address like in your example you do not perform a DNS lookup you just send the packet there. In the request URI you must put the address of the remote end, not your own address. In your example user A calls user 264512380973sip%3A264512380973@172.30.140.57 on the same Proxy and not a remote one. See the illustrated examples from: http://www.tech-invite.com/Ti-sip-ex3261.html Adrian On Feb 11, 2011, at 1:17 PM, Toyima Dias wrote: Adrian, i'm checking the rfc...but even i have a question...when UA sends an INVITE to it's proxy to a phone for example (obviously not registered on the proxy), the proxy will check the RURI of this invite and it will se the following: user A sends the invite to its proxy : INVITE sip:264512380973@172.30.140.57 sip%3A264512380973@172.30.140.57;transport=UDP SIP/2.0 (172.30.140.57 is the IP of proxy A) Where does the DNS takes part? the domain is it's ip address...i'm quite confuse, any help would b e appreciated Thanks 2011/2/11 Toyima Dias toyim...@gmail.com Thanks Adrian...reading the RFC3263! Thanks! 2011/2/11 Adrian Georgescu a...@ag-projects.com The proxy is using DNS to lookup the destination server. Google for RFC 3263 Adrian On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote: Hello community, I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a domain that he doesn't now? i mean...user A is registered in proxy AA, if A wants to call to another user in another domain (not registered in the Proxy AA) how does this proxy should handle the call? how does he now where to send this call? Any clarification would be appreciated :) ___ 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] Basic doubt of sip routing
COOL Adrian...many thanks for your kindly answers...by the way, i've checked on the rfc that the client must use NAPTR and SRV to resolve domains! 2011/2/11 Adrian Georgescu a...@ag-projects.com If your SIP device support dialing only phone numbers, you need a translation mechanism, this you can implement in the SIP proxy. You can use standard ENUM (http://www.faqs.org/rfc/rfc3764.txt), local database lookups, configuration logic to translate the number into a fully qualified SIP address or many other methods, there are plenty of OpenSIPS modules that do such phone number translation. Adrian On Feb 11, 2011, at 2:04 PM, Toyima Dias wrote: create...got it...that means that if i have a phone registered in proxy A, and i want to call userB, A has no idea where B resides, at all...how does A know the domain of B? he must put in the RURI of the invite userB@domain_of_b, right? how does A knows the domain of B? does A must press in the phone: usearB(a number)@DOMAIN_OF_B? sorry for many questions, keep reading the rfc... 2011/2/11 Adrian Georgescu a...@ag-projects.com SIP routing works exactly like email. How did you know to email this list? Adrian On Feb 11, 2011, at 1:42 PM, Toyima Dias wrote: Thanks Adrian... So...how does ALice now that bob is in the biloxi.com domain? per the rfc 3263 section 4 (client usage) the ua must use DNS to determine where to send a call...but i have a softphone righ now, and i'm trying to make a call like this: 234...@proxy2.com (inserted by me), not just puting the numbermaybe i'm saying something crazy, but i'm confuse! 2011/2/11 Adrian Georgescu a...@ag-projects.com The SIP proxy lookups up the domain part, what appears after the @ sign before any parameters separated by ; if is an IP address like in your example you do not perform a DNS lookup you just send the packet there. In the request URI you must put the address of the remote end, not your own address. In your example user A calls user 264512380973sip%3A264512380973@172.30.140.57 on the same Proxy and not a remote one. See the illustrated examples from: http://www.tech-invite.com/Ti-sip-ex3261.html Adrian On Feb 11, 2011, at 1:17 PM, Toyima Dias wrote: Adrian, i'm checking the rfc...but even i have a question...when UA sends an INVITE to it's proxy to a phone for example (obviously not registered on the proxy), the proxy will check the RURI of this invite and it will se the following: user A sends the invite to its proxy : INVITE sip:264512380973@172.30.140.57 sip%3A264512380973@172.30.140.57;transport=UDP SIP/2.0 (172.30.140.57 is the IP of proxy A) Where does the DNS takes part? the domain is it's ip address...i'm quite confuse, any help would b e appreciated Thanks 2011/2/11 Toyima Dias toyim...@gmail.com Thanks Adrian...reading the RFC3263! Thanks! 2011/2/11 Adrian Georgescu a...@ag-projects.com The proxy is using DNS to lookup the destination server. Google for RFC 3263 Adrian On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote: Hello community, I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call for a domain that he doesn't now? i mean...user A is registered in proxy AA, if A wants to call to another user in another domain (not registered in the Proxy AA) how does this proxy should handle the call? how does he now where to send this call? Any clarification would be appreciated :) ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
Ok guys,
Few issues still (after updating from trunk).
As suggested, I removed the engage_rtp_proxy from the b2b opensips
instance.
I noticed the following debug from the opensips:
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360 - invalid query,
execution aborted
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: PGRES_FATAL_ERROR
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: ERROR: null value in
column e3_sid violates not-null constraint#012
Looking on the postgres log, here is the failed SQL statement:
2011-02-11 12:49:06 UTC ERROR: null value in column e3_sid violates
not-null constraint
2011-02-11 12:49:06 UTC STATEMENT: insert into b2b_logic
(si_key,scenario,sparam0,sparam1,sparam2,sparam3,sparam4,sdp,sstate,next_sstate,e1_type,e1_sid,e1_to,e1_from,e1_key,e2_type,e2_sid,e2_to,e2_from,e2_key
) values ('545.0','','','','','','','',-3,0,0,'','
sip:17864776626@190.124.220.12:5060','sip:359883327749@69.25.128.234
','B2B.608.661',1,'','sip:17864776626@190.124.220.12:5060','
sip:359883327749@69.25.128.234','B2B.545.4207959')
I am using the default b2b postgres tables.
So next, I have the following config on the rtpproxy opensips (not the b2b
one):
#
*route[1] {
fix_nated_contact();
if (is_method(INVITE)) {
rewritehostport(184.106.168.144:5061);
if (rtpproxy_offer(eo,184.106.168.144))
t_on_reply(1);
}
else if (method == BYE || method == CANCEL) {
unforce_rtp_proxy();
}
..
}
onreply_route[1] {
if (!(status=~183 || status=~200)) {
drop;
}
rtpproxy_answer(FA);
}*
#
As result, when I initiate a call, I get the following on the syslog:
Feb 11 12:52:48 sms /root/opensips-1.6.4-tls/opensips[21754]:
INFO:nathelper:rtpp_test: rtp proxy udp:184.106.168.144:22332 found,
support for it enabled
Feb 11 12:52:48 sms /root/opensips-1.6.4-tls/opensips[21753]:
INFO:nathelper:rtpp_test: rtp proxy udp:184.106.168.144:22332 found,
support for it enabled
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21746]:
DBG:nathelper:force_rtp_proxy: Forcing body:#012[v=0#015#012o=-
229796569696953 1 IN IP4 190.124.220.12#015#012s=-#015#012c=IN IP4
190.124.220.12
#015#012t=0 0#015#012m=audio 18338 RTP/AVP 0 101#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16]
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21746]:
DBG:core:parse_to: display={011359883327749}, ruri={
sip:359883327749@69.25.128.233}
Feb 11 12:53:05 sms rtpproxy[21731]: DBUG:handle_command: received command
21746_6 LA 4512c49c3cd0db1b410744fe0ced15bf@69.25.128.233 190.124.220.12
18338 as612bc040;1 B2B.599.537;1
Feb 11 12:53:05 sms kernel: [7145167.526106] rtpproxy[21731]: segfault at 0
ip 004053e9 sp 7fff71948b00 error 4 in rtpproxy[40+e000]
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: hash 23820 label 1987919557 branch 0
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: REF_UNSAFE:[0x7fc0f89b4f10] after is 2
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: reply matched (T=0x7fc0f89b4f10)!
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_check: end=0x7fc0f89b4f10
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1)
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: timeout waiting reply from a RTP proxy
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: proxy udp:184.106.168.144:22332 does
not respond, disable it
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: can't send command to a RTP proxy
Connection refused
repeating over 100
times
Obviously the RTPproxy dies.
What I noticed is, when i remove
*rtpproxy_answer(FA);*
from the onreply_route, the RTPproxy does not dies.
Any ideas what I am doing wrong ?
Thank you.
-- Kamen
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[OpenSIPS-Users] FW: CANCELs with no transaction
Hi All,
Need help with a nagging issue:
UA-Opensips 1-Opensips 2-PSTN
UA sends an invite on Opensips 1, and is routed via do_routing() to Opensips
2, Opensips 2 uses do_routing to get to the PSTN, call starts ringing.
UA cancels call before answer, but now t_check_trans fails and the CANCEL is
not passed onto the PSTN, with the result that the call rings forever and
can only be terminated by the remote answering and dropping the call or
through a timeout.
The scripts on Opensips 1 Opensips 2 is virtuall identical:
How do I get the CANCEL to the PSTN ?
route{
.
if (is_method(CANCEL) ) {
route(5); # drop media proxy
if (t_check_trans()){ # this always fails after a do_routing()
xlog(L_INFO,CANCEL
Transaction[$fd/$fu/$rd/$ru/$si/]\n);
t_relay();
exit;
};
exit;
}
}
route[4] {
xlog(L_INFO,Route4 [$fd/$fu/$rd/$ru/$si/]\n);
$avp(i:102)=1; # Default dr-group
route(10); # Do custom stuff
t_on_failure(4);
if (do_routing($avp(i:102))){
xlog(L_INFO,Route4 Route to Dyna Group:
$avp(i:102)[$fd/$fu/$rd/$ru/$si/]\n);
t_newtran();
route(1);
exit;
};
xlog(L_INFO,Route4 No Route to Host[$fd/$fu/$rd/$ru/$si/]\n);
sl_reply_error();
exit;
}
Regards
Juri Nysschen http://www.greydotelecom.net/bcard/jnysschen.htm
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Re: [OpenSIPS-Users] OpenSIPS handling B2B features
It's very simple setup a Conference server using OpenSIPS and Asterisk. So use asterisk. Regards, s Il 27/01/2011 17:39, Anca Vamanu ha scritto: Toyima, I am sorry, I don't have experience in setting up conference systems, so I can not make a recommendation. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS no presentity entry in Database
Hi, i´m new to the Opensips community. I started a few days ago and i´m now at the point to post my first question, because i cant fiddle out my mistake in configuration. This is what i´ve done so far. - Setting up 3 Virtual Machines (1x Debian Lenny Server, 2x Windows XP Host with X-Lite Clients) [The VM´s can communicate with each other] - Download Opensips 1.6.4-2-notls - compiled and installed opensips (works as should i think) (means: i cant see error messages in syslog, startup and restart don´t show errors or warnings) - edited the opensips.cfg file to enable mysql support and presence Here´s my problem: I can call from one X-Lite Client to the other one (works in both directions), but the problem is i can´t see the other users online state (presence). i´m using x-lite 4 and right beside the contact it writes waiting for response - but it doesn´t change. i searched all over the net - but i can´t find a solution for my problem. i think that the opensips.cfg file is correct (with the -c option i don´t get errors) and the calling works fine. I would kindly ask for your help - if you tell me what information or file you need exactly i´ll post it right here. Thanks in advance - greetings from vienna, Mario -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-no-presentity-entry-in-Database-tp5979212p5979212.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MySQL tables using the opensipsdbctl shell script
What are you trying to do ? On Wed, Feb 2, 2011 at 1:28 PM, Robin Malhotra rocky...@gmail.com wrote: Step 3: Create MySQL tables using the opensipsdbctl shell script. The syntax for this utility follows: opensipsdbctl create db name or db_path, optional I'm getting the following error for the above syntax bash: syntax error near unexpected token `newline' what's wrong here ? might be silly question ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Venkatesh ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS 1.6 on Ubuntu
Will do Dave - thanks for following up! Sent from my iPhone 4 On Feb 4, 2011, at 15:57, Dave Singer dave.sin...@wideideas.com wrote: Tyler, Just went through the OpenSIPS default script webminar = http://www.opensips.org/html/docs/video/webinar005/ And while the audio at the beginning is bad (and very end), it is only just a little bit and it is because it was coming through a bad connection to the seminar where the webinar was recorded. If there truely is a problem with some of them try downloading them instead of using the browser streaming. Also list which one(s) you have trouble with. Dave On Thu, Feb 3, 2011 at 4:01 PM, Tyler Merritt ty...@fonality.com wrote: Dave, The audio on some of the webinars that I have watched has been almost unintelligible :( I like webinars - I present many of them in my work for our customers, but I couldn't really hear well. I can't attend the live webinars as I'm in Tokyo - they happen at like 3 am. Anyway to clean up the audio? Bogdan - can I send you a mic better mic :) Tyler Merritt. Sales Engineer. Contact: tmerr...@fonality.com | 310.861.4300 x 8850 | fonality.com | SE Blog ___ 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 1.6 on Ubuntu
Guys I a newbie to OpenSIPS I have installed opensips and mysql on ubuntu following some instructions. I have also installed x-lite. Now how to register a user in opensips and to use it with the client ? I am stuck, please let me know Regards Ricky ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS starting Error
When I enter opensipsctl start INFO: Starting OpenSIPS : ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start failed I checked /var/log/messages and got following Feb 4 18:02:23 ubuntu kernel: [ 6747.275248] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:28 ubuntu kernel: [ 6752.266702] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:33 ubuntu kernel: [ 6757.257700] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:38 ubuntu kernel: [ 6762.248970] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:43 ubuntu kernel: [ 6767.240168] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:48 ubuntu kernel: [ 6772.231405] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:53 ubuntu kernel: [ 6777.222692] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:58 ubuntu kernel: [ 6782.213859] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:03 ubuntu kernel: [ 6787.205741] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:08 ubuntu kernel: [ 6792.196974] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:13 ubuntu kernel: [ 6797.188208] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:18 ubuntu kernel: [ 6802.182420] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:23 ubuntu kernel: [ 6807.170216] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:28 ubuntu kernel: [ 6812.161873] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:33 ubuntu kernel: [ 6817.153087] intel ips :00:1f.6: CPU power or thermal limit exceeded -- Venkatesh ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RE-INVITEs being sent to original contact doesn't properly adjust RTP ports on transfer?
Just an update on this: it's ridiculously hard. We've done some major surgery on the route logic, and at this point I have the strange condition where opensips seems to be sending multiple ACKs to the carrier on a single reINVITE. The carrier should be sending us two invites - one for each leg of the call (because we are transferring the call into a DID we own). I am tcpdumping the packets and we have tons of these ACKs flying all directions. I still have to make a mod based on the to domain of the first ACK, but I don't think that is going to clear everything up all at once. Why would we generate multiple ACKs? Some loop in my routing logic? Sent from my iPhone 4 On Feb 4, 2011, at 22:34, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Tyler, So ngrep-ing on proxy, you do not see the second re-INVITE (which leads to one way audio)A possibility is that the re-INVITE may by-pass your opensips. Do you do record-routing also for sequential requests ? There are some bogus UAC/UAS that continuously update the route set, even after the dialog was setup. So maybe the first re-InVITE works ok as you correctly do RR for initial INVITE, but second re-INVITE fails because UAC/UAS expect RR on first re-INVITE too Just a supposition Regards, Bogdan Tyler Merritt wrote: Hi, We've got three parties for this example: A, B, C A - Asterisk end-point Polycom B - Asterisk end-point Polycom C - Outside end-point Uniden OpenSIPs sits in front of the Asterisk servers and communicates with a carrier C5 switch directly (same local area network inside a lab facility) A calls C - call completes - talk, no issues. C presses the transfer button, which is a flash-hook putting A on hold. C dials B. B answers the call - both parties talk. C presses the flash-hook button again in order to complete the transfer. A can hear B - B cannot hear A. The RTP debug from Asterisk shows that RTP packets from B are still going to C. B didn't get the RE-INVITE apparently - but I cannot figure out where the packet is. It's not showing up in OpenSIPs sip_trace, and it's definitely not getting to Asterisk. I don't have control of the Carrier-side C5 to check, and they have been slow to provide me with a wireshark trace. Is there anything else I could do in OpenSIPs to determine if the RE-INVITE is not being handled properly besides what I've already mentioned? Thanks in advance. Tyler -- Bogdan-Andrei Iancu OpenSIPS Event - expo, conf, social, bootcamp 2 - 4 February 2011, ITExpo, Miami, USA OpenSIPS solutions and know-how ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
Please get a gdb trace from the core file.
Thanks,
Ovidiu
On Fri, Feb 11, 2011 at 8:31 AM, Kamen Petrov kamen.pet...@gmail.com wrote:
Ok guys,
Few issues still (after updating from trunk).
As suggested, I removed the engage_rtp_proxy from the b2b opensips
instance.
I noticed the following debug from the opensips:
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360 - invalid query,
execution aborted
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: PGRES_FATAL_ERROR
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: ERROR: null value in
column e3_sid violates not-null constraint#012
Looking on the postgres log, here is the failed SQL statement:
2011-02-11 12:49:06 UTC ERROR: null value in column e3_sid violates
not-null constraint
2011-02-11 12:49:06 UTC STATEMENT: insert into b2b_logic
(si_key,scenario,sparam0,sparam1,sparam2,sparam3,sparam4,sdp,sstate,next_sstate,e1_type,e1_sid,e1_to,e1_from,e1_key,e2_type,e2_sid,e2_to,e2_from,e2_key
) values
('545.0','','','','','','','',-3,0,0,'','sip:17864776626@190.124.220.12:5060','sip:359883327749@69.25.128.234','B2B.608.661',1,'','sip:17864776626@190.124.220.12:5060','sip:359883327749@69.25.128.234','B2B.545.4207959')
I am using the default b2b postgres tables.
So next, I have the following config on the rtpproxy opensips (not the b2b
one):
#
route[1] {
fix_nated_contact();
if (is_method(INVITE)) {
rewritehostport(184.106.168.144:5061);
if (rtpproxy_offer(eo,184.106.168.144))
t_on_reply(1);
}
else if (method == BYE || method == CANCEL) {
unforce_rtp_proxy();
}
..
}
onreply_route[1] {
if (!(status=~183 || status=~200)) {
drop;
}
rtpproxy_answer(FA);
}
#
As result, when I initiate a call, I get the following on the syslog:
Feb 11 12:52:48 sms /root/opensips-1.6.4-tls/opensips[21754]:
INFO:nathelper:rtpp_test: rtp proxy udp:184.106.168.144:22332 found,
support for it enabled
Feb 11 12:52:48 sms /root/opensips-1.6.4-tls/opensips[21753]:
INFO:nathelper:rtpp_test: rtp proxy udp:184.106.168.144:22332 found,
support for it enabled
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21746]:
DBG:nathelper:force_rtp_proxy: Forcing body:#012[v=0#015#012o=-
229796569696953 1 IN IP4 190.124.220.12#015#012s=-#015#012c=IN IP4
190.124.220.12
#015#012t=0 0#015#012m=audio 18338 RTP/AVP 0 101#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16]
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21746]:
DBG:core:parse_to: display={011359883327749},
ruri={sip:359883327749@69.25.128.233}
Feb 11 12:53:05 sms rtpproxy[21731]: DBUG:handle_command: received command
21746_6 LA 4512c49c3cd0db1b410744fe0ced15bf@69.25.128.233 190.124.220.12
18338 as612bc040;1 B2B.599.537;1
Feb 11 12:53:05 sms kernel: [7145167.526106] rtpproxy[21731]: segfault at
0 ip 004053e9 sp 7fff71948b00 error 4 in rtpproxy[40+e000]
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: hash 23820 label 1987919557 branch 0
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: REF_UNSAFE:[0x7fc0f89b4f10] after is 2
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: reply matched (T=0x7fc0f89b4f10)!
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_check: end=0x7fc0f89b4f10
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1)
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: timeout waiting reply from a RTP proxy
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: proxy udp:184.106.168.144:22332 does
not respond, disable it
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: can't send command to a RTP proxy
Connection refused
repeating over 100
times
Obviously the RTPproxy dies.
What I noticed is, when i remove
rtpproxy_answer(FA);
from the onreply_route, the RTPproxy does not dies.
Any ideas what I am doing wrong ?
Thank you.
-- Kamen
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Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
Hi Kamen, On 02/11/2011 03:31 PM, Kamen Petrov wrote: Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]: ERROR:db_postgres:db_postgres_store_result: 0x7b9360: ERROR: null value in column e3_sid violates not-null constraint#012 There was a problem with the db schema for the b2b_logic table - lots of wrong NOT NULL constraints there. I have just fixed it. Please take the new schema from svn and replace the table. Obviously the RTPproxy dies. What I noticed is, when i remove /_rtpproxy_answer(FA);_/ from the onreply_route, the RTPproxy does not dies. Are you using the newest version of rtpproxy? Any ideas what I am doing wrong ? Thank you. -- Kamen Regards, -- Anca Vamanu OpenSIPS Developer ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
The last core i have is:
-rw--- 1 root root 43188224 Feb 10 11:49 /core
I did the attached tests 1 or 2 hours ago and the system time now is Fri
Feb 11 14:29:14 UTC 2011.
I guess there is no new core :(
On 11 February 2011 16:23, Ovidiu Sas o...@voipembedded.com wrote:
Please get a gdb trace from the core file.
Thanks,
Ovidiu
On Fri, Feb 11, 2011 at 8:31 AM, Kamen Petrov kamen.pet...@gmail.com
wrote:
Ok guys,
Few issues still (after updating from trunk).
As suggested, I removed the engage_rtp_proxy from the b2b opensips
instance.
I noticed the following debug from the opensips:
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360 - invalid query,
execution aborted
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: PGRES_FATAL_ERROR
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: ERROR: null value
in
column e3_sid violates not-null constraint#012
Looking on the postgres log, here is the failed SQL statement:
2011-02-11 12:49:06 UTC ERROR: null value in column e3_sid violates
not-null constraint
2011-02-11 12:49:06 UTC STATEMENT: insert into b2b_logic
(si_key,scenario,sparam0,sparam1,sparam2,sparam3,sparam4,sdp,sstate,next_sstate,e1_type,e1_sid,e1_to,e1_from,e1_key,e2_type,e2_sid,e2_to,e2_from,e2_key
) values
('545.0','','','','','','','',-3,0,0,'','
sip:17864776626@190.124.220.12:5060','sip:359883327749@69.25.128.234
','B2B.608.661',1,'','sip:17864776626@190.124.220.12:5060','
sip:359883327749@69.25.128.234','B2B.545.4207959')
I am using the default b2b postgres tables.
So next, I have the following config on the rtpproxy opensips (not the
b2b
one):
#
route[1] {
fix_nated_contact();
if (is_method(INVITE)) {
rewritehostport(184.106.168.144:5061);
if (rtpproxy_offer(eo,184.106.168.144))
t_on_reply(1);
}
else if (method == BYE || method == CANCEL) {
unforce_rtp_proxy();
}
..
}
onreply_route[1] {
if (!(status=~183 || status=~200)) {
drop;
}
rtpproxy_answer(FA);
}
#
As result, when I initiate a call, I get the following on the syslog:
Feb 11 12:52:48 sms /root/opensips-1.6.4-tls/opensips[21754]:
INFO:nathelper:rtpp_test: rtp proxy udp:184.106.168.144:22332 found,
support for it enabled
Feb 11 12:52:48 sms /root/opensips-1.6.4-tls/opensips[21753]:
INFO:nathelper:rtpp_test: rtp proxy udp:184.106.168.144:22332 found,
support for it enabled
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21746]:
DBG:nathelper:force_rtp_proxy: Forcing body:#012[v=0#015#012o=-
229796569696953 1 IN IP4 190.124.220.12#015#012s=-#015#012c=IN IP4
190.124.220.12
#015#012t=0 0#015#012m=audio 18338 RTP/AVP 0 101#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-16]
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21746]:
DBG:core:parse_to: display={011359883327749},
ruri={sip:359883327749@69.25.128.233}
Feb 11 12:53:05 sms rtpproxy[21731]: DBUG:handle_command: received
command
21746_6 LA 4512c49c3cd0db1b410744fe0ced15bf@69.25.128.233190.124.220.12
18338 as612bc040;1 B2B.599.537;1
Feb 11 12:53:05 sms kernel: [7145167.526106] rtpproxy[21731]: segfault
at
0 ip 004053e9 sp 7fff71948b00 error 4 in
rtpproxy[40+e000]
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: hash 23820 label 1987919557 branch 0
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: REF_UNSAFE:[0x7fc0f89b4f10] after is 2
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: reply matched (T=0x7fc0f89b4f10)!
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_check: end=0x7fc0f89b4f10
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:reply_received: org. status uas=100, uac[0]=0 local=0
is_invite=1)
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: timeout waiting reply from a RTP
proxy
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: proxy udp:184.106.168.144:22332
does
not respond, disable it
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: can't send command to a RTP proxy
Connection refused
repeating over 100
times
Obviously the RTPproxy dies.
What I noticed is, when i remove
Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
Then please remove the old core file and make sure that you have the
latest source on both servers.
On Fri, Feb 11, 2011 at 9:27 AM, Kamen Petrov kamen.pet...@gmail.com wrote:
The last core i have is:
-rw--- 1 root root 43188224 Feb 10 11:49 /core
I did the attached tests 1 or 2 hours ago and the system time now is Fri
Feb 11 14:29:14 UTC 2011.
I guess there is no new core :(
On 11 February 2011 16:23, Ovidiu Sas o...@voipembedded.com wrote:
Please get a gdb trace from the core file.
Thanks,
Ovidiu
On Fri, Feb 11, 2011 at 8:31 AM, Kamen Petrov kamen.pet...@gmail.com
wrote:
Ok guys,
Few issues still (after updating from trunk).
As suggested, I removed the engage_rtp_proxy from the b2b opensips
instance.
I noticed the following debug from the opensips:
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360 - invalid query,
execution aborted
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: PGRES_FATAL_ERROR
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: ERROR: null
value in
column e3_sid violates not-null constraint#012
Looking on the postgres log, here is the failed SQL statement:
2011-02-11 12:49:06 UTC ERROR: null value in column e3_sid violates
not-null constraint
2011-02-11 12:49:06 UTC STATEMENT: insert into b2b_logic
(si_key,scenario,sparam0,sparam1,sparam2,sparam3,sparam4,sdp,sstate,next_sstate,e1_type,e1_sid,e1_to,e1_from,e1_key,e2_type,e2_sid,e2_to,e2_from,e2_key
) values
('545.0','','','','','','','',-3,0,0,'','sip:17864776626@190.124.220.12:5060','sip:359883327749@69.25.128.234','B2B.608.661',1,'','sip:17864776626@190.124.220.12:5060','sip:359883327749@69.25.128.234','B2B.545.4207959')
I am using the default b2b postgres tables.
So next, I have the following config on the rtpproxy opensips (not the
b2b
one):
#
route[1] {
fix_nated_contact();
if (is_method(INVITE)) {
rewritehostport(184.106.168.144:5061);
if (rtpproxy_offer(eo,184.106.168.144))
t_on_reply(1);
}
else if (method == BYE || method == CANCEL) {
unforce_rtp_proxy();
}
..
}
onreply_route[1] {
if (!(status=~183 || status=~200)) {
drop;
}
rtpproxy_answer(FA);
}
#
As result, when I initiate a call, I get the following on the syslog:
Feb 11 12:52:48 sms /root/opensips-1.6.4-tls/opensips[21754]:
INFO:nathelper:rtpp_test: rtp proxy udp:184.106.168.144:22332 found,
support for it enabled
Feb 11 12:52:48 sms /root/opensips-1.6.4-tls/opensips[21753]:
INFO:nathelper:rtpp_test: rtp proxy udp:184.106.168.144:22332 found,
support for it enabled
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21746]:
DBG:nathelper:force_rtp_proxy: Forcing body:#012[v=0#015#012o=-
229796569696953 1 IN IP4 190.124.220.12#015#012s=-#015#012c=IN IP4
190.124.220.12
#015#012t=0 0#015#012m=audio 18338 RTP/AVP 0 101#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-16]
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21746]:
DBG:core:parse_to: display={011359883327749},
ruri={sip:359883327749@69.25.128.233}
Feb 11 12:53:05 sms rtpproxy[21731]: DBUG:handle_command: received
command
21746_6 LA 4512c49c3cd0db1b410744fe0ced15bf@69.25.128.233
190.124.220.12
18338 as612bc040;1 B2B.599.537;1
Feb 11 12:53:05 sms kernel: [7145167.526106] rtpproxy[21731]: segfault
at
0 ip 004053e9 sp 7fff71948b00 error 4 in
rtpproxy[40+e000]
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: hash 23820 label 1987919557 branch 0
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: REF_UNSAFE:[0x7fc0f89b4f10] after is 2
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_reply_matching: reply matched (T=0x7fc0f89b4f10)!
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:t_check: end=0x7fc0f89b4f10
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21748]:
DBG:tm:reply_received: org. status uas=100, uac[0]=0 local=0
is_invite=1)
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: timeout waiting reply from a RTP
proxy
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: proxy udp:184.106.168.144:22332
does
not respond, disable it
Feb 11 12:53:06 sms /root/opensips-1.6.4-tls/opensips[21746]:
ERROR:nathelper:send_rtpp_command: can't send command to a
Re: [OpenSIPS-Users] OpenSIPS starting Error
http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-starting-Error-td5994344.html#a5994453 http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-starting-Error-td5994344.html On Fri, Feb 4, 2011 at 5:08 PM, Venkatesh N venkatesh...@gmail.com wrote: When I enter opensipsctl start INFO: Starting OpenSIPS : ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start failed I checked /var/log/messages and got following Feb 4 18:02:23 ubuntu kernel: [ 6747.275248] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:28 ubuntu kernel: [ 6752.266702] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:33 ubuntu kernel: [ 6757.257700] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:38 ubuntu kernel: [ 6762.248970] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:43 ubuntu kernel: [ 6767.240168] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:48 ubuntu kernel: [ 6772.231405] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:53 ubuntu kernel: [ 6777.222692] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:02:58 ubuntu kernel: [ 6782.213859 +16782213859] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:03 ubuntu kernel: [ 6787.205741 +16787205741] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:08 ubuntu kernel: [ 6792.196974] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:13 ubuntu kernel: [ 6797.188208] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:18 ubuntu kernel: [ 6802.182420] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:23 ubuntu kernel: [ 6807.170216] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:28 ubuntu kernel: [ 6812.161873 +16812161873] intel ips :00:1f.6: CPU power or thermal limit exceeded Feb 4 18:03:33 ubuntu kernel: [ 6817.153087 +16817153087] intel ips :00:1f.6: CPU power or thermal limit exceeded -- Venkatesh ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
Anca:
* There was a problem with the db schema for the b2b_logic table - lots of
wrong NOT NULL constraints there. I have just fixed it. Please take the new
schema from svn and replace the table.*
-- Seems to be fine now, thank you.
* Are you using the newest version of rtpproxy?*
-- I am running 1.2.0 right now. I have been running 1.2.1 before but with
the same success. I moved back to 1.2.0 mainly because the debug does not
work with 1.2.1 and I can't see what happens in the background.
Ovidiu:
* Then please remove the old core file and make sure that you have the
latest source on both servers.*
-- I removed the old core file, tested a new call and got into the same
issue (as described before: segfault on the rtpproxy). A new core haven't
been generated. Both servers uses the same opensips setup with different
config files (loaded with: *-f file*)
On theory, I should have rtpproxy_offer on the route and rtpproxy_answer
on the onreply_route right ? Since that is the case when I have segfault
on the rtpproxy.
If I remove the rtpproxy_answer form the onreply_route, there is no
segfault, but there is no audio as well.
Please advise.
Your help guys is highly appreciated !
Kamen Petrov
On 11 February 2011 16:30, Ovidiu Sas o...@voipembedded.com wrote:
Then please remove the old core file and make sure that you have the
latest source on both servers.
On Fri, Feb 11, 2011 at 9:27 AM, Kamen Petrov kamen.pet...@gmail.com
wrote:
The last core i have is:
-rw--- 1 root root 43188224 Feb 10 11:49 /core
I did the attached tests 1 or 2 hours ago and the system time now is Fri
Feb 11 14:29:14 UTC 2011.
I guess there is no new core :(
On 11 February 2011 16:23, Ovidiu Sas o...@voipembedded.com wrote:
Please get a gdb trace from the core file.
Thanks,
Ovidiu
On Fri, Feb 11, 2011 at 8:31 AM, Kamen Petrov kamen.pet...@gmail.com
wrote:
Ok guys,
Few issues still (after updating from trunk).
As suggested, I removed the engage_rtp_proxy from the b2b opensips
instance.
I noticed the following debug from the opensips:
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360 - invalid query,
execution aborted
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360:
PGRES_FATAL_ERROR
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: ERROR: null
value in
column e3_sid violates not-null constraint#012
Looking on the postgres log, here is the failed SQL statement:
2011-02-11 12:49:06 UTC ERROR: null value in column e3_sid
violates
not-null constraint
2011-02-11 12:49:06 UTC STATEMENT: insert into b2b_logic
(si_key,scenario,sparam0,sparam1,sparam2,sparam3,sparam4,sdp,sstate,next_sstate,e1_type,e1_sid,e1_to,e1_from,e1_key,e2_type,e2_sid,e2_to,e2_from,e2_key
) values
('545.0','','','','','','','',-3,0,0,'','
sip:17864776626@190.124.220.12:5060','sip:359883327749@69.25.128.234
','B2B.608.661',1,'','sip:17864776626@190.124.220.12:5060','
sip:359883327749@69.25.128.234','B2B.545.4207959')
I am using the default b2b postgres tables.
So next, I have the following config on the rtpproxy opensips (not
the
b2b
one):
#
route[1] {
fix_nated_contact();
if (is_method(INVITE)) {
rewritehostport(184.106.168.144:5061);
if (rtpproxy_offer(eo,184.106.168.144))
t_on_reply(1);
}
else if (method == BYE || method == CANCEL) {
unforce_rtp_proxy();
}
..
}
onreply_route[1] {
if (!(status=~183 || status=~200)) {
drop;
}
rtpproxy_answer(FA);
}
#
As result, when I initiate a call, I get the following on the syslog:
Feb 11 12:52:48 sms /root/opensips-1.6.4-tls/opensips[21754]:
INFO:nathelper:rtpp_test: rtp proxy udp:184.106.168.144:22332
found,
support for it enabled
Feb 11 12:52:48 sms /root/opensips-1.6.4-tls/opensips[21753]:
INFO:nathelper:rtpp_test: rtp proxy udp:184.106.168.144:22332
found,
support for it enabled
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21746]:
DBG:nathelper:force_rtp_proxy: Forcing body:#012[v=0#015#012o=-
229796569696953 1 IN IP4 190.124.220.12#015#012s=-#015#012c=IN IP4
190.124.220.12
#015#012t=0 0#015#012m=audio 18338 RTP/AVP 0 101#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-16]
Feb 11 12:53:05 sms /root/opensips-1.6.4-tls/opensips[21746]:
DBG:core:parse_to: display={011359883327749},
Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
Also, this is how I am running the rtpproxy:
23414 ?Ss 0:00 /usr/local/bin/rtpproxy -s udp:184.106.168.144
22332 -u root root -p /var/run/rtpproxy/rtpproxy.pid -F -l 184.106.168.144
And here is the nathelper config for both opensips and b2b:
modparam(nathelper, rtpproxy_sock, udp:184.106.168.144:22332)
modparam(nathelper, force_socket, udp:184.106.168.144:22332)
modparam(nathelper, rtpproxy_retr, 2)
modparam(nathelper, received_avp, $avp(i:42))
modparam(nathelper, ping_nated_only, 1)
modparam(nathelper, rtpproxy_autobridge, 1)
modparam(nathelper, sipping_bflag, 8)
modparam(nathelper, sipping_from, sip:pin...@platform.worldtalkinc.com
)
modparam(nathelper, sipping_method, INFO)
Does anything of that seems suspicious to you ?
On 11 February 2011 16:42, Kamen Petrov kamen.pet...@gmail.com wrote:
Anca:
* There was a problem with the db schema for the b2b_logic table - lots
of wrong NOT NULL constraints there. I have just fixed it. Please take the
new schema from svn and replace the table.*
-- Seems to be fine now, thank you.
* Are you using the newest version of rtpproxy?*
-- I am running 1.2.0 right now. I have been running 1.2.1 before but with
the same success. I moved back to 1.2.0 mainly because the debug does not
work with 1.2.1 and I can't see what happens in the background.
Ovidiu:
* Then please remove the old core file and make sure that you have the
latest source on both servers.*
-- I removed the old core file, tested a new call and got into the same
issue (as described before: segfault on the rtpproxy). A new core haven't
been generated. Both servers uses the same opensips setup with different
config files (loaded with: *-f file*)
On theory, I should have rtpproxy_offer on the route and rtpproxy_answer
on the onreply_route right ? Since that is the case when I have segfault
on the rtpproxy.
If I remove the rtpproxy_answer form the onreply_route, there is no
segfault, but there is no audio as well.
Please advise.
Your help guys is highly appreciated !
Kamen Petrov
On 11 February 2011 16:30, Ovidiu Sas o...@voipembedded.com wrote:
Then please remove the old core file and make sure that you have the
latest source on both servers.
On Fri, Feb 11, 2011 at 9:27 AM, Kamen Petrov kamen.pet...@gmail.com
wrote:
The last core i have is:
-rw--- 1 root root 43188224 Feb 10 11:49 /core
I did the attached tests 1 or 2 hours ago and the system time now is
Fri
Feb 11 14:29:14 UTC 2011.
I guess there is no new core :(
On 11 February 2011 16:23, Ovidiu Sas o...@voipembedded.com wrote:
Please get a gdb trace from the core file.
Thanks,
Ovidiu
On Fri, Feb 11, 2011 at 8:31 AM, Kamen Petrov kamen.pet...@gmail.com
wrote:
Ok guys,
Few issues still (after updating from trunk).
As suggested, I removed the engage_rtp_proxy from the b2b opensips
instance.
I noticed the following debug from the opensips:
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360 - invalid
query,
execution aborted
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360:
PGRES_FATAL_ERROR
Feb 11 12:49:06 sms /root/opensips-1.6.4-tls/opensips[21621]:
ERROR:db_postgres:db_postgres_store_result: 0x7b9360: ERROR: null
value in
column e3_sid violates not-null constraint#012
Looking on the postgres log, here is the failed SQL statement:
2011-02-11 12:49:06 UTC ERROR: null value in column e3_sid
violates
not-null constraint
2011-02-11 12:49:06 UTC STATEMENT: insert into b2b_logic
(si_key,scenario,sparam0,sparam1,sparam2,sparam3,sparam4,sdp,sstate,next_sstate,e1_type,e1_sid,e1_to,e1_from,e1_key,e2_type,e2_sid,e2_to,e2_from,e2_key
) values
('545.0','','','','','','','',-3,0,0,'','
sip:17864776626@190.124.220.12:5060','sip:359883327749@69.25.128.234
','B2B.608.661',1,'','sip:17864776626@190.124.220.12:5060','
sip:359883327749@69.25.128.234','B2B.545.4207959')
I am using the default b2b postgres tables.
So next, I have the following config on the rtpproxy opensips (not
the
b2b
one):
#
route[1] {
fix_nated_contact();
if (is_method(INVITE)) {
rewritehostport(184.106.168.144:5061);
if (rtpproxy_offer(eo,184.106.168.144))
t_on_reply(1);
}
else if (method == BYE || method == CANCEL) {
unforce_rtp_proxy();
}
..
}
onreply_route[1] {
if (!(status=~183 || status=~200)) {
drop;
}
rtpproxy_answer(FA);
}
#
As result, when I initiate a call, I get the following on the
Re: [OpenSIPS-Users] OpenSIPS no presentity entry in Database
What kind of presence do you use (configuration option in xlite)? end-to-end: that should work out of the box presence-agent: opensips must be configured as presence server, probably with proper xcap authorization rules (or disable them) klaus Am 01.02.2011 00:07, schrieb ViennaCivicEP2: Hi, i´m new to the Opensips community. I started a few days ago and i´m now at the point to post my first question, because i cant fiddle out my mistake in configuration. This is what i´ve done so far. - Setting up 3 Virtual Machines (1x Debian Lenny Server, 2x Windows XP Host with X-Lite Clients) [The VM´s can communicate with each other] - Download Opensips 1.6.4-2-notls - compiled and installed opensips (works as should i think) (means: i cant see error messages in syslog, startup and restart don´t show errors or warnings) - edited the opensips.cfg file to enable mysql support and presence Here´s my problem: I can call from one X-Lite Client to the other one (works in both directions), but the problem is i can´t see the other users online state (presence). i´m using x-lite 4 and right beside the contact it writes waiting for response - but it doesn´t change. i searched all over the net - but i can´t find a solution for my problem. i think that the opensips.cfg file is correct (with the -c option i don´t get errors) and the calling works fine. I would kindly ask for your help - if you tell me what information or file you need exactly i´ll post it right here. Thanks in advance - greetings from vienna, Mario ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS no presentity entry in Database
And just a followup from what Klaus mentioned here is a link from the OpenSIPS tutorial page on how you can set up Presence http://www.opensips.org/Resources/DocsPapPa On Fri, Feb 11, 2011 at 8:46 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: What kind of presence do you use (configuration option in xlite)? end-to-end: that should work out of the box presence-agent: opensips must be configured as presence server, probably with proper xcap authorization rules (or disable them) klaus Am 01.02.2011 00:07, schrieb ViennaCivicEP2: Hi, i´m new to the Opensips community. I started a few days ago and i´m now at the point to post my first question, because i cant fiddle out my mistake in configuration. This is what i´ve done so far. - Setting up 3 Virtual Machines (1x Debian Lenny Server, 2x Windows XP Host with X-Lite Clients) [The VM´s can communicate with each other] - Download Opensips 1.6.4-2-notls - compiled and installed opensips (works as should i think) (means: i cant see error messages in syslog, startup and restart don´t show errors or warnings) - edited the opensips.cfg file to enable mysql support and presence Here´s my problem: I can call from one X-Lite Client to the other one (works in both directions), but the problem is i can´t see the other users online state (presence). i´m using x-lite 4 and right beside the contact it writes waiting for response - but it doesn´t change. i searched all over the net - but i can´t find a solution for my problem. i think that the opensips.cfg file is correct (with the -c option i don´t get errors) and the calling works fine. I would kindly ask for your help - if you tell me what information or file you need exactly i´ll post it right here. Thanks in advance - greetings from vienna, Mario ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] rtpproxy_stream2uac
Hi Cris, On 02/09/2011 02:35 PM, chris wrote: Want to play back an in call announcement using rtpproxy. This is available in rtpproxy itself and is accessible through the rtpproxy module for kamailio but doesn’t seem to be available in the opensips nathelper implementation. It is in OpenSIPS also, probalby you missed it in readme - http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#rtpproxy_stream2xxx . Regards, -- Anca Vamanu OpenSIPS Developer ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
On 02/11/2011 03:31 PM, Kamen Petrov wrote:
/onreply_route[1] {
if (!(status=~183 || status=~200)) {
drop;
}
rtpproxy_answer(FA);
/
Maybe you could try to use other flags, or renounce at one at a time to
see which one results in segmentation fault. You should also report this
to the rtpproxy list.
--
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OpenSIPS Developer
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Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
Hi Anca,
Ok, I managed it work your way.
The key was not in the rtpproxy_answer but the rtpproxy_offer :)
Once again thanks to you and Ovidiu for your great help !
So just for the record if someone else face the same issue: segfault in the
rtpproxy on the onreply_route: don't look only the rtpproxy_answer but also
play with the rtpproxy_offer
On 11 February 2011 18:30, Anca Vamanu a...@opensips.org wrote:
On 02/11/2011 03:31 PM, Kamen Petrov wrote:
*onreply_route[1] {
if (!(status=~183 || status=~200)) {
drop;
}
rtpproxy_answer(FA);
*
Maybe you could try to use other flags, or renounce at one at a time to see
which one results in segmentation fault. You should also report this to the
rtpproxy list.
--
Anca Vamanu
OpenSIPS Developer
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Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
Please report your crash on the rtpproxy list and provide a way to reproduce it.
Rtpproxy should not crash that easy.
Regards,
Ovidiu Sas
On Fri, Feb 11, 2011 at 12:04 PM, Kamen Petrov kamen.pet...@gmail.com wrote:
Hi Anca,
Ok, I managed it work your way.
The key was not in the rtpproxy_answer but the rtpproxy_offer :)
Once again thanks to you and Ovidiu for your great help !
So just for the record if someone else face the same issue: segfault in the
rtpproxy on the onreply_route: don't look only the rtpproxy_answer but also
play with the rtpproxy_offer
On 11 February 2011 18:30, Anca Vamanu a...@opensips.org wrote:
On 02/11/2011 03:31 PM, Kamen Petrov wrote:
onreply_route[1] {
if (!(status=~183 || status=~200)) {
drop;
}
rtpproxy_answer(FA);
Maybe you could try to use other flags, or renounce at one at a time to
see which one results in segmentation fault. You should also report this to
the rtpproxy list.
--
Anca Vamanu
OpenSIPS Developer
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Re: [OpenSIPS-Users] Weird behaviour
Hi Dave,
Yeah, you're right.. Basically allow only REGISTER requests from anywhere and
the rest check the source ip.
Great ideea.
I will implement it as soon as possible.
Thanks,
Adrian Vasile
y...@opennet.ro
On Feb 10, 2011, at 10:41 PM, Dave Singer wrote:
Adrian,
I was just thinking about the implementing no response for INVITE a
little more...
You would want to handle the response checking similar to the
register. If not found in the cache where you check the location
table if there is a registered user at the source ip.
That way it can handle opensips reboots and other situations where the
cache is lost or unavailable. Like a memcached server fails.
Advantage to using external memcached vs local cache would be that
cache would not be cleared on opensips restart.
Dave
On Thu, Feb 10, 2011 at 11:16 AM, Dave Singer dave.sin...@wideideas.com
wrote:
I've found that generally they start out with the sip NOTIFY or
OPTIONS message. So recently I set in the script to drop them from
sources I'm not expecting them from. Might not work so well for some
situation like ATA's sending pings for nat keep alives. But for the
nat to keep open, generally it doesn't need a response, just as long
as they keep sending the packets. Some devices I've seen actually send
essentially an empty packet to the sip port, just enough to keep the
nat alive but opensips just discards it because it is empty.
The one I do send a reply to is my network monitoring server. Kind of
helpful to know when things stop responding. :-)
If an ATA model need to actually get a reply you could on registration
check the model listed in the sip agent header and use localcache or
memcached to store the source IP as ok to respond to. See
http://www.opensips.org/Resources/DocsCoreFcn16#toc98
cache_store and cache_fetch
at registration something like
save(location);
cache_store(local, ping_ok_$si, ok, 86000);
and at ping
if ( $rm =~ OPTIONS|NOTIFY ) {
if( $si == monitor server || $cache_fetch(local,
pingok_$si, $avp(i:5)) {
sl_send_reply(200, Ok);
}
drop;
}
Might not need pike if they never start the brute force scan because
they didn't get the initial reply.
I just came up with this the other day so it is an unproved theory.
The other day I left a packet capture running over night on the
testing server and in the morning I saw all the failed register
attempts. I looked back to the first packet from the registering
source and found the first one was an OPTIONS packet and thus my
theory.
Could apply it to INVITE and other messages. For registrations if
there wasn't a hit in the cache you would want to do a db lookup to
see if the from user is one of yours. But generally that would only be
for a first time registration since registrations usually happen every
30 min. (This is just brainstorming) ;-)
Let me know if you implement some of it and what results you find.
Dave
On Thu, Feb 10, 2011 at 10:28 AM, Adrian Vasile y...@opennet.ro wrote:
I know of these issues. And all client are either behind NAT either separate
voice vlans.
As for securing the proxy. What methods either than Pike combined with
fail2ban would you advise?
And I finally found the culprit. Auth INVITE:
When enabled, authorization is required for initial incoming INVITE
requests from the SIP proxy.
On Feb 10, 2011, at 6:57 PM, Dave Singer wrote:
Adrian,
There are lots of people out there with servers doing sip scans to see
if an ip will respond to a sip ping (NOTIFY or OPTIONS message). Then
they will either try to send register and/or invites for all sorts of
numbers trying to get a hit. Of course the invites are not actual
calls so if the sip scanner gets an ATA, the customer answers the
phone and there is no one there. Depending on the scanner it may keep
trying through it's whole list of common sip source accounts. Then it
can get interesting. The scanner would then mark the IP as a success
and the hacker can then start trying to send calls through it. Though
likely they would try a call to something like a Home Depot number and
when the customer answers they just say sorry wrong number and mark
the IP off their list. Customer is left alone till the next scanner
comes sniffing.
So ATA's many times have settings for not answering calls from places
that shouldn't be sending them calls. The options are usually
something like calls ok: from register server, from proxy server,
call to registered user, auth call or similar.
See what you can find in the docs for that model.
Dave
On Thu, Feb 10, 2011 at 5:07 AM, Adrian Vasile y...@opennet.ro wrote:
Hi,
I attached the trace.
why does the cisco spa ask for authorization?
Thanks,
Adrian Vasile
y...@opennet.ro
On Feb 10, 2011, at 12:42 PM, Laszlo wrote:
Hi Adrian,
2011/2/10 Adrian Vasile y...@opennet.ro
Hello all,
Maybe it has happened to you too.. I've got a couple
Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
You are right.
I just escalated the scenario to de...@rtpproxy.org
Thank you.
On 11 February 2011 19:15, Ovidiu Sas o...@voipembedded.com wrote:
Please report your crash on the rtpproxy list and provide a way to
reproduce it.
Rtpproxy should not crash that easy.
Regards,
Ovidiu Sas
On Fri, Feb 11, 2011 at 12:04 PM, Kamen Petrov kamen.pet...@gmail.com
wrote:
Hi Anca,
Ok, I managed it work your way.
The key was not in the rtpproxy_answer but the rtpproxy_offer :)
Once again thanks to you and Ovidiu for your great help !
So just for the record if someone else face the same issue: segfault in
the
rtpproxy on the onreply_route: don't look only the rtpproxy_answer but
also
play with the rtpproxy_offer
On 11 February 2011 18:30, Anca Vamanu a...@opensips.org wrote:
On 02/11/2011 03:31 PM, Kamen Petrov wrote:
onreply_route[1] {
if (!(status=~183 || status=~200)) {
drop;
}
rtpproxy_answer(FA);
Maybe you could try to use other flags, or renounce at one at a time to
see which one results in segmentation fault. You should also report this
to
the rtpproxy list.
--
Anca Vamanu
OpenSIPS Developer
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Re: [OpenSIPS-Users] FW: CANCELs with no transaction
I have a similar problem, but not solution, my probably is actually occurring
because the originating UA is ignoring a contact header that is sent back
during a 183 progress message. OpenSIPS uses information from that contact
header to figure out where to relay the incoming message (BYE in my case,
CANCEL in yours). It seems like it would be possible for OpenSIPS to use a
call-id or tag to determine where to relay the message though.
Russell Bierschbach
em: rbierschb...@telepointglobal.commailto:rjphill...@telepointglobal.com,
im: rbierschb...@hotmail.commailto:rbierschb...@hotmail.com
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Juri Nysschen
Sent: Friday, February 11, 2011 7:44 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] FW: CANCELs with no transaction
Hi All,
Need help with a nagging issue:
UA-Opensips 1-Opensips 2-PSTN
UA sends an invite on Opensips 1, and is routed via do_routing() to Opensips 2,
Opensips 2 uses do_routing to get to the PSTN, call starts ringing.
UA cancels call before answer, but now t_check_trans fails and the CANCEL is
not passed onto the PSTN, with the result that the call rings forever and can
only be terminated by the remote answering and dropping the call or through a
timeout.
The scripts on Opensips 1 Opensips 2 is virtuall identical:
How do I get the CANCEL to the PSTN ?
route{
.
if (is_method(CANCEL) ) {
route(5); # drop media proxy
if (t_check_trans()){ # this always fails after a do_routing()
xlog(L_INFO,CANCEL Transaction[$fd/$fu/$rd/$ru/$si/]\n);
t_relay();
exit;
};
exit;
}
}
route[4] {
xlog(L_INFO,Route4 [$fd/$fu/$rd/$ru/$si/]\n);
$avp(i:102)=1; # Default dr-group
route(10); # Do custom stuff
t_on_failure(4);
if (do_routing($avp(i:102))){
xlog(L_INFO,Route4 Route to Dyna Group:
$avp(i:102)[$fd/$fu/$rd/$ru/$si/]\n);
t_newtran();
route(1);
exit;
};
xlog(L_INFO,Route4 No Route to Host[$fd/$fu/$rd/$ru/$si/]\n);
sl_reply_error();
exit;
}
Regards
Juri Nysschenhttp://www.greydotelecom.net/bcard/jnysschen.htm
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Re: [OpenSIPS-Users] Weird behaviour
Adrian,
Probably want to only respond to registers that are to valid user
accounts, drop the rest, as they start scanning with like 100, 101,
., 5000, etc
Dave
On Fri, Feb 11, 2011 at 6:25 AM, Adrian Vasile y...@opennet.ro wrote:
Hi Dave,
Yeah, you're right.. Basically allow only REGISTER requests from anywhere and
the rest check the source ip.
Great ideea.
I will implement it as soon as possible.
Thanks,
Adrian Vasile
y...@opennet.ro
On Feb 10, 2011, at 10:41 PM, Dave Singer wrote:
Adrian,
I was just thinking about the implementing no response for INVITE a
little more...
You would want to handle the response checking similar to the
register. If not found in the cache where you check the location
table if there is a registered user at the source ip.
That way it can handle opensips reboots and other situations where the
cache is lost or unavailable. Like a memcached server fails.
Advantage to using external memcached vs local cache would be that
cache would not be cleared on opensips restart.
Dave
On Thu, Feb 10, 2011 at 11:16 AM, Dave Singer dave.sin...@wideideas.com
wrote:
I've found that generally they start out with the sip NOTIFY or
OPTIONS message. So recently I set in the script to drop them from
sources I'm not expecting them from. Might not work so well for some
situation like ATA's sending pings for nat keep alives. But for the
nat to keep open, generally it doesn't need a response, just as long
as they keep sending the packets. Some devices I've seen actually send
essentially an empty packet to the sip port, just enough to keep the
nat alive but opensips just discards it because it is empty.
The one I do send a reply to is my network monitoring server. Kind of
helpful to know when things stop responding. :-)
If an ATA model need to actually get a reply you could on registration
check the model listed in the sip agent header and use localcache or
memcached to store the source IP as ok to respond to. See
http://www.opensips.org/Resources/DocsCoreFcn16#toc98
cache_store and cache_fetch
at registration something like
save(location);
cache_store(local, ping_ok_$si, ok, 86000);
and at ping
if ( $rm =~ OPTIONS|NOTIFY ) {
if( $si == monitor server || $cache_fetch(local,
pingok_$si, $avp(i:5)) {
sl_send_reply(200, Ok);
}
drop;
}
Might not need pike if they never start the brute force scan because
they didn't get the initial reply.
I just came up with this the other day so it is an unproved theory.
The other day I left a packet capture running over night on the
testing server and in the morning I saw all the failed register
attempts. I looked back to the first packet from the registering
source and found the first one was an OPTIONS packet and thus my
theory.
Could apply it to INVITE and other messages. For registrations if
there wasn't a hit in the cache you would want to do a db lookup to
see if the from user is one of yours. But generally that would only be
for a first time registration since registrations usually happen every
30 min. (This is just brainstorming) ;-)
Let me know if you implement some of it and what results you find.
Dave
On Thu, Feb 10, 2011 at 10:28 AM, Adrian Vasile y...@opennet.ro wrote:
I know of these issues. And all client are either behind NAT either
separate
voice vlans.
As for securing the proxy. What methods either than Pike combined with
fail2ban would you advise?
And I finally found the culprit. Auth INVITE:
When enabled, authorization is required for initial incoming INVITE
requests from the SIP proxy.
On Feb 10, 2011, at 6:57 PM, Dave Singer wrote:
Adrian,
There are lots of people out there with servers doing sip scans to see
if an ip will respond to a sip ping (NOTIFY or OPTIONS message). Then
they will either try to send register and/or invites for all sorts of
numbers trying to get a hit. Of course the invites are not actual
calls so if the sip scanner gets an ATA, the customer answers the
phone and there is no one there. Depending on the scanner it may keep
trying through it's whole list of common sip source accounts. Then it
can get interesting. The scanner would then mark the IP as a success
and the hacker can then start trying to send calls through it. Though
likely they would try a call to something like a Home Depot number and
when the customer answers they just say sorry wrong number and mark
the IP off their list. Customer is left alone till the next scanner
comes sniffing.
So ATA's many times have settings for not answering calls from places
that shouldn't be sending them calls. The options are usually
something like calls ok: from register server, from proxy server,
call to registered user, auth call or similar.
See what you can find in the docs for that model.
Dave
On Thu, Feb 10, 2011 at 5:07 AM, Adrian Vasile y...@opennet.ro wrote:
Hi,
I attached the trace.
why does the cisco spa ask for
[OpenSIPS-Users] Reject INVITEs with invalid (unable to be parsed) headers
What is the easiest way to identify traffic with invalid headers? Specifically, the from and to URIs. For example, if OpenSIPS is unable to parse a from URI, would $fu be NULL? Thanks. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Weird behaviour
That's why I dropped the ideea of having numbered usernames…
On Feb 11, 2011, at 10:45 PM, Dave Singer wrote:
Adrian,
Probably want to only respond to registers that are to valid user
accounts, drop the rest, as they start scanning with like 100, 101,
., 5000, etc
Dave
On Fri, Feb 11, 2011 at 6:25 AM, Adrian Vasile y...@opennet.ro wrote:
Hi Dave,
Yeah, you're right.. Basically allow only REGISTER requests from anywhere and
the rest check the source ip.
Great ideea.
I will implement it as soon as possible.
Thanks,
Adrian Vasile
y...@opennet.ro
On Feb 10, 2011, at 10:41 PM, Dave Singer wrote:
Adrian,
I was just thinking about the implementing no response for INVITE a
little more...
You would want to handle the response checking similar to the
register. If not found in the cache where you check the location
table if there is a registered user at the source ip.
That way it can handle opensips reboots and other situations where the
cache is lost or unavailable. Like a memcached server fails.
Advantage to using external memcached vs local cache would be that
cache would not be cleared on opensips restart.
Dave
On Thu, Feb 10, 2011 at 11:16 AM, Dave Singer dave.sin...@wideideas.com
wrote:
I've found that generally they start out with the sip NOTIFY or
OPTIONS message. So recently I set in the script to drop them from
sources I'm not expecting them from. Might not work so well for some
situation like ATA's sending pings for nat keep alives. But for the
nat to keep open, generally it doesn't need a response, just as long
as they keep sending the packets. Some devices I've seen actually send
essentially an empty packet to the sip port, just enough to keep the
nat alive but opensips just discards it because it is empty.
The one I do send a reply to is my network monitoring server. Kind of
helpful to know when things stop responding. :-)
If an ATA model need to actually get a reply you could on registration
check the model listed in the sip agent header and use localcache or
memcached to store the source IP as ok to respond to. See
http://www.opensips.org/Resources/DocsCoreFcn16#toc98
cache_store and cache_fetch
at registration something like
save(location);
cache_store(local, ping_ok_$si, ok, 86000);
and at ping
if ( $rm =~ OPTIONS|NOTIFY ) {
if( $si == monitor server || $cache_fetch(local,
pingok_$si, $avp(i:5)) {
sl_send_reply(200, Ok);
}
drop;
}
Might not need pike if they never start the brute force scan because
they didn't get the initial reply.
I just came up with this the other day so it is an unproved theory.
The other day I left a packet capture running over night on the
testing server and in the morning I saw all the failed register
attempts. I looked back to the first packet from the registering
source and found the first one was an OPTIONS packet and thus my
theory.
Could apply it to INVITE and other messages. For registrations if
there wasn't a hit in the cache you would want to do a db lookup to
see if the from user is one of yours. But generally that would only be
for a first time registration since registrations usually happen every
30 min. (This is just brainstorming) ;-)
Let me know if you implement some of it and what results you find.
Dave
On Thu, Feb 10, 2011 at 10:28 AM, Adrian Vasile y...@opennet.ro wrote:
I know of these issues. And all client are either behind NAT either
separate
voice vlans.
As for securing the proxy. What methods either than Pike combined with
fail2ban would you advise?
And I finally found the culprit. Auth INVITE:
When enabled, authorization is required for initial incoming INVITE
requests from the SIP proxy.
On Feb 10, 2011, at 6:57 PM, Dave Singer wrote:
Adrian,
There are lots of people out there with servers doing sip scans to see
if an ip will respond to a sip ping (NOTIFY or OPTIONS message). Then
they will either try to send register and/or invites for all sorts of
numbers trying to get a hit. Of course the invites are not actual
calls so if the sip scanner gets an ATA, the customer answers the
phone and there is no one there. Depending on the scanner it may keep
trying through it's whole list of common sip source accounts. Then it
can get interesting. The scanner would then mark the IP as a success
and the hacker can then start trying to send calls through it. Though
likely they would try a call to something like a Home Depot number and
when the customer answers they just say sorry wrong number and mark
the IP off their list. Customer is left alone till the next scanner
comes sniffing.
So ATA's many times have settings for not answering calls from places
that shouldn't be sending them calls. The options are usually
something like calls ok: from register server, from proxy server,
call to registered user, auth call or similar.
See what you can find in the docs for that model.
Dave
