Re: [OpenSIPS-Users] Channel Limit
Hi, I am trying to limit number of channels on a called DID going to opensips. I read the following doc: http://www.opensips.org/Resources/DocsTutConcurrentCalls how do I use it for inbound calls coming from PSTN to opensips and limit simultaneous calls on it? How would you like to configure your channel limit? Static or dynamic? You need to identify the source of the number (User Part of RURI?), mabye make a lookup against a db and then test, if the channel limit is reached. Its not so complicated. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Multi-domain and reinvite authentications
Hi, Is there a better implementation? Yes, don't ask for authentication for a re-INVITE :) Hi Iñaki, Is this the right implementation or a workaround? (in Flavio Goncalves' book I see the authentication of re-invites...) There could be a security issue without this authentication? (for example a custom packet with a fake to_tag and a route header? There are several UA which cannot handle AUTH on reinvite. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] solved ; ( Re: parallel fork and mediaproxy = false behaviour?
Hi,
Is there an option to prevent this behavior with mediaproxy?
opensips: 10.20.20.159 and 10.20.30.159
UACs: 10.20.20.25 and 10.20.20.26
UAS: 17.17.17.167
You have something wrong on you opensips.cfg, for sure ... We have lot
of UAC's working on that scenario you described, without any problem.
Good to hear that.
Are you using the dialog module? ... if yes, take into account it
limitations to work with parallel forking.
You are working not with engage_media_proxy() then?
Of course, becasue engage_media_proxy NEEDS the dialog module ... and
it's a know limitation of dialog module that it doesn't work AT ALL with
parallel forking, or with multiple 1XX replies.
Better if you limit your uses of the dialog module to the minimum ... let
say ... to 0 .. ;-)
At which time are you calling use_media_proxy() then? On the 1st INVITE
or later with the 200 OK with SDP? Is there any example out there?
On First invite after checking if needed, on_reply for 1XX or 200 with SDP,
on
re-invites
You could see and example on sipwise.com
I was not able to get it up and running with use_media_proxy and
end_media_session, since the mediasession was ended, if a BYE for the
2nd branch arrived.
I changed a lot in my script and now its working with engage_media_proxy
as expected. I have no idea why its working now. The only relevant think
I have changed was. Strange, I will try which changed fixed that point.
old:
route[1] {
...
engage_media_proxy();
...
t_on_branch(1);
t_relay(),
...
}
new:
route[1] {
...
engage_media_proxy();
...
t_on_branch(1);
t_on_reply(1);
route(3),
...
}
route[1] {
...
engage_media_proxy();
...
t_on_reply(2);
t_relay();
...
}
BR
Uwe
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[OpenSIPS-Users] eBootcamp for opensips
Bogdan, I couple of month ago there was an idea to have a kind of elearning Bootcamp for opensips. Is it still planned? Depending on the price I would like to book 2 or 3 slots - maybe this would help planing it. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] solved ; ( Re: parallel fork and mediaproxy = false behaviour?
Raúl Alexis Betancor Santana schrieb:
Uwe Kastens escribió:
I was not able to get it up and running with use_media_proxy and
end_media_session, since the mediasession was ended, if a BYE for the
2nd branch arrived.
That could only occurs if you get something wrong with the branches. For
sure :-)
Yes, I am sure about this. Is there any good tutorial out there?
I changed a lot in my script and now its working with engage_media_proxy
as expected. I have no idea why its working now. The only relevant think
I have changed was. Strange, I will try which changed fixed that point.
Do you have a duplicated route[1] ?? ...
no, sorr, a typo, should be:
new:
route[1] {
...
engage_media_proxy();
...
t_on_branch(1);
t_on_reply(1);
route(3),
...
}
route[3] {
...
..
t_on_reply(2);
t_relay();
...
}
--
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[OpenSIPS-Users] parallel fork and mediaproxy = false behaviour?
Hello @all, My favorite szenario again:) Parallel forked INVITE to 2 UACs, both are sending back 200 OK with SDP, UAS sends an bye for the 2nd call leg. Mediaproxy is using the SDP information from the 2nd UACs, which has dropped the call already. Is there an option to prevent this behavior with mediaproxy? opensips: 10.20.20.159 and 10.20.30.159 UACs: 10.20.20.25 and 10.20.20.26 UAS: 17.17.17.167 1) Callflow from opensips to the UACs |Time | 10.20.20.159 | 10.20.20.26 | 10.20.20.25 | |3,446| INVITE SDP ( CLEARMODE) | |SIP From: sip:00497097...@sip.domain.de To:sip:00499089751...@10.20.30.159 | |(5060) -- (5060) | | |3,446| INVITE SDP ( CLEARMODE) | |SIP From: sip:00497097...@sip.domain.de To:sip:00499089751...@10.20.30.159 | |(5060) -- (5060) | |3,446| 100 Trying| | |SIP Status | |(5060) -- (5060) | | |3,446| 100 Trying| | |SIP Status | |(5060) -- (5060) | |3,446| 200 OK SDP ( g711U) | |SIP Status | |(5060) -- (5060) | |3,446| 200 OK SDP ( g711U) | |SIP Status | |(5060) -- (5060) | | |3,447| RTP (g711A) | |RTP Num packets:982 Duration:19.620s SSRC:0x2A498D0F | |(5054) -- (18572) | |3,480| CANCEL| | |SIP Request | |(5060) -- (5060) | | |3,481| 200 OK| | |SIP Status | |(5060) -- (5060) | | |3,483| ACK | | |SIP Request | |(5060) -- (5060) | |3,492| ACK | | |SIP Request | |(5060) -- (5060) | | |3,493| BYE | | |SIP Request | |(5060) -- (5060) | | |3,493| 200 OK| | |SIP Status | |(5060) -- (5060) | | |23,076 | BYE | | |SIP Request | |(5060) -- (5060) | |23,077 | 200 OK| | |SIP Status | |(5060) -- (5060) | 2) Callflow beetwen opensips and UAS |Time | 17.17.17.167| 10.20.30.159 | |0,000| INVITE SDP ( CLEARMODE) |SIP From: sip:00497097...@sip.domain.de To:sip:00499089751...@10.20.30.159 | |(5060) -- (5100) | |0,005| 100 Giving a try |SIP Status | |(5060) -- (5100) | |0,040| 200 OK SDP ( g711U) |SIP Status | |(5060) -- (5100) | |0,041| ACK | |SIP Request | |(5060) -- (5100) | |0,049| 200 OK SDP ( g711U) |SIP Status | |(5060) -- (5100) | |0,051| ACK | |SIP Request | |(5060) -- (5100) | |0,051| BYE | |SIP Request | |(5060) -- (5100) | |0,066| 200 OK| |SIP Status | |(5060) -- (5100) | |19,629 | BYE | |SIP Request | |(5060) -- (5100) | |19,637 | 200 OK| |SIP Status | |(5060) -- (5100) | 3) mediaproxy log attached 4) opensips log attached BR Uwe -- kiste lat: 54.322684, lon: 10.13586 media.anon.gz Description: GNU Zip compressed data opensips.log_anon.gz Description: GNU Zip compressed data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE
Hi, IMHO a proxy shouldn't behave as a UAC. Perhaps it can monitor dialogs and so because this features just requires requests inspection, there is no intrusion (adding a Record-Route parameter is not intrusion XD). But behaving as an UAC is 100% intrusion. Yes, OpenSIPS is very flexible and can be used to solve some UA problems, but the proxy shouldn't be the key for this purpose (IMHO). Ok. I am with you. But for example looking at the problem with mediaproxy (see email from this morning), opensips is doing to much or to less ATM. So mediaproxy/opensips will talk to the wrong SDP Ports, since its using the 2nd 200 OK with SDP from the UAC answer. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel fork and mediaproxy = false behaviour?
Hi, Is there an option to prevent this behavior with mediaproxy? opensips: 10.20.20.159 and 10.20.30.159 UACs: 10.20.20.25 and 10.20.20.26 UAS: 17.17.17.167 You have something wrong on you opensips.cfg, for sure ... We have lot of UAC's working on that scenario you described, without any problem. Good to hear that. Are you using the dialog module? ... if yes, take into account it limitations to work with parallel forking. You are working not with engage_media_proxy() then? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE
Hi, Yes, I've replied to that mail right now. It seems to be a bug in mediaproxy. PS: IMHO you should try to avoid those two 200 OK (INVITE) at the same time (even if it's correct). Perhaps you could add a Wait(1) in top of the dialplan of the second Asterisk server so if there won't be a race between CANCEL and 200 (INVITE). The UAC are not under my control. Or even better: why you send the INVITE to both Asterisk at the same time (parallel forking)? Is not enough for you to do load balancing and serial forking in case of failure)? (of course it could be non suitable in your case). No, since the subscriber with the asterisk UAC is using the INVITE as LB and Failover Solution. Since this is nothing against RFC I would like to implement it. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE
Hi The UAC are not under my control. Yes, but you could instruct that UAC to send traffic to a new B2BUA server in your network, right? :) I could do this, but thats a solution for maybe a single installation not for more. The setup is getting more komplex. I will keep this option in mind Or even better: why you send the INVITE to both Asterisk at the same time (parallel forking)? Is not enough for you to do load balancing and serial forking in case of failure)? (of course it could be non suitable in your case). No, since the subscriber with the asterisk UAC is using the INVITE as LB and Failover Solution. Since this is nothing against RFC I would like to implement it. What do you mean with INVITE as LB and Failover Solution? The UAC which will answer 1st will get the call. This depends on the answer time which might be greater, if there is more load on the UAC etc.pp. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel fork and mediaproxy = false behaviour?
Hi, Is there an option to prevent this behavior with mediaproxy? opensips: 10.20.20.159 and 10.20.30.159 UACs: 10.20.20.25 and 10.20.20.26 UAS: 17.17.17.167 You have something wrong on you opensips.cfg, for sure ... We have lot of UAC's working on that scenario you described, without any problem. Good to hear that. Are you using the dialog module? ... if yes, take into account it limitations to work with parallel forking. You are working not with engage_media_proxy() then? Of course, becasue engage_media_proxy NEEDS the dialog module ... and it's a know limitation of dialog module that it doesn't work AT ALL with parallel forking, or with multiple 1XX replies. Better if you limit your uses of the dialog module to the minimum ... let say ... to 0 .. ;-) At which time are you calling use_media_proxy() then? On the 1st INVITE or later with the 200 OK with SDP? Is there any example out there? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel fork and mediaproxy = false behaviour?
Raúl Alexis Betancor Santana schrieb: On Friday 23 October 2009 14:14:52 Uwe Kastens wrote: Hi, Is there an option to prevent this behavior with mediaproxy? opensips: 10.20.20.159 and 10.20.30.159 UACs: 10.20.20.25 and 10.20.20.26 UAS: 17.17.17.167 You have something wrong on you opensips.cfg, for sure ... We have lot of UAC's working on that scenario you described, without any problem. Good to hear that. Are you using the dialog module? ... if yes, take into account it limitations to work with parallel forking. You are working not with engage_media_proxy() then? Of course, becasue engage_media_proxy NEEDS the dialog module ... and it's a know limitation of dialog module that it doesn't work AT ALL with parallel forking, or with multiple 1XX replies. Better if you limit your uses of the dialog module to the minimum ... let say ... to 0 .. ;-) At which time are you calling use_media_proxy() then? On the 1st INVITE or later with the 200 OK with SDP? Is there any example out there? On First invite after checking if needed, on_reply for 1XX or 200 with SDP, on re-invites You could see and example on sipwise.com Thanks. How would you handle parallel forking in that case? If I match for end_media_session() on BYE the media_session is dropped if the UAS sends the BYE for the 2nd call leg BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE
Hi Borgan, Sorry by trying to debug the problem I understood the hole picture. I think it might be a bug or a feature request for the tm module. The setup is: PSTN-GW - opensips as statefull proxy - AST1 + AST2 If I make a call from pstn over the opensips to a specific SIP-URI, the call will be forked parallel to AST1 and AST2. This is done statefull via tm (relay). AST1 will send a 200 OK with SDP, tm will generate a CANCEL message for the 2nd branch to AST2. AST2 has already sent a 200 OK with SDP and will therefore send 200 OK for the CANCEL request. The problem is, that both 200 OK with SDP are sent back to the PSTN GW, which has ACKed one call already and will get a 2nd 200 OK with the same branch but different call-id. This is ignored because the PSTN-GW is not aware about branches/call-id. So there are 2 possible solutions: - The PSTN GW needs to send a BYE to the branch which comes later on with the same branch but different call-id. - opensips TM should send a bye if the CANCEL if the call is forked parallel and the CANCEL Message is answered with a 200 OK. I know there is a lot of discussion about this issue, but I need a solution BR Uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, as I understand from you, from end devices (GW, as1 and as2) everything work ok, but the dialog state on opensips is not properly kept?? Regards, Bogdan Uwe Kastens wrote: Hello Bogdan, Now we changed the behaviour of the UAC. One of them will send a BYE and this is relayed to the PSTN GW which drops the call, since opensips will not handle the BYE locally. So loose_route is done and the BYE is relayed to the PSTN GW. The following is happening: 1) INVITE from PSTN GW 2) parallel forking to ast1 and ast2 (branches z9hG4bK51f6.9afa91c3.1 and z9hG4bK51f6.9afa91c3.0) 3) ast1 sends an 200 OK (branch z9hG4bK51f6.9afa91c3.0) 4) opensips sends an cancel to ast2 (branch z9hG4bK51f6.9afa91c3.1) 5) opensips receives the 200 OK from ast1 and sends an ACK (branch is changing here to z9hG4bK51f6.9afa91c3.3) 6) opensips receives 200 OK from ast2 from the INVITE (branch z9hG4bK51f6.9afa91c3.1)and sends an ACK (branch is changing to z9hG4bK51f6.9afa91c3.3) 7) opensips reives 200 OK from ast2 for the cancel request ( branch z9hG4bK51f6.9afa91c3.1) 8) opensips receives BYE from ast2 with branch z9hG4bK40d1af5d 9) opensips is doing loose_route and sends the BYE to the PSTN GW The only thing I could see on the logs is: WARNING:dialog:dlg_onroute: tight matching failed for BYE with callid='393105a419950c1f265f298914662...@10.20.30.100'/46, ftag='as63949c6e'/10, ttag='as0d1597ca'/10 and direction=0 Oct 21 09:09:15 asne02 /usr/sbin/opensips[15615]: WARNING:dialog:dlg_onroute: dialog identification elements are callid='393105a419950c1f265f298914662...@10.20.30.100'/46, caller tag='as0d1597ca'/10, callee tag='as79debd51'/10 Why is the opensips not handling the BYE locally and only closing one branch? BR UWe Bogdan-Andrei Iancu schrieb: Hi Uwe, Uwe Kastens wrote: Hi Bogdan, So actually both legs do send 200 OK (but one faster than the other)..so there is kind on race between the 200 OK from the slow branch and the CANCEL from OpenSIPS...is this the case? Exactly If so, the UAS will simply reply with negative reply to CANCEL (decline it) and opensips (for INVITE transaction) will not close the second branch as there is a 200 OK (and not a 487) received RFC3261 says that a proxy must send all 200 OK (for a call), even if more than one, to the UAC - the UAC is the one who will decide what branch to keep and it will fire a BYE for the other branch. Could this explan, why only the 2nd Node will get the BYE, if the call is released behind the opensips? yes, because the caller will hung up only one of the callee branch, so the BYE will go to only one of them. The other branch will remain up and will be the ongoing call. Regards, Bogdan ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips authentication
Hi, Are you controlling all servers? If so, you can implement a trust based on IP Adresses in your script. BR Uwe Pacho Baratta [fabbricadigitale] schrieb: Hi all, i’d like to know how should I do to place a call to an Opensips requesting authentication. This is the environment: PBX1 à Opensips1 à Opensips2 à PBX2 A user from the PBX1 wants to place a call to a user on the PBX2. The Opensips1 tries to place the call but the Opensips2 is asking for authorization. What should I do? Thanks all, Pacho fabbrica*digitale* srl *Pacho Baratta | Senior Systems Engineer * Tecnhology Engineering - Via A.Volta, 3 - 26041 – Casalmaggiore - CR Phone +39 0375 284600 Fax +39 02 57760002 _mailto:p.bara...@fabbricadigitale.it _www.fabbricadigitale.it http://www.fabbricadigitale.it/_ _ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips authentication
Ok, What kind of credentials do you have for opensips2? Individual for each account at PBX1 or one account for all? BR Uwe Pacho Baratta [fabbricadigitale] schrieb: Unfortunately not, i have no authority on Opensips2. fabbricadigitale srl Pacho Baratta | Senior Systems Engineer Tecnhology Engineering - Via A.Volta, 3 - 26041 – Casalmaggiore - CR Phone +39 0375 284600 Fax +39 02 57760002 mailto:p.bara...@fabbricadigitale.it www.fabbricadigitale.it -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: giovedì 22 ottobre 2009 09:11 To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips authentication Hi, Are you controlling all servers? If so, you can implement a trust based on IP Adresses in your script. BR Uwe Pacho Baratta [fabbricadigitale] schrieb: Hi all, i’d like to know how should I do to place a call to an Opensips requesting authentication. This is the environment: PBX1 à Opensips1 à Opensips2 à PBX2 A user from the PBX1 wants to place a call to a user on the PBX2. The Opensips1 tries to place the call but the Opensips2 is asking for authorization. What should I do? Thanks all, Pacho fabbrica*digitale* srl *Pacho Baratta | Senior Systems Engineer * Tecnhology Engineering - Via A.Volta, 3 - 26041 – Casalmaggiore - CR Phone +39 0375 284600 Fax +39 02 5776000--2 _mailto:p.bara...@fabbricadigitale.it _www.fabbricadigitale.it http://www.fabbricadigitale.it/_ _ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE
Hi, which has ACKed one call already and will get a 2nd 200 OK with the same branch but different call-id. Different call-id? Perhaps yo mean different To tag as the Call-ID is generated by the UAC (the gw) and must be the same in both legs. You are correct. The Call-ID AND branch is the same for both legs This is ignored because the PSTN-GW is not aware about branches/call-id. But what is the real problem? in the gw side you said that it ignores the second 200 coming from AST2, so there is no problem in the gw, right? The problem is, that: - rtp information is mixed between ast1 and ast2 - if ast1 or ast2 is sending a BYE, the hole call is dropped Perhaps the problem could be that AST2 replies the INVITE so it will be waiting for the ACK for ~32 seconds. You could also drop the second 200 in OpenSIPS by checking in reply_route[0] check_trans(). It will return false for the second 200 OK as the transaction was removed upon recepit of the first 200. So the call drop(). However it solves nothing since AST2 remains waiting for the ACK. Ok, it makes no sense then. So there is only one possible solution So there are 2 possible solutions: - The PSTN GW needs to send a BYE to the branch which comes later on with the same branch but different call-id. Again: replace call-id with To-tag :) Yes Yes, this is the RFC 3261 solution. The issue should be handled by the UAC rather than by a proxy. - opensips TM should send a bye if the CANCEL if the call is forked parallel and the CANCEL Message is answered with a 200 OK. The fact is that a proxy should never send a BYE. Yes, it could and it's I know there is a lot of discussion about this issue, but I need a solution What is exactly the issue? is the above explained by me? Yes. I was able to step on testing and found out, that our reference system (softsiwtch) is handling it correctly. The asterisk servers we are using as mediagateways are unable to handle it correctly - so I will need a fix for them. Anybody know if this has been fixed on asterisk? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE
Hi, Iñaki Baz Castillo schrieb: El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió: What is exactly the issue? is the above explained by me? Yes. I was able to step on testing and found out, that our reference system (softsiwtch) is handling it correctly. The asterisk servers we are using as mediagateways are unable to handle it correctly - so I will need a fix for them. Anybody know if this has been fixed on asterisk? I don't understand, why is noa an issue of Asterisk? Isn't a problem in your gateway as it ignores the second 200 (INVITE) while it should send ACK for it and then a BYE? Because Asterisk is my media gw. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE
Hi, system (softsiwtch) is handling it correctly. The asterisk servers we are using as mediagateways are unable to handle it correctly - so I will need a fix for them. Anybody know if this has been fixed on asterisk? I don't understand, why is noa an issue of Asterisk? Isn't a problem in your gateway as it ignores the second 200 (INVITE) while it should send ACK for it and then a BYE? Because Asterisk is my media gw. I don 't understand... first you said that the GW send a call to OpenSIPS and OpenSIPS to both AST1 and AST2 (Asterisk). And the problem is in the GW because ignores the second 200. Am I wrong? The setup is: asterisk(gw) opensips ast1+ast2 No, you are right. So I need to fix that problem on the asterisk(gw) not on AST1 and AST2. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE
Hi, After reading lots of docs and mailing lists, it looks like there is now solution for asterisk available and looks like that might be a long way till then. Maybe its possible to implement that feature in TM? http://www.codename-pineapple.org/doc/html/sip3_dialog_match.html BR Uwe Iñaki Baz Castillo schrieb: El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió: I don 't understand... first you said that the GW send a call to OpenSIPS and OpenSIPS to both AST1 and AST2 (Asterisk). And the problem is in the GW because ignores the second 200. Am I wrong? The setup is: asterisk(gw) opensips ast1+ast2 No, you are right. So I need to fix that problem on the asterisk(gw) not on AST1 and AST2. ok ok. I remember that Olle (chan_sip) commented that Asterisk was tested for this scenario (receiving two 200 for INVITE) in a SIPit, but I don't remember the results... :) Of course, you should enable pedantic=yes so in this way Asterisk is supposed to match To/From tags also. However I would trust it too much... -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE
Hi, From my point of view I have no option but to solve that problem. If you look at this special situation there is no solution to solve it with asterisk without massive rewriting the code - or just hacking it in. So yes from my point of view I would like to have that feature in TM and it might help more people outside. BR Uwe Brett Nemeroff schrieb: Personally,I think broken UACs should behave like broken UACs. If you start making exceptions, then they don't get fixed. Things that are rigged to make them appear to work turn into problems that are hard to detect. or become ignored until they become larger problems. Like Bogdan said, just because you can, doesn't mean you should. :) -Brett On Thu, Oct 22, 2009 at 2:17 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Iñaki Baz Castillo wrote: El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió: Hi, After reading lots of docs and mailing lists, it looks like there is now solution for asterisk available and looks like that might be a long way till then. If you mean the link below take into account that it's chan_sip3 which is not implemented in asterisk at all since nobody wants to support Olee to do that. Asterisk is not interested in SIP. Maybe its possible to implement that feature in TM? I really expect a proxy shouldn't behave as a UAC. Inaki, Uwe, Such a feature is possible to do in TM (technically speaking) , but my doubt is if this is the correct thing to do - because as you said, more or less is not the job of a proxy to sort out such situation (even if it can ;) ). So the question actually is: do we want to be rigorous about what we should or should not do, or we want to add some extra options to to help with interoperability of some broken/stupid entities?? Blue pill ? Red pill ?? :D Regards, Bogdan ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE
Hello Bogdan, Now we changed the behaviour of the UAC. One of them will send a BYE and this is relayed to the PSTN GW which drops the call, since opensips will not handle the BYE locally. So loose_route is done and the BYE is relayed to the PSTN GW. The following is happening: 1) INVITE from PSTN GW 2) parallel forking to ast1 and ast2 (branches z9hG4bK51f6.9afa91c3.1 and z9hG4bK51f6.9afa91c3.0) 3) ast1 sends an 200 OK (branch z9hG4bK51f6.9afa91c3.0) 4) opensips sends an cancel to ast2 (branch z9hG4bK51f6.9afa91c3.1) 5) opensips receives the 200 OK from ast1 and sends an ACK (branch is changing here to z9hG4bK51f6.9afa91c3.3) 6) opensips receives 200 OK from ast2 from the INVITE (branch z9hG4bK51f6.9afa91c3.1)and sends an ACK (branch is changing to z9hG4bK51f6.9afa91c3.3) 7) opensips reives 200 OK from ast2 for the cancel request ( branch z9hG4bK51f6.9afa91c3.1) 8) opensips receives BYE from ast2 with branch z9hG4bK40d1af5d 9) opensips is doing loose_route and sends the BYE to the PSTN GW The only thing I could see on the logs is: WARNING:dialog:dlg_onroute: tight matching failed for BYE with callid='393105a419950c1f265f298914662...@10.20.30.100'/46, ftag='as63949c6e'/10, ttag='as0d1597ca'/10 and direction=0 Oct 21 09:09:15 asne02 /usr/sbin/opensips[15615]: WARNING:dialog:dlg_onroute: dialog identification elements are callid='393105a419950c1f265f298914662...@10.20.30.100'/46, caller tag='as0d1597ca'/10, callee tag='as79debd51'/10 Why is the opensips not handling the BYE locally and only closing one branch? BR UWe Bogdan-Andrei Iancu schrieb: Hi Uwe, Uwe Kastens wrote: Hi Bogdan, So actually both legs do send 200 OK (but one faster than the other)..so there is kind on race between the 200 OK from the slow branch and the CANCEL from OpenSIPS...is this the case? Exactly If so, the UAS will simply reply with negative reply to CANCEL (decline it) and opensips (for INVITE transaction) will not close the second branch as there is a 200 OK (and not a 487) received RFC3261 says that a proxy must send all 200 OK (for a call), even if more than one, to the UAC - the UAC is the one who will decide what branch to keep and it will fire a BYE for the other branch. Could this explan, why only the 2nd Node will get the BYE, if the call is released behind the opensips? yes, because the caller will hung up only one of the callee branch, so the BYE will go to only one of them. The other branch will remain up and will be the ongoing call. Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] send bye in case of parallel forked call
Hi, I have the following requirement: If a from tm generated cancel is answered with a 200 OK I want to send a BYE to the UAC. Is this possible? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] send bye in case of parallel forked call
Hi Alex, Any other option to solve this 200 OK for INVITE relayed after CANCEL issue with opensips and asterisk? http://lists.kamailio.org/pipermail/devel/2008-August/015209.html BR Uwe Alex Balashov schrieb: No. -- Sent from mobile device On Oct 21, 2009, at 9:34 AM, Uwe Kastens ki...@kiste.org wrote: Hi, I have the following requirement: If a from tm generated cancel is answered with a 200 OK I want to send a BYE to the UAC. Is this possible? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE
Hello again, I was wondering if there might be a bug with the correct handling of Cancel in case of receiving and final answer. I will fork one Call to 2 nodes. One node answers a little faster than the other and will get the call. Opensips will send a CANCEL for the other node which is sending a SIP/2.0 200 OK before receiving the CANCEL. So this node is not answering with a 487 but with a 200/OK. Opensips seems to drop the call leg and so the BYE from that node could not be handled. Is this behaviour RFC conform? I will attach one ngrep and one opensips logfile BR Uwe Uwe Kastens schrieb: Hi, I am using opensips to fork calls to UAs which are registrered from different IPs/Ports. If one UA accepts the INVITE the other UAs will get a CANCEL. Now I have one subscriber with 2 asterisk server which asked me to send a BYE after the CANCEL. Otherwise he wants me to send an BYE which could not be processed correctly on the opensips. I am pretty sure, that this kind of handling would not be RFC conform and so its not possible to handle this inside the routing script. Or did I missed something? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 anon_log.gz Description: GNU Zip compressed data anon_ngrep.gz Description: GNU Zip compressed data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE
Hi Bogdan, So actually both legs do send 200 OK (but one faster than the other)..so there is kind on race between the 200 OK from the slow branch and the CANCEL from OpenSIPS...is this the case? Exactly If so, the UAS will simply reply with negative reply to CANCEL (decline it) and opensips (for INVITE transaction) will not close the second branch as there is a 200 OK (and not a 487) received RFC3261 says that a proxy must send all 200 OK (for a call), even if more than one, to the UAC - the UAC is the one who will decide what branch to keep and it will fire a BYE for the other branch. Could this explan, why only the 2nd Node will get the BYE, if the call is released behind the opensips? BR Uwe Regards, Bogdan Uwe Kastens wrote: Hello again, I was wondering if there might be a bug with the correct handling of Cancel in case of receiving and final answer. I will fork one Call to 2 nodes. One node answers a little faster than the other and will get the call. Opensips will send a CANCEL for the other node which is sending a SIP/2.0 200 OK before receiving the CANCEL. So this node is not answering with a 487 but with a 200/OK. Opensips seems to drop the call leg and so the BYE from that node could not be handled. Is this behaviour RFC conform? I will attach one ngrep and one opensips logfile BR Uwe Uwe Kastens schrieb: Hi, I am using opensips to fork calls to UAs which are registrered from different IPs/Ports. If one UA accepts the INVITE the other UAs will get a CANCEL. Now I have one subscriber with 2 asterisk server which asked me to send a BYE after the CANCEL. Otherwise he wants me to send an BYE which could not be processed correctly on the opensips. I am pretty sure, that this kind of handling would not be RFC conform and so its not possible to handle this inside the routing script. Or did I missed something? BR Uwe ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] parallel forking and CANCEL/BYE
Hi, I am using opensips to fork calls to UAs which are registrered from different IPs/Ports. If one UA accepts the INVITE the other UAs will get a CANCEL. Now I have one subscriber with 2 asterisk server which asked me to send a BYE after the CANCEL. Otherwise he wants me to send an BYE which could not be processed correctly on the opensips. I am pretty sure, that this kind of handling would not be RFC conform and so its not possible to handle this inside the routing script. Or did I missed something? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] rewrite user part in all branches from location lookup
Hi, How can I rewrite the user part of all branches I get back from lookup(location)? Do I need to serialize 1st? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] how do you determine first registration of a device
Hi Alex, Can anyone provide advice on how to determine first registration of a phone coming back online in another way? Depends on the phone I would say. From my point of view I would say, that is very hard to track, since some phones acts very strange in that case. Maybe short expire would help? Can you tell more about your setup? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] howto = mediaproxy on lenny
Hi Euge, That might be an option in a testing environment. I don't like the idea to upgrade production systems to debian unstable To much pkg are changing in short period. BR Uwe Euge Serrano schrieb: Hello Uwe, I recommend to upgrade your Lenny to Squeeze, after that you will be able to install it without problems You can follow those easy steps http://www.go2linux.org/how-to-upgrade-from-debian-lenny-to-squeeze Euge On 25/09/09 17:31, Uwe Kastens ki...@kiste.org wrote: Hello, I was wondering why I was able to build mediaproxy packages on debian lenny (stable) but been unable to install them. It looks like, one needs only to build the python-application in the correct version as dpk from source. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] solved = Re: explizit handling auf replyto
Hi Bogdan,
Thanks again for your help.
I had a problem in my script, that causes some loop on one opensips
server. This causes, that the opensips server gets an INVITE from
itself, which was dropped by a security rule. After solving the cause
for that looping the expicit handlling - which caused some other trouble
- was not needed anymore.
BR
Uwe
Bogdan-Andrei Iancu schrieb:
Hi Uwe,
Sorry for the delay - I took a look at the logs and there is what I
understand from there:
- I guess you do a parallel forking as the 486 comes from a second
branch (id 1 starting from 0) (see ;branch=z9hG4bK52ad.7d455a97.1 )
- The TM decides to store the reply without relaying
DBG:tm:relay_reply: branch=1, save=1, relay=-1 TM does not relay it
as probably there is no final reply on the first branch (id = 0).
So, the final reply is not forwarded because not all branches are
completed yet, so TM cannot pick yet a final reply to be sent to UAC.
It is a typical behaviour during parallel forking.
Regards,
Bogdan
Uwe Kastens wrote:
Hello,
Ok. I need to have that forward() in my configuration the get answers
like 404, 486, 487 back to my asterisk. Reading your statements this
should not be possible since I use t_relay for the requests and the
replys should be routed by default.
I will make a trace and post it to the list. One with forward and the
other without.
BR and Thanks in advance
Uwe
Bogdan-Andrei Iancu schrieb:
Uwe,
forward() is a function exclusivly used for REQUESTS - for replies,
nothing needs to be done as OpenSIPS will do it automatically:
1) if the requests was statefully forwarded (via t_relay() ), the
transaction will contain all the info to route back the reply
2) if the requests was statelessly forwarded (via forward() ), the VIA
stack (in received reply) will contain all the info to route back the reply
Regards,
Bogdan
Uwe Kastens wrote:
Hi Bogdan,
I need to route this
replys with an reply_route and forward them explicitly to the pstn
gateway.
and how do you do reply routing ?? replies are automatically routed
based on VIA stack and you cannot influence this from script.
...
if (!t_relay()) {
sl_reply_error();
};
t_on_reply(1);
}
onreply_route[1] {
xlog(L_DBG,== route [1] t_on_failure msg-len from $si:$sp:
$ml \n$mb\n ==\n);
forward(10.20.30.101:5100);
}
BR
Uwe
Regards,
Bogdan
This not as it should be?
BR
Uwe
using forward() forces a stateless behaviour of OpenSIPS. To be able to
catch replies for a requests (what on_reply route does) it requires a
statefull approach. This is the reason it does not work for you.
If you use t_relay() instead of forward(), you should be able to use
failure and onreply routes.
Regards,
Bogdan
Uwe Kastens wrote:
Hello,
I am using opensips 1.5.1 and I have the problem, that busy messages
are
not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
route that message via forward(IP:port) to the media gw in
on_replyto,
busy is processed correctly. If not, the busy is not send to the
mediagw.
So I was wondering if I had to handle some replyto messages?
BR
uwe
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[OpenSIPS-Users] howto = mediaproxy on lenny
Hello, I was wondering why I was able to build mediaproxy packages on debian lenny (stable) but been unable to install them. It looks like, one needs only to build the python-application in the correct version as dpk from source. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips with asterisk = relay REGISTER
Hello Bogdan,
Thank you for the example. In that case the asterisk have to accept the
registration without starting a auth itself?
BR
Uwe
Bogdan-Andrei Iancu schrieb:
Hi Uwe,
If you look at the default opensips script, you have a section (by
default commented out) where the REGISTER requests are authenticated and
if passing the auth doing save(location).
What you have to do is, after the REGISTER auth, instead of pushing the
REGISTER to the local registrar (via save()), simply forward it further
to Asterisk:
if (is_method(REGISTER)) {
# authenticate the REGISTER requests
if (!www_authorize(, subscriber)) {
www_challenge(, 0);
exit;
}
if (!check_to()) {
sl_send_reply(403,Forbidden auth ID);
exit;
}
# auth done - send it to registrar
consume_credentials();
$du = sip:ASTERISK_IP:ASTERISK_PORT;
t_relay();
exit;
}
Regards,
Bogdan
Uwe Kastens wrote:
Hello,
Has anybody a starting point for me to achieve the following:
UAC should register with asterisk put should be pre-authorized with
opensips. I saw an EMail from Bogdan, that this should be possible but
ATM I could only use opensips as a registrar or route all sip messages
through opensips.
Anyone has maybe a hint where to start or maybe an example?
BR
Uwe
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Re: [OpenSIPS-Users] questions about log?
Hi, you can define the syslog facility in opensips.cfg. After that you can put the log to any location. BR Uwe ASHWINI NAIDU schrieb: By default the logging of opensips will be done in */var/log/syslog* in debian systems and */var/log/messages* in redhat based systems 2009/9/9 zhangchao1 zhangchao...@163.com mailto:zhangchao...@163.com Hello everybody, dose anyone know where the log file is? 中国制造,讲述中国60年往事 http://news.163.com/madeinchina/index.html?from=mailfooter ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips with asterisk = relay REGISTER
Hello, Has anybody a starting point for me to achieve the following: UAC should register with asterisk put should be pre-authorized with opensips. I saw an EMail from Bogdan, that this should be possible but ATM I could only use opensips as a registrar or route all sip messages through opensips. Anyone has maybe a hint where to start or maybe an example? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] explizit handling auf replyto
Hi Bogdan,
I need to route this
replys with an reply_route and forward them explicitly to the pstn gateway.
and how do you do reply routing ?? replies are automatically routed
based on VIA stack and you cannot influence this from script.
...
if (!t_relay()) {
sl_reply_error();
};
t_on_reply(1);
}
onreply_route[1] {
xlog(L_DBG,== route [1] t_on_failure msg-len from $si:$sp:
$ml \n$mb\n ==\n);
forward(10.20.30.101:5100);
}
BR
Uwe
Regards,
Bogdan
This not as it should be?
BR
Uwe
using forward() forces a stateless behaviour of OpenSIPS. To be able to
catch replies for a requests (what on_reply route does) it requires a
statefull approach. This is the reason it does not work for you.
If you use t_relay() instead of forward(), you should be able to use
failure and onreply routes.
Regards,
Bogdan
Uwe Kastens wrote:
Hello,
I am using opensips 1.5.1 and I have the problem, that busy messages are
not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
route that message via forward(IP:port) to the media gw in on_replyto,
busy is processed correctly. If not, the busy is not send to the mediagw.
So I was wondering if I had to handle some replyto messages?
BR
uwe
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Re: [OpenSIPS-Users] Newbie To OpenSIPS
Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] explizit handling auf replyto
Hi, Replies are automatically routed only if they are statefully routed. Statefull = t_relay() ? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] explizit handling auf replyto
Hello,
Ok. I need to have that forward() in my configuration the get answers
like 404, 486, 487 back to my asterisk. Reading your statements this
should not be possible since I use t_relay for the requests and the
replys should be routed by default.
I will make a trace and post it to the list. One with forward and the
other without.
BR and Thanks in advance
Uwe
Bogdan-Andrei Iancu schrieb:
Uwe,
forward() is a function exclusivly used for REQUESTS - for replies,
nothing needs to be done as OpenSIPS will do it automatically:
1) if the requests was statefully forwarded (via t_relay() ), the
transaction will contain all the info to route back the reply
2) if the requests was statelessly forwarded (via forward() ), the VIA
stack (in received reply) will contain all the info to route back the reply
Regards,
Bogdan
Uwe Kastens wrote:
Hi Bogdan,
I need to route this
replys with an reply_route and forward them explicitly to the pstn gateway.
and how do you do reply routing ?? replies are automatically routed
based on VIA stack and you cannot influence this from script.
...
if (!t_relay()) {
sl_reply_error();
};
t_on_reply(1);
}
onreply_route[1] {
xlog(L_DBG,== route [1] t_on_failure msg-len from $si:$sp:
$ml \n$mb\n ==\n);
forward(10.20.30.101:5100);
}
BR
Uwe
Regards,
Bogdan
This not as it should be?
BR
Uwe
using forward() forces a stateless behaviour of OpenSIPS. To be able to
catch replies for a requests (what on_reply route does) it requires a
statefull approach. This is the reason it does not work for you.
If you use t_relay() instead of forward(), you should be able to use
failure and onreply routes.
Regards,
Bogdan
Uwe Kastens wrote:
Hello,
I am using opensips 1.5.1 and I have the problem, that busy messages are
not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
route that message via forward(IP:port) to the media gw in on_replyto,
busy is processed correctly. If not, the busy is not send to the mediagw.
So I was wondering if I had to handle some replyto messages?
BR
uwe
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Re: [OpenSIPS-Users] explizit handling auf replyto
Hi, you do not need to do any routing in onreply route at all, in none of the case (stateless or statefull) I will make a trace and post it to the list. One with forward and the other without. make a trace and opensips logs. I have attached opensips.log with debug=9. w_forward_an.gz = with forward wo_forward_an.gz = without forward BR Uwe -- kiste lat: 54.322684, lon: 10.13586 w_forward_an.gz Description: GNU Zip compressed data wo_forward_an.gz Description: GNU Zip compressed data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Newbie To OpenSIPS
Hi, Opensips is a SIP router not a media gateway. So far you will need something that will take care of comvert TDM/PSTN to sip. There should be lot of examples for this kind of setup. BR Uwe Kemp, Larry schrieb: Thank you for the response UK. We are looking at using OpenSIPS on Linux to act as a SIP Router to handoff SIP calls to other carriers that are also using SIP, and explore the use of the agent registration server functionality. I have already previously installed and debugged Asterisk on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or SS7 carriers, or your own Class 4 or 5 environment. Maybe I am misunderstanding as far as what OpenSIPS can actually do? Thanks. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Newbie To OpenSIPS
Hi, Good question and not easy to answer. ACME is expensive AND you will need somebody to configure it in a way you will need it. So as an redundant option your talking about 100-150K. To buy a big name won't prevent you from implementing, bugsearching. My personal opinion: Take less money, look for good consultants and try it with opensource. BR Uwe Kemp, Larry schrieb: Certainly. If I just wanted to pass my SIP to other carriers or have them connect to my SIP customers could I use OpenSIPS for that alone, or would I still need some other sort of session border controller? Larry Kemp -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 1:31 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Yes, this could be an option. But a very expensive one :-) BR Uwe Kemp, Larry schrieb: So I would use OpenSIPS behind say like an Acme Packet http://www.acmepacket.com/ Session Border Controller or a MetaSwitch http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk to those Gateways via SIP and route my customer's VOIP traffic from their Asterisk PBX's to those devices that speak SIP, right? Lars -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 12:19 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Opensips is a SIP router not a media gateway. So far you will need something that will take care of comvert TDM/PSTN to sip. There should be lot of examples for this kind of setup. BR Uwe Kemp, Larry schrieb: Thank you for the response UK. We are looking at using OpenSIPS on Linux to act as a SIP Router to handoff SIP calls to other carriers that are also using SIP, and explore the use of the agent registration server functionality. I have already previously installed and debugged Asterisk on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or SS7 carriers, or your own Class 4 or 5 environment. Maybe I am misunderstanding as far as what OpenSIPS can actually do? Thanks. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Newbie To OpenSIPS
Hi, Yes, this could be an option. But a very expensive one :-) BR Uwe Kemp, Larry schrieb: So I would use OpenSIPS behind say like an Acme Packet http://www.acmepacket.com/ Session Border Controller or a MetaSwitch http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk to those Gateways via SIP and route my customer's VOIP traffic from their Asterisk PBX's to those devices that speak SIP, right? Lars -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 12:19 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Opensips is a SIP router not a media gateway. So far you will need something that will take care of comvert TDM/PSTN to sip. There should be lot of examples for this kind of setup. BR Uwe Kemp, Larry schrieb: Thank you for the response UK. We are looking at using OpenSIPS on Linux to act as a SIP Router to handoff SIP calls to other carriers that are also using SIP, and explore the use of the agent registration server functionality. I have already previously installed and debugged Asterisk on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or SS7 carriers, or your own Class 4 or 5 environment. Maybe I am misunderstanding as far as what OpenSIPS can actually do? Thanks. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] explizit handling auf replyto
Hi Bogdan, Seems that my question was not very clear. I would expect that reply messages would be handled automatically, if I use t_relay. This seems not to happen in my setup. I need to route this replys with an reply_route and forward them explicitly to the pstn gateway. This not as it should be? BR Uwe using forward() forces a stateless behaviour of OpenSIPS. To be able to catch replies for a requests (what on_reply route does) it requires a statefull approach. This is the reason it does not work for you. If you use t_relay() instead of forward(), you should be able to use failure and onreply routes. Regards, Bogdan Uwe Kastens wrote: Hello, I am using opensips 1.5.1 and I have the problem, that busy messages are not passed to the mediagw. Anything else is working (ACK etc.pp.). If I route that message via forward(IP:port) to the media gw in on_replyto, busy is processed correctly. If not, the busy is not send to the mediagw. So I was wondering if I had to handle some replyto messages? BR uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] explizit handling auf replyto
Hello, I am using opensips 1.5.1 and I have the problem, that busy messages are not passed to the mediagw. Anything else is working (ACK etc.pp.). If I route that message via forward(IP:port) to the media gw in on_replyto, busy is processed correctly. If not, the busy is not send to the mediagw. So I was wondering if I had to handle some replyto messages? BR uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] solution found Re: opensips register as ua = Re: forward register
Hi, I am settig for the user in realtime asterisk the ip of the peer to the opensips adress and port. Relevant is to set a username which is needed to find the user at opensips. Users can than register to opensips and be found via Location server. If no user can be found on opensips I will send a 482 back to the asterisk. Works for me. BR Uwe Uwe Kastens schrieb: Hello, I am able to register with opensips or relay register to my asterisk server. What I want is: a) insert/delete the UL of opensips each time a registration/deregistration takes place at the asterisk (Should work, if I use on reply route and insert information with exec or sql). b) make/delete the registration manually with asterisk each the UA registers with opensips. a) could be working, but sound not like a stable solution b) sounds easy, but isn't, since the register needs to be done with one of the listening ports of the opensips. Therefore sipsak is not an option. So the work have to be done from opensips which needs to work as own UA and handle 401 etc.pp. Did I miss a magic feature of opensips here? BR Uwe Uwe Kastens schrieb: Hi Bogdan, So first registrions stays with OpenSIPsm while the next ones are forwarded to Asterisk, right? Nearly. IF the user is registered with opensips, opensips should send a registration to asterisk (and keep at alive). If more registrations are done with the same user account, opensips could send an new registration or send a keepalive. If all registration with opensips are expired or deleted opensips should delete the registration with asterisk. For the one registered with Asterisk, should opensips stay in the middle (as mid-registrar) or Asterisk will directly register the client contact? opensips should stay in the middle. So there is no direct communication between User and asterisk BR Uwe Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, Sorry, trying again: Users should register to opensips. If one successfull registration at opensips for one account is present, opensips should register with asterisk (plus would be, if the data could be different). Background is, that I would like to use the parallel forking with opensips, since my special asterisk config would allow only one simultanios registration per account. Br Uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, Sorry to repeat, but still not clear for me :D... 1) you want to store all registration on OpenSIPS and Asterisk in the same time? 2) from x (= n + m ) received registation, n should be done on OpenSIPS and m forwarded on Asterisk ? Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, If the Account make the 1st sucessfull registration it should be relayed to an asterisk. The 2nd, 3rd ... registration could be relayed. If all registrations have expired or a deleted the registration on the asterisk should be removed as well. Br uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, something like, if you already registered 4 times, the fifth one to be forwarded to Asterisk instead of local registering ? Regards, Bogdan Uwe Kastens wrote: Hi list, I would like to implement the following. A user should be able to register n times with opensips. If one or more registers with a user account exists the opensips should register with that account to an asterisk. I thought a while about this, put there is no way to implement it with internal functions? BR Uwe ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] forward register
Hi Bogdan, If the Account make the 1st sucessfull registration it should be relayed to an asterisk. The 2nd, 3rd ... registration could be relayed. If all registrations have expired or a deleted the registration on the asterisk should be removed as well. Br uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, something like, if you already registered 4 times, the fifth one to be forwarded to Asterisk instead of local registering ? Regards, Bogdan Uwe Kastens wrote: Hi list, I would like to implement the following. A user should be able to register n times with opensips. If one or more registers with a user account exists the opensips should register with that account to an asterisk. I thought a while about this, put there is no way to implement it with internal functions? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] forward register
Hi Bogdan, Sorry, trying again: Users should register to opensips. If one successfull registration at opensips for one account is present, opensips should register with asterisk (plus would be, if the data could be different). Background is, that I would like to use the parallel forking with opensips, since my special asterisk config would allow only one simultanios registration per account. Br Uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, Sorry to repeat, but still not clear for me :D... 1) you want to store all registration on OpenSIPS and Asterisk in the same time? 2) from x (= n + m ) received registation, n should be done on OpenSIPS and m forwarded on Asterisk ? Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, If the Account make the 1st sucessfull registration it should be relayed to an asterisk. The 2nd, 3rd ... registration could be relayed. If all registrations have expired or a deleted the registration on the asterisk should be removed as well. Br uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, something like, if you already registered 4 times, the fifth one to be forwarded to Asterisk instead of local registering ? Regards, Bogdan Uwe Kastens wrote: Hi list, I would like to implement the following. A user should be able to register n times with opensips. If one or more registers with a user account exists the opensips should register with that account to an asterisk. I thought a while about this, put there is no way to implement it with internal functions? BR Uwe ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] forward register
Hi Bogdan, So first registrions stays with OpenSIPsm while the next ones are forwarded to Asterisk, right? Nearly. IF the user is registered with opensips, opensips should send a registration to asterisk (and keep at alive). If more registrations are done with the same user account, opensips could send an new registration or send a keepalive. If all registration with opensips are expired or deleted opensips should delete the registration with asterisk. For the one registered with Asterisk, should opensips stay in the middle (as mid-registrar) or Asterisk will directly register the client contact? opensips should stay in the middle. So there is no direct communication between User and asterisk BR Uwe Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, Sorry, trying again: Users should register to opensips. If one successfull registration at opensips for one account is present, opensips should register with asterisk (plus would be, if the data could be different). Background is, that I would like to use the parallel forking with opensips, since my special asterisk config would allow only one simultanios registration per account. Br Uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, Sorry to repeat, but still not clear for me :D... 1) you want to store all registration on OpenSIPS and Asterisk in the same time? 2) from x (= n + m ) received registation, n should be done on OpenSIPS and m forwarded on Asterisk ? Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, If the Account make the 1st sucessfull registration it should be relayed to an asterisk. The 2nd, 3rd ... registration could be relayed. If all registrations have expired or a deleted the registration on the asterisk should be removed as well. Br uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, something like, if you already registered 4 times, the fifth one to be forwarded to Asterisk instead of local registering ? Regards, Bogdan Uwe Kastens wrote: Hi list, I would like to implement the following. A user should be able to register n times with opensips. If one or more registers with a user account exists the opensips should register with that account to an asterisk. I thought a while about this, put there is no way to implement it with internal functions? BR Uwe ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips register as ua = Re: forward register
Hello, I am able to register with opensips or relay register to my asterisk server. What I want is: a) insert/delete the UL of opensips each time a registration/deregistration takes place at the asterisk (Should work, if I use on reply route and insert information with exec or sql). b) make/delete the registration manually with asterisk each the UA registers with opensips. a) could be working, but sound not like a stable solution b) sounds easy, but isn't, since the register needs to be done with one of the listening ports of the opensips. Therefore sipsak is not an option. So the work have to be done from opensips which needs to work as own UA and handle 401 etc.pp. Did I miss a magic feature of opensips here? BR Uwe Uwe Kastens schrieb: Hi Bogdan, So first registrions stays with OpenSIPsm while the next ones are forwarded to Asterisk, right? Nearly. IF the user is registered with opensips, opensips should send a registration to asterisk (and keep at alive). If more registrations are done with the same user account, opensips could send an new registration or send a keepalive. If all registration with opensips are expired or deleted opensips should delete the registration with asterisk. For the one registered with Asterisk, should opensips stay in the middle (as mid-registrar) or Asterisk will directly register the client contact? opensips should stay in the middle. So there is no direct communication between User and asterisk BR Uwe Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, Sorry, trying again: Users should register to opensips. If one successfull registration at opensips for one account is present, opensips should register with asterisk (plus would be, if the data could be different). Background is, that I would like to use the parallel forking with opensips, since my special asterisk config would allow only one simultanios registration per account. Br Uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, Sorry to repeat, but still not clear for me :D... 1) you want to store all registration on OpenSIPS and Asterisk in the same time? 2) from x (= n + m ) received registation, n should be done on OpenSIPS and m forwarded on Asterisk ? Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, If the Account make the 1st sucessfull registration it should be relayed to an asterisk. The 2nd, 3rd ... registration could be relayed. If all registrations have expired or a deleted the registration on the asterisk should be removed as well. Br uwe Bogdan-Andrei Iancu schrieb: Hi Uwe, something like, if you already registered 4 times, the fifth one to be forwarded to Asterisk instead of local registering ? Regards, Bogdan Uwe Kastens wrote: Hi list, I would like to implement the following. A user should be able to register n times with opensips. If one or more registers with a user account exists the opensips should register with that account to an asterisk. I thought a while about this, put there is no way to implement it with internal functions? BR Uwe ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] forward register
Hi list, I would like to implement the following. A user should be able to register n times with opensips. If one or more registers with a user account exists the opensips should register with that account to an asterisk. I thought a while about this, put there is no way to implement it with internal functions? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] register performance with sipp
Hi, I think your problem is, that you are mixing some settings: client side should be sipp -sf register.xml -inf user.csv ... you can use -sn OR -sf not both together BR Uwe ram schrieb: On Mon, Jun 22, 2009 at 3:17 AM, Iñaki Baz Castillo i...@aliax.net mailto:i...@aliax.net wrote: El Domingo, 21 de Junio de 2009, Uwe Kastens escribió: Hello, Anybody experience with measuring REGISTER performance with sipp? I made some tests and I was wondering how many requests should be possible with opensips/sipps (radius against mysql). It looks like, that I can handle easily 500 REGISTER requests per sec on a XEN Domain (one for sipp and one for opensips), database is on dedicated quad-core server. I found out that my freeradius config caused some trouble (max_requests). What could I expect with that setup? Hi, I did some tests about it (with no radius however): http://lists.opensips.org/pipermail/users/2008-December/002074.html Hi as per the document i have download the shell script and created csv file ( wheren iam intiating sipp transmitter) 1. opensips ( 10.1.1.1 port 5060) 2. Receiver IP 10.1.1.2 port 5060 ./sipp -sn uas -d 0 -p 5060 -i 10.1.1.2 -rsa 10.1.1.1:5060 http://10.1.1.1:5060 -trace_msg and its running 3. transmitter ip 10.1.1.3 sipp -sn uac 10.1.1.2:5060 http://10.1.1.2:5060 -sf script1.xml -inf csv_file -m 5000 -r 100 -rp 1000 -auth_uri 10.1.1.1:5060 http://10.1.1.1:5060 -trace_err -- Scenario Screen [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 100.0(0 ms)/1.000s 5060 5.90 s 300 10.1.1.2:5060(UDP) 0 new calls during 0.859 s period 0 ms scheduler resolution 300 calls (limit 300) Peak was 300 calls, after 3 s 1 Running, 300 Paused, 88 Woken up 0 dead call msg (discarded)0 out-of-call msg (discarded) 3 open sockets Messages Retrans Timeout Unexpected-Msg REGISTER -- 300 837 0 100 -- 0 0 0 0 401 -- 0 0 0 0 REGISTER -- 0 0 0 100 -- 0 0 0 0 200 -- 0 0 0 0 -- Test Terminated - Statistics Screen --- [1-9]: Change Screen -- Start Time | 2009-08-11 06:57:53:657 1249988273.657719 Last Reset Time| 2009-08-11 06:57:58:701 1249988278.701451 Current Time | 2009-08-11 06:57:59:561 1249988279.561174 -+---+-- Counter Name | Periodic value| Cumulative value -+---+-- Elapsed Time | 00:00:00:859 | 00:00:05:903 Call Rate |0.000 cps | 50.822 cps -+---+-- Incoming call created |0 |0 OutGoing call created |0 | 300 Total Call created | | 300 Current Call | 300 | -+---+-- Successful call|0 |0 Failed call|0 |0 -+---+-- Call Length| 00:00:00:000 | 00:00:00:000 -- Test Terminated Server side -- Scenario Screen [1-9]: Change Screen -- Port Total-time Total-calls Transport 5060 162.33 s 300 UDP 0 new calls during 0.080 s period 8 ms scheduler resolution 0 callsPeak was 1 calls, after 102 s 0 Running, 1 Paused, 1 Woken up 837 dead call msg (discarded) 3 open sockets Messages Retrans Timeout Unexpected-Msg
Re: [OpenSIPS-Users] Forcing reinvite for t38 on different pstn gw
Hi, Is it possible to handle reinvites in that way, that I can send them to a special pstn gw? This looks a little tricky, since I need to drop the 1st invite. No, that would not be in the slightest bit compatible with SIP protocol mechanics as described per the RFC. The initial INVITE establishes the dialog, and without that initial request there cannot be sequential in-dialog requests - and therefore, no re-INVITEs. [...] The only way you can pull this off is to decide in advance whether the call needs to go through a special PSTN GW when routing the initial INVITE. Ok, thats not possible with T38, since the codec is 1st established as normale codec. If one of the devices gets a fax ton it will iniitate a reinvite with t38. Thanks BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] backport for dialog profile for 1.5
Thomas, Thomas Gelf schrieb: Even a backport wouldn't help you, as profiles, vars and flags can change multiple times during a dialog they are not stored to db unless you restart OpenSIPS - to let dialogs cleanly survive a stop /start sequence. Good point. Ok so a backport make no sense. What you want to retrieve is however available via MI-modules, I'm for example preferring the XML-RPC one. I will try this out. Looks simple from the docs - is it that simple? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Multi-homed systems
Hi, set mhomed=1 ? BR Uwe Gordon Ross schrieb: I've setup a multi-homed OpenSIPS system with two ethernet interfaces (eth0 eth1). The problem I've got, is that regardless of which physical interface the packets leave the box, they always have the same source IP address. I.e. Packets leaving eth0 have the IP address of eth1. Is there any way to control this, and either tell OpenSIPS to use the interface IP address, or to specify, for this route, use this source IP address ? GTG ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Fwd: opensips+asterisk call dropping in 20 seconds
Hi, You are missing some ACKs in one direction. Looks like you missed some record_route loose_route entries in your config? Wireshark/ngrep is your best friend :-) Good luck BR Uwe ram schrieb: On Tue, Jul 7, 2009 at 10:46 PM, cesar.fiestas fiestas.ce...@gmail.com mailto:fiestas.ce...@gmail.com wrote: In my opinion the 20 sec drop call is due to a NAT issue, check your NAT setup and or configuration All are Public IP's any other suggestions Ram ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is opensips a front end to asterisk?
You are right. We all started from the same point and asked questions to learn a lot. The more specific the question is, the better the answer would match. I think your setup is not new, but it depends on your requirement and your setup. BTW: What was the initial question? :) BR Uwe li...@grounded.net schrieb: I love how joining pretty much any new mailing list and asking initial questions leads to the typical 'you should realize how difficult this is' replies. That's nothing new since there are countless folks who have aspirations without the follow through but not everyone. And really, all of you learned the same way, asking sometimes stupid, but a lot of questions, reading, playing with and getting to know, the software. Well, maybe not the developers :). Anyhow, I'd still love to see some feedback on my original question. Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] sip2pstn: P-Asserted-Identity and P-Preferred-Identity
Hi list, This is not exactly a opensips issue. I don't if anybody give me a hint. Until today I was very sure the the P-Asserted-Identity is trusted and the P-Preferred-Identity is untrusted. So it is wise to map the asserted to the pstn number which is the carrier trusted (network provided) and the preferred is a number for clip no screening. I discussed with a vendor which will send me a ddi-number for a pbx as asserted and the main number as preferred. The RFC is not very clear in that point - or did I read the wrong ones. BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] backport for dialog profile for 1.5
Hello, I miss the option to find out via db how much calls are online for defined dialog profiles. There is an option in 1.6 where profiles is usable via database. Would there be a backport to 1.5 for that field? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Error using MySQL with opensips
Hi, Maybe you have forgotten to create the tables for opensips? opensipsdbctl create db name or db_path, optional .(creates a new database) opensips is missing the version table BR Uwe anurag schrieb: Hi, I’m new to OpenSIPs. I want to use OpenSIPs as Presence server. I’ve compiled OpenSIPs (1.5.1) for presence server and installed it on Linux. Also, I’ve enabled presence parameters in config file with mysql database. My DB details are: User- opensips Passwd-opensipsrw DBname-test While initializing opensips in this config I’m getting following error: Jul 2 03:00:43 [16521] DBG:db_mysql:db_mysql_new_connection: server version is 5.0.46-enterprise-log Jul 2 03:00:43 [16521] ERROR:db_mysql:db_mysql_submit_query: driver error on query: Table 'test.version' doesn't exist Jul 2 03:00:43 [16521] ERROR:core:db_do_query: error while submitting query Jul 2 03:00:43 [16521] ERROR:core:db_table_version: error in db_query Jul 2 03:00:43 [16521] ERROR:core:db_check_table_version: querying version for table presentity Jul 2 03:00:43 [16521] ERROR:presence:mod_init: error during table version check Jul 2 03:00:43 [16521] ERROR:core:init_mod: failed to initialize module presence Jul 2 03:00:43 [16521] ERROR:core:main: error while initializing modules Jul 2 03:00:43 [16521] DBG:presence_xml:destroy: start Here is my config (from opensips.cfg): # - presence params - /* uncomment the following lines if you want to enable presence */ #modparam(presence|presence_xml, db_url, mysql://opensips:opensip...@192.168.8.76/opensips) modparam(presence|presence_xml, db_url, mysql://opensips:opensip...@192.168.8.76/test) modparam(presence_xml, force_active, 1) modparam(presence, server_address, sip:192.168.8.83:6060) Pls help! Thanx in advance, Anurag ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Error using MySQL with opensips
Hi, opensipsdbctl create Therefore you will need configure the opensipsctlrc file. BR Uwe Anurag Guru schrieb: Thanx BR. Could you pls tell me from where I can find info on usage for opensipsdbctl? or some related documentation. Thanx, Anurag Hi, Maybe you have forgotten to create the tables for opensips? opensipsdbctl create db name or db_path, optional .(creates a new database) opensips is missing the version table BR -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Error using MySQL with opensips
Hi, So you can connect via cli: mysql -u root -p -h 192.168.8.83 and you can create/drop a database? mysqladmin create opensips -p -u root -h 192.168.8.83 The error looks like a mysql access right problem BR Uwe anurag schrieb: Thanx Uwe, I've modified opensipsctlrc for DB details as following: === DBENGINE=MYSQL ## database host DBHOST=192.168.8.83 ## database name (for ORACLE this is TNS name) DBNAME=opensips # database path used by dbtext or db_berkeley # DB_PATH=/usr/local/etc/opensips/dbtext ## database read/write user DBRWUSER=opensips ## password for database read/write user DBRWPW=opensipsrw === However, now when I'm running opensipsdbctl create it is giving access denied error: = [r...@bplinux90 sbin]# ./opensipsdbctl create MySQL password for root: INFO: test server charset ERROR 1045 (28000): Access denied for user 'root'@'bplinux90' (using password: YES) ERROR 1045 (28000): Access denied for user 'root'@'bplinux90' (using password: YES) Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /root/opensips/install/lib/opensips/opensipsctl/opensipsdbctl.mysql: line 114: [: =: unary operator expected INFO: creating database opensips ... ERROR 1045 (28000): Access denied for user 'root'@'bplinux90' (using password: YES) ERROR: Creating core database and grant privileges failed! [r...@bplinux90 sbin]# = Though, I'm able to login as root user MySQL database by supplying configured password (mysql). Pls tell if there is any other setting that I need to do to resolve this issue. Thanx in advance, Anurag -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: Friday, July 03, 2009 1:39 PM To: Anurag Guru Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Error using MySQL with opensips Hi, opensipsdbctl create Therefore you will need configure the opensipsctlrc file. BR Uwe Anurag Guru schrieb: Thanx BR. Could you pls tell me from where I can find info on usage for opensipsdbctl? or some related documentation. Thanx, Anurag Hi, Maybe you have forgotten to create the tables for opensips? opensipsdbctl create db name or db_path, optional .(creates a new database) opensips is missing the version table BR -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] parallel forking vs. serial
Hello list, I was wondering if somebody has a hint for me. I have a requirement to connect a customer with two UAs. It should be a failover solution. I would like to do this via normal registrar functionality without loosing parallel forking for all other customers. The 1st idea was to work with the q-value. But I think this works only in an single branch env. How could that be done in the easiest way? (My prefered way would be to tell them to delay the call pickup on the failover UA) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 408 Timeout with X-Lite
Gordon, Which version of opensips you are testing with? Have you enabled multi domain support for register, urloc etc.pp.? Maybe you can post the head of your config. BR Uwe Gordon Ross schrieb: Starting with an empty DB, I created a domain and I created a subscriber in OpenSIPS. # opensipsctl domain add blah # opensipsctl add 2...@blah 1234 Looking at the database, the user domain are in the tables. Firing up X-Lite, I put the following in as the SIP account details: Display Name: Gordon User name: 2345 Password: 1234 Authorisation user name: Domain: blah X-Lite comes back with a 408 - Request Timeout message. Doing a tcpdump shows a batch of REGISTER packets. After a while, the server responds with 408 Request Timeout packets. Messages then starts getting: /usr/local/sbin/opensips[14893]: ERROR:registrar:update_contacts: invalid cseq for aor 2345 Doing a google, it seems that this problem appears when there is already an entry in the locations table. But when I first start up OpenSIPS XLite, the locations table is empty ! After XLite it started, I do get entries in the locations table. One strange thing is that the domain column is blank. (But there are entries in most of the other columns) Can someone enlighten me as to the stupid mistake I'm making ? Thanks, GTG ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] 408 Timeout with X-Lite
Gordon, Strange so far. I cannot see any wrong configuration on a 1st view. Could you see if auth is working and only writing to the USRLOC is failing? (Maybe put some xlog statements around the register part). The error ocurs by saving the contact into the DB. Have you tried with another client? BR Uwe Gordon Ross schrieb: On 02/07/2009 09:13, Uwe Kastens ki...@kiste.org wrote: Which version of opensips you are testing with? 1.5.1 Have you enabled multi domain support for register, urloc etc.pp.? Yes. However, in the process of posting the config (below) I noticed that I hadn't un-commented the line: modparam(alias_db|auth_db|usrloc|uri_db, use_domain, 1) I've uncommented this, cleaned out the locations table and re-started OpenSIPS then X-Lite. The locations table now has the domain column completed, but I'm still getting a 408 :-( Maybe you can post the head of your config. I hope this is enough. Let me know if you want any more. Ta. GTG ### Modules Section #set module path mpath=/usr/local/lib64/opensips/modules/ /* uncomment next line for MySQL DB support */ #loadmodule db_mysql.so loadmodule db_postgres.so loadmodule signaling.so loadmodule sl.so loadmodule tm.so loadmodule rr.so loadmodule maxfwd.so loadmodule usrloc.so loadmodule registrar.so loadmodule textops.so loadmodule mi_fifo.so loadmodule uri_db.so loadmodule uri.so loadmodule xlog.so loadmodule acc.so /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule auth.so loadmodule auth_db.so /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule alias_db.so /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see multi-module params section ) */ loadmodule domain.so /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule presence.so #loadmodule presence_xml.so # - setting module-specific parameters --- # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) # - rr params - # add value to ;lr param to cope with most of the UAs modparam(rr, enable_full_lr, 1) # do not append from tag to the RR (no need for this script) modparam(rr, append_fromtag, 0) # - registrar params - modparam(registrar, method_filtering, 1) /* uncomment the next line to disable parallel forking via location */ # modparam(registrar, append_branches, 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam(registrar, max_contacts, 10) # - usrloc params - #modparam(usrloc, db_mode, 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam(usrloc, db_mode, 1) modparam(usrloc, db_url, postgres://opensips:opensip...@localhost/opensips) # - uri_db params - /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam(uri_db, use_uri_table, 0) modparam(uri_db, db_url, ) # - acc params - /* what sepcial events should be accounted ? */ modparam(acc, early_media, 1) modparam(acc, report_ack, 1) modparam(acc, report_cancels, 1) /* by default ww do not adjust the direct of the sequential requests. in rr module */ modparam(acc, detect_direction, 0) /* account triggers (flags) */ modparam(acc, failed_transaction_flag, 3) modparam(acc, log_flag, 1) modparam(acc, log_missed_flag, 2) /* uncomment the following lines to enable DB accounting also */ modparam(acc, db_flag, 1) modparam(acc, db_missed_flag, 2) # - auth_db params - /* uncomment the following lines if you want to enable the DB based authentication */ modparam(auth_db, calculate_ha1, yes) modparam(auth_db, password_column, password) modparam(auth_db, db_url, # mysql://opensips:opensip...@localhost/opensips) postgres://opensips:opensip...@localhost/opensips) modparam(auth_db, load_credentials, ) # - alias_db params - /* uncomment the following lines if you want to enable the DB based aliases */ modparam(alias_db, db_url, # mysql://opensips:opensip...@localhost/opensips) postgres://opensips:opensip...@localhost/opensips) # - domain params - /* uncomment the following lines to enable multi-domain detection support */ modparam(domain, db_url, # mysql://opensips:opensip...@localhost/opensips) postgres://opensips:opensip...@localhost/opensips) modparam(domain, db_mode, 1) # Use caching # - multi-module params - /* uncomment the following line if you want to enable multi-domain support
Re: [OpenSIPS-Users] 1.6 and mysql = crash Re: DIALOG not deletedon BYE
Brett, Which error do you mean? The one with the dialogs which are not deleted from RAM? BR Uwe br...@nemeroff.com schrieb: Wouldbthis error manifest without the registrar module? I saw dialog counts incorrect on a stateful loadbalancer I built and was hoping this had something to do with it. -Brett Sent from my Verizon Wireless BlackBerry -Original Message- From: Bogdan-Andrei Iancu bog...@voice-system.ro Date: Wed, 01 Jul 2009 18:38:34 To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] 1.6 and mysql = crash Re: DIALOG not deleted on BYE Just to keep the list informed - the error had nothing to do with mysql, was because of the latest changes on the REGISTRAR module - the bug was found and fixed on SVN. Regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, I see the core was not generated, so no bt :(can you reproduce the crash? can you get a core file and a bt ? Thanks and regards, Bogdan Uwe Kastens wrote: Bogdan, Sorry for bothering again. I tried the latest trunk from svn and opensips is dying after accessing the mysql db. I will attach the trace. BR Uwe Bogdan-Andrei Iancu schrieb: OK - with the fix from SVN you should be able to call loose_route() as many times you want without any risk - just let me know if it works as expected. Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, Again, thanks a lot for your help. The loose_route() seems to cause the problem, but somehow its needed to pass byes correctly to the UA. So I need to work a little on my skript. I will try the 1.6 ASAP and let you know the result. BR Uwe Bogdan-Andrei Iancu schrieb: If you could test, a fix is available on 1.6 (trunk) version - if ok, I will do the backport. Regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Thanks for the traces. Looking at the opensips logs, I say you do loose_route() twice for the ACK which looks twice for the dialog and increase the ref twice for the dialogthis is why the ref never gets back to 0 to allow the dialog to be destroyed.. Could you confirm this for me ? even if it's a script error , the dialog module should cope with it..I will look for a fix. Thanks and regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Uwe Kastens wrote: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. yes, it sounds like. Unfort I was not able to find out what the states and the events means. You can find the meaning of each state in: modules/dialog/dlg_hash.h So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? I can try : ) could you (privately if needed) please send me the the full logs for the entire call (debug=6) - for the non working part. Thanks and regards, Bogdan BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact
Re: [OpenSIPS-Users] Is opensips a front end to asterisk?
Hi, I will try to answer some questions. I can say, that I am working with a kind of load balancing / redundancy for asterisk servers with opensips. Its working perfectly. li...@grounded.net schrieb: I've come across this project a few times but have been having a bit of a time confirming just what the project does. I thought perhaps the best way would be to join the list and ask. My task is to put together a scalable asterisk based pbx system. Because the boxes will initially have more than they really should installed on them, we need to limit the number of users per box to perhaps 50. Right now, the plan calls for every box to have a second one for redundancy. I was planning on manually redirecting connections (for now) but it sounds like opensips could take care of a number of issues. I have multiple providers (WANs) at one location but was thinking that for highest reliability, that I might have three locations to be safe unless there are better ideas. One would be the location where the initial user connection is made, such as a proxy/load balancer. Then, two separate physical locations and networks for redundancy. The front end could use both sites as needed but if something went down, could re-route users/sessions to the redundant location. This of course is where my questions about opensips come in. -From what I can tell, opensips could act as a pbx on it's own but it can act as a proxy/load balancer/gateway to asterisk systems as well. Yes. But its a question how you will define PBX. There are several modules for opensips which could do some PBX things - I never worked with them. -If this is the case, would there be a way of creating a distributed environment, like as in a web server farm, making scaling quite easy. If you are talking about scaling in a way that you can add more asterisk servers to have more users, yes. But there might be some limitations. -Does opensips handle only new incoming connections or could it actually move sessions from a down server to another which is still up? As far as I know there is no way to switch an active connection from one server to the other. And to be honest I do not know any payable commercical solution that is able to handle this. -Would there be any control, or even any need depending on how the back end can be set up, by which to control which pbx/pair that someone registers to? Hard to tell. You can use opensips and route any request by using different tests, rewrite URIs etc.pp. But I think you might want to have the users register on your asterisk. -Would I have some method of controlling how many people can register on any one box? Hmm. Everything is possible :) There might be a better way but I would start sharing such Load information before routing a register request to an asterisk. So you could check via sql how many users are already registered to your asterisk and choose the one with the lowest amount of UA. I think the openips LB module is more designed for INVITES. I have implemented a solution with several asterisk servers and opensip servers as a kind of carrier solution. BR Uwe Thank you very much for this information as it will help to first understand what the project can do. Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DIALOG not deleted on BYE
Hi Bogdan, The problem is fixed with the changes in dlg_handlers.c rev 5806. Thanks a lot BR Uwe Bogdan-Andrei Iancu schrieb: OK - with the fix from SVN you should be able to call loose_route() as many times you want without any risk - just let me know if it works as expected. Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, Again, thanks a lot for your help. The loose_route() seems to cause the problem, but somehow its needed to pass byes correctly to the UA. So I need to work a little on my skript. I will try the 1.6 ASAP and let you know the result. BR Uwe Bogdan-Andrei Iancu schrieb: If you could test, a fix is available on 1.6 (trunk) version - if ok, I will do the backport. Regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Thanks for the traces. Looking at the opensips logs, I say you do loose_route() twice for the ACK which looks twice for the dialog and increase the ref twice for the dialogthis is why the ref never gets back to 0 to allow the dialog to be destroyed.. Could you confirm this for me ? even if it's a script error , the dialog module should cope with it..I will look for a fix. Thanks and regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Uwe Kastens wrote: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. yes, it sounds like. Unfort I was not able to find out what the states and the events means. You can find the meaning of each state in: modules/dialog/dlg_hash.h So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? I can try : ) could you (privately if needed) please send me the the full logs for the entire call (debug=6) - for the non working part. Thanks and regards, Bogdan BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ 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] How to insert the IP address of user in radius request.
Hi, I am facing a similar situation. We need to verify that a REGISTER comes from the same srcip we have configured in our database. I am thinking about doing this by making a select into an AVP and verfying the value of the AVP with the $si. If this is successfull the UA would be saved into the location and/or would be able to make a call. This should be possible with radius_avp as well. Looking at performance I would make the DIGEST Auth 1st and if this is succesfull check the IPs. BR uwe Tung Tran schrieb: Hi Mr. Bogdan We need it for IP authorize besides DIGEST auth, that is not standard anyway but business requirements. We use MSSQL to do DIGEST authorize and we need an extra security layer based on source IP, that is also a request by govements in my contry. So last but not lease, I would like someone can help me how to add this feature as soon ass possible Thank you very much for your help Tung - Original Message - From: Bogdan-Andrei Iancu bog...@voice-system.ro To: Tung Tran tr.t...@gmail.com Cc: users@lists.opensips.org Sent: Friday, June 26, 2009 2:24 AM Subject: Re: [OpenSIPS-Users] How to insert the IP address of user in radius request. Hi Tung, I see the difference - unfortunately there is no way (at the moment) to add custom info to the RADIUS auth header, but it should be an extension that can be done - out of curiosity? why do you need this in the AUTH request, as this info is not used in the DIGEST auth. Regards, Bogdan Tung Tran wrote: Dear Mr. Bogdan, I know we can insert the source IP address in account request before sending it to Radius, however can I insert it in AUTHORIZE request instead? Thank you very much for your reply. Tung - Original Message - From: Bogdan-Andrei Iancu bog...@voice-system.ro To: Tung Tran tr.t...@gmail.com Cc: users@lists.opensips.org Sent: Tuesday, June 23, 2009 6:04 PM Subject: Re: [OpenSIPS-Users] How to insert the IP address of user in radius request. Hi Tung, First of all you should upgrade to 1.5 version (see http://www.opensips.org/Resources/Downloads). For your problem, use extra accounting - you can account whatever extra info you want. See: http://www.opensips.org/html/docs/modules/1.5.x/acc.html#ACC-extra-id To get the source IP, use the $si pseudo-variable (see http://www.opensips.org/Resources/DocsCoreVar15#toc71). Regards, Bogdan Tung Tran wrote: Hi all, I get a request to insert the public IP address of the sip softphone or IP Phone/ATA (end-point) in the Radius request sending to Radius server. I am thinking about to mod the auth_radius module to insert that IP in SIP-URI-User field, likely this one: Original Sip-Uri-User = 985512405 After mod: Sip-Uri-User = 985512...@1.2.3.4 Where 1.2.3.4 is the IP of SIP end-point, not the IP address of Opensips/Opensers servers. But I dont know where I should play with. Any one had done it before or know where we can edit, pls help me. BTW, I am using openser 1.2.2 version. Thanks in advance Tung ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] src ip check on Register = Re: How to insert the IP address of user in radius request.
Hi,
this is the script part, that is doing the job. ATM just only logging
loadmodule avpops.so
# avpops
modparam(avpops, db_url,mysql://:x...@abcd.domain.de/testdb)
if (method==REGISTER) {
if (!radius_www_authorize()) {
www_challenge(, 0);
exit;
};
avp_db_query(select ip from src_ip where number='$au',
$avp(s:srcip));
if ($avp(s:srcip)!=$si){
xlog($au should have SRC_IP $avp(s:srcip), but has $si);
}
save(location) ;
exit;
}
BR
Uwe
Uwe Kastens schrieb:
Hi,
I am facing a similar situation. We need to verify that a REGISTER comes
from the same srcip we have configured in our database. I am thinking
about doing this by making a select into an AVP and verfying the value
of the AVP with the $si. If this is successfull the UA would be saved
into the location and/or would be able to make a call.
This should be possible with radius_avp as well.
Looking at performance I would make the DIGEST Auth 1st and if this is
succesfull check the IPs.
BR
uwe
Tung Tran schrieb:
Hi Mr. Bogdan
We need it for IP authorize besides DIGEST auth, that is not standard anyway
but business requirements.
We use MSSQL to do DIGEST authorize and we need an extra security layer
based on source IP, that is also a request by govements in my contry.
So last but not lease, I would like someone can help me how to add this
feature as soon ass possible
Thank you very much for your help
Tung
- Original Message -
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: Tung Tran tr.t...@gmail.com
Cc: users@lists.opensips.org
Sent: Friday, June 26, 2009 2:24 AM
Subject: Re: [OpenSIPS-Users] How to insert the IP address of user in radius
request.
Hi Tung,
I see the difference - unfortunately there is no way (at the moment) to
add custom info to the RADIUS auth header, but it should be an extension
that can be done - out of curiosity? why do you need this in the AUTH
request, as this info is not used in the DIGEST auth.
Regards,
Bogdan
Tung Tran wrote:
Dear Mr. Bogdan,
I know we can insert the source IP address in account request before
sending it to Radius, however can I insert it in AUTHORIZE request
instead?
Thank you very much for your reply.
Tung
- Original Message - From: Bogdan-Andrei Iancu
bog...@voice-system.ro
To: Tung Tran tr.t...@gmail.com
Cc: users@lists.opensips.org
Sent: Tuesday, June 23, 2009 6:04 PM
Subject: Re: [OpenSIPS-Users] How to insert the IP address of user in
radius request.
Hi Tung,
First of all you should upgrade to 1.5 version (see
http://www.opensips.org/Resources/Downloads).
For your problem, use extra accounting - you can account whatever extra
info you want. See:
http://www.opensips.org/html/docs/modules/1.5.x/acc.html#ACC-extra-id
To get the source IP, use the $si pseudo-variable (see
http://www.opensips.org/Resources/DocsCoreVar15#toc71).
Regards,
Bogdan
Tung Tran wrote:
Hi all,
I get a request to insert the public IP address of the sip softphone or
IP Phone/ATA (end-point) in the Radius request sending to Radius
server.
I am thinking about to mod the auth_radius module to insert that IP in
SIP-URI-User field, likely this one:
Original
Sip-Uri-User = 985512405
After mod:
Sip-Uri-User = 985512...@1.2.3.4
Where 1.2.3.4 is the IP of SIP end-point, not the IP address of
Opensips/Opensers servers.
But I dont know where I should play with.
Any one had done it before or know where we can edit, pls help me.
BTW, I am using openser 1.2.2 version.
Thanks in advance
Tung
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Re: [OpenSIPS-Users] mysql database connection error
Hi, Hmmm. - password to long Have you mysqlaccess on the system? This is how the output looks on my testsystem mysqlaccess localhost opensips opensips -U root -P mysqlaccess Version 2.06, 20 Dec 2000 By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be) Changes by Steve Harvey (s...@vex.net) This software comes with ABSOLUTELY NO WARRANTY. Password for MySQL superuser root: Access-rights for USER 'opensips', from HOST 'localhost', to DB 'opensips' +-+---+ +-+---+ | Select_priv | Y | | Lock_tables_priv | Y | | Insert_priv | Y | | Execute_priv| Y | | Update_priv | Y | | Repl_slave_priv | N | | Delete_priv | Y | | Repl_client_priv | N | | Create_priv | Y | | Create_view_priv | Y | | Drop_priv | Y | | Show_view_priv | Y | | Reload_priv | N | | Create_routine_priv | Y | | Shutdown_priv | N | | Alter_routine_priv | Y | | Process_priv| N | | Create_user_priv | N | | File_priv | N | | Ssl_type| ? | | Grant_priv | N | | Ssl_cipher | ? | | References_priv | Y | | X509_issuer | ? | | Index_priv | Y | | X509_subject| ? | | Alter_priv | Y | | Max_questions | 0 | | Show_db_priv| N | | Max_updates | 0 | | Super_priv | N | | Max_connections | 0 | | Create_tmp_table_priv | Y | | Max_user_connections | 0 | +-+---+ +-+---+ NOTE:A password is required for user `opensips' :-( The following rules are used: db: '%','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y' host : 'Not processed: host-field is not empty in db-table.' user : '%','opensips','*30C28A928E2BE5EFD59FF20CB8705B31ACCF3730','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0' BUGs can be reported by email to b...@mysql.com BR Uwe Brett Nemeroff schrieb: Yeah, it's in there. I'm really puzzled. This should be the easy part. ;) Any other ideas? On Tue, Jun 30, 2009 at 10:25 AM, Uwe Kastens ki...@kiste.org mailto:ki...@kiste.org wrote: Hi, looks good for me. Did you reload mysql after grant or flush priv? Should be, otherwise you wont be able to connect via mysql client. Your /etc/hosts have the entry for localhost? BR Uwe Brett Nemeroff schrieb: yeah, I tried localhost, 127.0.0.1, and the actual ip (I usually use localhost) here's my connect string: modparam(auth_db, db_url, mysql://opensips:23u83fwhw...@localhost/opensips) Here's the mysql grant: GRANT ALL ON opensips.* TO 'opensips'@'localhost' IDENTIFIED BY '23u83fwhwkgh'; On Tue, Jun 30, 2009 at 9:56 AM, Uwe Kastens ki...@kiste.org mailto:ki...@kiste.org mailto:ki...@kiste.org mailto:ki...@kiste.org wrote: Brett, Could you post your URL from your config please? 127.0.0.1 is not the same as localhost! BR Uwe Hey all, sorry for such a noob question here, but I just can't figure out what I'm doing wrong.. I'm getting the error: Jun 30 15:36:33 nguenj297 /usr/local/sbin/opensips[10159]: ERROR:db_mysql:db_mysql_new_connection: driver error(1045): Access denied for user 'opensips'@'localhost' (using password: YES) So of course, I checked the usernames and passwords.. I tried logging in manually with: mysql -u opensips -h localhost -p opensips (with the same password of course) And it works fine.. so I'm not sure where the hangup is.. Where should I debug this? Thanks, Brett ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org mailto:Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 -- kiste lat: 54.322684, lon: 10.13586 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mysql database connection error
Hmm, And you did change the host in the opensips url from localhost to 127.0.0.1? For a test change the 127.0.0.1 in the mysql to % and try again. And mabye change the password with update user set password=password(yourpassword) where user=opensips; Your password entry looks like from a elder version. BR Uwe Brett Nemeroff schrieb: Here is what I got: password too long? really? it's not that long.. shrug mysqlaccess Version 2.06, 20 Dec 2000 By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be) Changes by Steve Harvey (s...@vex.net mailto:s...@vex.net) This software comes with ABSOLUTELY NO WARRANTY. Password for MySQL superuser root: Access-rights for USER 'opensips', from HOST 'localhost', to DB 'opensips' +-+---+ +-+---+ | Select_priv | Y | | Lock_tables_priv | Y | | Insert_priv | Y | | Execute_priv| Y | | Update_priv | Y | | Repl_slave_priv | N | | Delete_priv | Y | | Repl_client_priv | N | | Create_priv | Y | | Create_view_priv | Y | | Drop_priv | Y | | Show_view_priv | Y | | Reload_priv | N | | Create_routine_priv | Y | | Shutdown_priv | N | | Alter_routine_priv | Y | | Process_priv| N | | Create_user_priv | N | | File_priv | N | | Ssl_type| ? | | Grant_priv | N | | Ssl_cipher | ? | | References_priv | Y | | X509_issuer | ? | | Index_priv | Y | | X509_subject| ? | | Alter_priv | Y | | Max_questions | 0 | | Show_db_priv| N | | Max_updates | 0 | | Super_priv | N | | Max_connections | 0 | | Create_tmp_table_priv | Y | | Max_user_connections | 0 | +-+---+ +-+---+ NOTE:A password is required for user `opensips' :-( The following rules are used: db: '127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y' host : 'Not processed: host-field is not empty in db-table.' user : '127.0.0.1','opensips','641b9f69397f5d64','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0' On Tue, Jun 30, 2009 at 10:41 AM, Uwe Kastens ki...@kiste.org mailto:ki...@kiste.org wrote: Hi, Hmmm. - password to long Have you mysqlaccess on the system? This is how the output looks on my testsystem mysqlaccess localhost opensips opensips -U root -P mysqlaccess Version 2.06, 20 Dec 2000 By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be) Changes by Steve Harvey (s...@vex.net mailto:s...@vex.net) This software comes with ABSOLUTELY NO WARRANTY. Password for MySQL superuser root: Access-rights for USER 'opensips', from HOST 'localhost', to DB 'opensips' +-+---+ +-+---+ | Select_priv | Y | | Lock_tables_priv | Y | | Insert_priv | Y | | Execute_priv| Y | | Update_priv | Y | | Repl_slave_priv | N | | Delete_priv | Y | | Repl_client_priv | N | | Create_priv | Y | | Create_view_priv | Y | | Drop_priv | Y | | Show_view_priv | Y | | Reload_priv | N | | Create_routine_priv | Y | | Shutdown_priv | N | | Alter_routine_priv | Y | | Process_priv| N | | Create_user_priv | N | | File_priv | N | | Ssl_type| ? | | Grant_priv | N | | Ssl_cipher | ? | | References_priv | Y | | X509_issuer | ? | | Index_priv | Y | | X509_subject| ? | | Alter_priv | Y | | Max_questions | 0 | | Show_db_priv| N | | Max_updates | 0 | | Super_priv | N | | Max_connections | 0 | | Create_tmp_table_priv | Y | | Max_user_connections | 0 | +-+---+ +-+---+ NOTE:A password is required for user `opensips' :-( The following rules are used: db: '%','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y' host : 'Not processed: host-field is not empty in db-table.' user : '%','opensips','*30C28A928E2BE5EFD59FF20CB8705B31ACCF3730','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0' BUGs can be reported by email to b...@mysql.com mailto:b...@mysql.com BR Uwe Brett Nemeroff schrieb: Yeah, it's in there. I'm really puzzled. This should be the easy part. ;) Any other ideas? On Tue, Jun 30, 2009 at 10:25 AM, Uwe Kastens ki
Re: [OpenSIPS-Users] mysql database connection error
Hi Brett, Could be broken libs. I would start opensips with strace and look for errors. Which OS are you using? BR Uwe Brett Nemeroff schrieb: Still can't connect :( What could I be doing wrong? I wonder if it's the mysql client libs somehow?! Access-rights for USER 'opensips', from HOST 'localhost', to DB 'opensips' +-+---+ +-+---+ | Select_priv | Y | | Lock_tables_priv | Y | | Insert_priv | Y | | Execute_priv| Y | | Update_priv | Y | | Repl_slave_priv | N | | Delete_priv | Y | | Repl_client_priv | N | | Create_priv | Y | | Create_view_priv | Y | | Drop_priv | Y | | Show_view_priv | Y | | Reload_priv | N | | Create_routine_priv | Y | | Shutdown_priv | N | | Alter_routine_priv | Y | | Process_priv| N | | Create_user_priv | N | | File_priv | N | | Ssl_type| ? | | Grant_priv | N | | Ssl_cipher | ? | | References_priv | Y | | X509_issuer | ? | | Index_priv | Y | | X509_subject| ? | | Alter_priv | Y | | Max_questions | 0 | | Show_db_priv| N | | Max_updates | 0 | | Super_priv | N | | Max_connections | 0 | | Create_tmp_table_priv | Y | | Max_user_connections | 0 | +-+---+ +-+---+ NOTE:A password is required for user `opensips' :-( The following rules are used: db: '127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y' host : 'Not processed: host-field is not empty in db-table.' user : '%','opensips','*4DDB979A2666D0CF0A83FCCED820A64E8EBB6AFD','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0' On Tue, Jun 30, 2009 at 11:01 AM, Uwe Kastens ki...@kiste.org mailto:ki...@kiste.org wrote: Hmm, And you did change the host in the opensips url from localhost to 127.0.0.1? For a test change the 127.0.0.1 in the mysql to % and try again. And mabye change the password with update user set password=password(yourpassword) where user=opensips; Your password entry looks like from a elder version. BR Uwe Brett Nemeroff schrieb: Here is what I got: password too long? really? it's not that long.. shrug mysqlaccess Version 2.06, 20 Dec 2000 By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be) Changes by Steve Harvey (s...@vex.net mailto:s...@vex.net mailto:s...@vex.net mailto:s...@vex.net) This software comes with ABSOLUTELY NO WARRANTY. Password for MySQL superuser root: Access-rights for USER 'opensips', from HOST 'localhost', to DB 'opensips' +-+---+ +-+---+ | Select_priv | Y | | Lock_tables_priv | Y | | Insert_priv | Y | | Execute_priv| Y | | Update_priv | Y | | Repl_slave_priv | N | | Delete_priv | Y | | Repl_client_priv | N | | Create_priv | Y | | Create_view_priv | Y | | Drop_priv | Y | | Show_view_priv | Y | | Reload_priv | N | | Create_routine_priv | Y | | Shutdown_priv | N | | Alter_routine_priv | Y | | Process_priv| N | | Create_user_priv | N | | File_priv | N | | Ssl_type| ? | | Grant_priv | N | | Ssl_cipher | ? | | References_priv | Y | | X509_issuer | ? | | Index_priv | Y | | X509_subject| ? | | Alter_priv | Y | | Max_questions | 0 | | Show_db_priv| N | | Max_updates | 0 | | Super_priv | N | | Max_connections | 0 | | Create_tmp_table_priv | Y | | Max_user_connections | 0 | +-+---+ +-+---+ NOTE:A password is required for user `opensips' :-( The following rules are used: db: '127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y' host : 'Not processed: host-field is not empty in db-table.' user : '127.0.0.1','opensips','641b9f69397f5d64','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0' On Tue, Jun 30, 2009 at 10:41 AM, Uwe Kastens ki...@kiste.org mailto:ki...@kiste.org mailto:ki...@kiste.org mailto:ki...@kiste.org wrote: Hi, Hmmm. - password to long Have you mysqlaccess
Re: [OpenSIPS-Users] mysql database connection error
Hi Brett, good to know BR Uwe Brett Nemeroff schrieb: Hey.. Ok I found the error and I feel like a complete idiot. I had modules loaded without a db_url specified for it (one module's db_url was missing). I think the whole idea of a default db_url is a mistake... that's my personal opinion. Thanks for helping me find this.. -Brett On Tue, Jun 30, 2009 at 12:48 PM, Uwe Kastens ki...@kiste.org mailto:ki...@kiste.org wrote: Hi Brett, Could be broken libs. I would start opensips with strace and look for errors. Which OS are you using? BR Uwe Brett Nemeroff schrieb: Still can't connect :( What could I be doing wrong? I wonder if it's the mysql client libs somehow?! Access-rights for USER 'opensips', from HOST 'localhost', to DB 'opensips' +-+---+ +-+---+ | Select_priv | Y | | Lock_tables_priv | Y | | Insert_priv | Y | | Execute_priv| Y | | Update_priv | Y | | Repl_slave_priv | N | | Delete_priv | Y | | Repl_client_priv | N | | Create_priv | Y | | Create_view_priv | Y | | Drop_priv | Y | | Show_view_priv | Y | | Reload_priv | N | | Create_routine_priv | Y | | Shutdown_priv | N | | Alter_routine_priv | Y | | Process_priv| N | | Create_user_priv | N | | File_priv | N | | Ssl_type| ? | | Grant_priv | N | | Ssl_cipher | ? | | References_priv | Y | | X509_issuer | ? | | Index_priv | Y | | X509_subject| ? | | Alter_priv | Y | | Max_questions | 0 | | Show_db_priv| N | | Max_updates | 0 | | Super_priv | N | | Max_connections | 0 | | Create_tmp_table_priv | Y | | Max_user_connections | 0 | +-+---+ +-+---+ NOTE:A password is required for user `opensips' :-( The following rules are used: db: '127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y' host : 'Not processed: host-field is not empty in db-table.' user : '%','opensips','*4DDB979A2666D0CF0A83FCCED820A64E8EBB6AFD','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0' On Tue, Jun 30, 2009 at 11:01 AM, Uwe Kastens ki...@kiste.org mailto:ki...@kiste.org mailto:ki...@kiste.org mailto:ki...@kiste.org wrote: Hmm, And you did change the host in the opensips url from localhost to 127.0.0.1? For a test change the 127.0.0.1 in the mysql to % and try again. And mabye change the password with update user set password=password(yourpassword) where user=opensips; Your password entry looks like from a elder version. BR Uwe Brett Nemeroff schrieb: Here is what I got: password too long? really? it's not that long.. shrug mysqlaccess Version 2.06, 20 Dec 2000 By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be mailto:yves.carl...@rug.ac.be) Changes by Steve Harvey (s...@vex.net mailto:s...@vex.net mailto:s...@vex.net mailto:s...@vex.net mailto:s...@vex.net mailto:s...@vex.net mailto:s...@vex.net mailto:s...@vex.net) This software comes with ABSOLUTELY NO WARRANTY. Password for MySQL superuser root: Access-rights for USER 'opensips', from HOST 'localhost', to DB 'opensips' +-+---+ +-+---+ | Select_priv | Y | | Lock_tables_priv | Y | | Insert_priv | Y | | Execute_priv| Y | | Update_priv | Y | | Repl_slave_priv | N | | Delete_priv | Y | | Repl_client_priv | N | | Create_priv | Y | | Create_view_priv | Y | | Drop_priv | Y | | Show_view_priv | Y | | Reload_priv | N | | Create_routine_priv | Y | | Shutdown_priv | N | | Alter_routine_priv | Y | | Process_priv| N | | Create_user_priv | N | | File_priv | N | | Ssl_type| ? | | Grant_priv | N | | Ssl_cipher | ? | | References_priv | Y
Re: [OpenSIPS-Users] Help needed to change database RW user/password
Hi, what kind of db are you using? Do you know how to change the password on your database backend? If so, you should use the same password for the user in your database and in the opensips config. BR Uwe srikanth R schrieb: Hi, I have got my SIP proxy up and running, but I am able to work only with the username 'opensips' and the password 'opensipsrw'. When I change the username and password in opensipsctlrc file I get an error message saying access denied. I am not asure about which variables to comment out and which variables to leave as such. Any help in this regard would be appreciated. Thanks, Srikanth Yahoo! recommends that you upgrade to the new and safer Internet Explorer 8 http://in.rd.yahoo.com/tagline_ie8_1/*http://downloads.yahoo.com/in/internetexplorer/. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No RADIUS traffic
Hi, This is rather strange. Are there any knows bugs for the libradiusclient package for the platform your are using? I would try to recompile that package. I remember there was a broken package in debian a couple of month ago, maybe on other platforms too. If not I have no further ideas anymore. Good luck BR Uwe Leon Li schrieb: Uwe, Strace output, nothing comes when I tries to register an endpoint. [...@cbr-a-sysdev1 lli]$ sudo /usr/local/bin/strace -f -e open /usr/local/sbin/opensips open(/etc/ld.so.preload, O_RDONLY)= -1 ENOENT (No such file or directory) open(/etc/ld.so.cache, O_RDONLY) = 3 open(/lib/libdl.so.2, O_RDONLY) = 3 open(/lib/libresolv.so.2, O_RDONLY) = 3 open(/lib/tls/libc.so.6, O_RDONLY)= 3 open(/usr/local/etc/opensips/opensips.cfg, O_RDONLY) = 3 open(/dev/urandom, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/signaling.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/sl.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/tm.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/rr.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/maxfwd.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/usrloc.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/registrar.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/textops.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/mi_fifo.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/uri_db.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/uri.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/xlog.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/acc.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/auth.so, O_RDONLY) = 4 open(/usr/local/lib/opensips/modules/auth_radius.so, O_RDONLY) = 4 open(/etc/ld.so.cache, O_RDONLY) = 4 open(/usr/local/lib/libradiusclient-ng.so.2, O_RDONLY) = 4 open(/lib/libcrypt.so.1, O_RDONLY)= 4 open(/lib/libnsl.so.1, O_RDONLY) = 4 open(/etc/resolv.conf, O_RDONLY) = 4 open(/etc/nsswitch.conf, O_RDONLY)= 4 open(/etc/ld.so.cache, O_RDONLY) = 4 open(/lib/libnss_files.so.2, O_RDONLY) = 4 open(/etc/host.conf, O_RDONLY)= 4 open(/etc/hosts, O_RDONLY)= 4 open(/etc/hosts, O_RDONLY)= 4 Listening on udp: 202.158.197.134 [202.158.197.134]:5060 tcp: 202.158.197.134 [202.158.197.134]:5060 Aliases: tcp: cbr-a-sysdev1:5060 tcp: cbr-a-sysdev1.aarnet.net.au:5060 udp: cbr-a-sysdev1:5060 udp: cbr-a-sysdev1.aarnet.net.au:5060 open(/etc/localtime, O_RDONLY)= 4 open(/dev/zero, O_RDWR) = 5 open(/usr/local/etc/radiusclient-ng/radiusclient.conf, O_RDONLY) = 6 open(/usr/local/etc/radiusclient-ng/dictionary, O_RDONLY) = 6 Process 25956 attached Process 25957 attached Process 25958 attached [pid 25958] open(/tmp/opensips_fifo, O_RDONLY|O_NONBLOCK) = 9 [pid 25958] open(/tmp/opensips_fifo, O_WRONLY|O_NONBLOCK) = 11 Ldd auth_radiu.so: [...@cbr-a-sysdev1 lli]$ /usr/bin/ldd /usr/local/lib/opensips/modules/auth_radius.so libradiusclient-ng.so.2 = /usr/local/lib/libradiusclient-ng.so.2 (0x00ee5000) libc.so.6 = /lib/tls/libc.so.6 (0x00111000) libcrypt.so.1 = /lib/libcrypt.so.1 (0x00eb) libnsl.so.1 = /lib/libnsl.so.1 (0x0064c000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x0067b000) Thanks, Leon -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: Friday, 26 June 2009 5:26 PM To: Leon Li Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] No RADIUS traffic Leon, Could you post the output of the strace call? And could you please post the output of ldd auth_radius.so ? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Update = dlg_handlers vs. dlg_db_handler DIALOG not deleted on BYE
Hi, If I use db_mode=1 the dialog is deleted from the database (as expected) but not from the memory. I will test with a fresh installation and maybe open a bug report. BR Uwe Uwe Kastens schrieb: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. Unfort I was not able to find out what the states and the events means. So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DIALOG not deleted on BYE
Hi Bogdan, Again, thanks a lot for your help. The loose_route() seems to cause the problem, but somehow its needed to pass byes correctly to the UA. So I need to work a little on my skript. I will try the 1.6 ASAP and let you know the result. BR Uwe Bogdan-Andrei Iancu schrieb: If you could test, a fix is available on 1.6 (trunk) version - if ok, I will do the backport. Regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Thanks for the traces. Looking at the opensips logs, I say you do loose_route() twice for the ACK which looks twice for the dialog and increase the ref twice for the dialogthis is why the ref never gets back to 0 to allow the dialog to be destroyed.. Could you confirm this for me ? even if it's a script error , the dialog module should cope with it..I will look for a fix. Thanks and regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Uwe Kastens wrote: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. yes, it sounds like. Unfort I was not able to find out what the states and the events means. You can find the meaning of each state in: modules/dialog/dlg_hash.h So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? I can try : ) could you (privately if needed) please send me the the full logs for the entire call (debug=6) - for the non working part. Thanks and regards, Bogdan BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPS boot Camp
Hello Bogdan, say 2 months and you can study and run the seminars by yourself. Included, you will have the possibility to fire questions to the teachers if you have something to clarify or if you got stuck with the labs What do you think of such approach ? I would love that, since one would be able to learn on weekend or after work. For some of us its not that easy to take a week off. No question that a personal training is better. I would take 2 Accounts if you will offer it. BR Uwe BTW: What about a opensips user meeting? I would prefer North of europe:) -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] 1.6 and mysql = crash Re: DIALOG not deleted on BYE
Bogdan, Sorry for bothering again. I tried the latest trunk from svn and opensips is dying after accessing the mysql db. I will attach the trace. BR Uwe Bogdan-Andrei Iancu schrieb: OK - with the fix from SVN you should be able to call loose_route() as many times you want without any risk - just let me know if it works as expected. Regards, Bogdan Uwe Kastens wrote: Hi Bogdan, Again, thanks a lot for your help. The loose_route() seems to cause the problem, but somehow its needed to pass byes correctly to the UA. So I need to work a little on my skript. I will try the 1.6 ASAP and let you know the result. BR Uwe Bogdan-Andrei Iancu schrieb: If you could test, a fix is available on 1.6 (trunk) version - if ok, I will do the backport. Regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Thanks for the traces. Looking at the opensips logs, I say you do loose_route() twice for the ACK which looks twice for the dialog and increase the ref twice for the dialogthis is why the ref never gets back to 0 to allow the dialog to be destroyed.. Could you confirm this for me ? even if it's a script error , the dialog module should cope with it..I will look for a fix. Thanks and regards, Bogdan Bogdan-Andrei Iancu wrote: Hi Uwe, Uwe Kastens wrote: Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. yes, it sounds like. Unfort I was not able to find out what the states and the events means. You can find the meaning of each state in: modules/dialog/dlg_hash.h So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? I can try : ) could you (privately if needed) please send me the the full logs for the entire call (debug=6) - for the non working part. Thanks and regards, Bogdan BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org
Re: [OpenSIPS-Users] No RADIUS traffic
Leon, Could you post the output of the strace call? And could you please post the output of ldd auth_radius.so ? BR Uwe Leon Li schrieb: Uwe, I tried the strace tool but no line is trying to use radius.seq. I manually created radius.seq like -rw-rw-rw-1 root root 0 Jun 25 00:45 radius.seq because it is not created for some reason. Will this be a problem? Regards, Leon -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: Tuesday, 23 June 2009 5:31 PM To: Leon Li Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] No RADIUS traffic Li, I was wondering about the answer from radius: WARNING: Ignoring Status-Server request due to security configuration If I try the same I will get an answer like: Received response ID 196, code 2, length = 20 Could you please check your shared secret. Also, I cannot find file /var/run/radius.seq. Is it created automatically? I should be there if radius will work - but remember your permissions. You can try one thing: set fork=no in opensips.cfg, install strace and start opensips with strace -f -e open opensips. Now start one attempt to register etc.pp. and watch the line with the seq. [pid 20680] open(/var/run/opensips/radius.seq, O_RDWR|O_CREAT|O_APPEND, 0666) = 13 BR Uwe Leon Li schrieb: Uwe, I got the following from RADIUS when issue the command you gave. rad_recv: Status-Server packet from host 127.0.0.1:39297, id=17, length=38 WARNING: Ignoring Status-Server request due to security configuration --- Walking the entire request list --- Nothing to do. Sleeping until we see a request. rad_recv: Status-Server packet from host 127.0.0.1:39297, id=17, length=38 WARNING: Ignoring Status-Server request due to security configuration --- Walking the entire request list --- So I assume that the radius server is working? Also, I cannot find file /var/run/radius.seq. Is it created automatically? Regards, Leon -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: Wednesday, 17 June 2009 6:01 PM To: Leon Li Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] No RADIUS traffic Leon, mysql.so in opensips is not needed for the radius authentication. Shared secrets for radius are correct? Anyway you should see some traffic on the radius server. Could you please test echo Message-Authenticator = 0x00 | radclient 127.0.0.1:1812 status shared secret You should see then traffic on radiusd -X If yes I would start checking permissions again BR uwe Leon Li schrieb: Hi Ashwini, I have added param for aut_radius, but no luck. L Why do I need mysql.so if the radius server will host all users credential? Regards, Leon *From:* ASHWINI NAIDU [mailto:ashwini.na...@gmail.com] *Sent:* Monday, 15 June 2009 2:52 PM *To:* Leon Li *Cc:* Uwe Kastens; users@lists.opensips.org *Subject:* Re: [OpenSIPS-Users] No RADIUS traffic On Mon, Jun 15, 2009 at 10:19 AM, ASHWINI NAIDU ashwini.na...@gmail.com mailto:ashwini.na...@gmail.com wrote: hi leon, But i do not see your openser communicating with radiusclient. modparam(auth_radius, radius_config, /etc/radiusclient-ng/radiusclient.conf) mention the path of radiusclient.conf properly. Your mysql support is also commented. *loadmodule mysql.so* On Mon, Jun 15, 2009 at 5:13 AM, Leon Li leon...@aarnet.edu.au mailto:leon...@aarnet.edu.au wrote: Here it is. ### Global Parameters # debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ debug=6 fork=no log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = /usr/local/etc/openser/tls/user/user-cert.pem #tls_private_key = /usr/local/etc/openser/tls/user/user-privkey.pem #tls_ca_list = /usr/local/etc/openser/tls/user/user-calist.pem listen=202.158.197.134 port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp
[OpenSIPS-Users] DIALOG not deleted on BYE
Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DIALOG not deleted on BYE
Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 dialog.gz Description: GNU Zip compressed data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DIALOG not deleted on BYE
Hi again, So I think it might be a bug. One direction (UA to PSTN) works everytime perfectly. It doesn't matter on which side the BYE is sent. If I try the other direction, the dialog will not be removed. Again it won't matter on which side the BYE is sent - the dialog will stay active. Unfort I was not able to find out what the states and the events means. So its not easy to debug further. Working direction: DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 6 DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to state 4, due event 1 Not Working DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 2, due event 2 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to state 3, due event 3 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to state 5, due event 7 DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to state 5, due event 1 Anyone could help please? BR Uwe Uwe Kastens schrieb: Hello again, I think the dialog is destroyed, if no reference is left. And so I asume the dialog is missing the ACK for the BYE. Or do I need to unref it manually via reply_route? I will attach the log. dialog:: hash=440:1838775488 state:: 5 user_flags:: 0 timestart:: 1246005835 timeout:: 0 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105 from_uri:: sip:9904...@10.20.138.105:5100 from_tag:: as619609ab caller_contact:: sip:9904...@10.20.138.105:5100 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:10.20.138.125:5100 to_uri:: sip:4315302...@asn2.domain.de:5100 to_tag:: ZdwulVArZJyQZ6lMpIk9pvPlzPV73upl callee_contact:: sip:4315302...@10.20.139.62:5060 callee_cseq:: 102 callee_route_set:: sip:10.20.138.145;lr;ftag=as619609ab;did=8b1.8ddb7a7 callee_bind_addr:: udp:10.20.138.125:5100 BR Uwe Uwe Kastens schrieb: Hello list, I am using DIALOG for the Concurrent calls limitation following the tutorial. Its working pretty well - in one direction :-) DIALOGs from UA to PSTN are deleted after processing the BYE. In the other direction I see that the BYE is processed correctly, but DIALOGs are staying in state 5. Where can I find the documentation for the states? Which will delete a DIALOG. The BYE or the ack for the BYE? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Some results = Re: register performance with sipp
Hi, In my testing setup I am not able to reach more than 500 cps without massive retransmits on db_mode=3. On db_mode=2 i reach with the same setup 900cps. I tried with one sipp client and with two - it makes no difference. All hosts are on xen except the db-server. My setup: - 1 or 2 sipp instances (it makes no difference) - 1 opensips 1.5.1 1024MB RAM - freeradius with 8192 max_connections, and 25 sql socks against mysql - opensips will register via rad...@mysql - usrloc db_mode=3 (only db) or db_mode=2 I generate user and password via pwgen and but them in radcheck db. I use a REGISTER scenario with sipp and start it like: sipp -sf register.xml -inf user.csv xxx.xxx.de -m 5000 -r 500 -nd Messages Retrans Timeout Unexpected-Msg REGISTER -- 5000 0 0 401 -- 5000 0 0 REGISTER -- 5000 360 200 -- 5000 0 7 -- Test Terminated - Statistics Screen --- [1-9]: Change Screen -- Start Time | 2009-06-22 09:21:17 Last Reset Time| 2009-06-22 09:21:27 Current Time | 2009-06-22 09:21:27 -+---+-- Counter Name | Periodic value| Cumulative value -+---+-- Elapsed Time | 00:00:00:202 | 00:00:10:218 Call Rate |0.000 cps | 489.333 cps -+---+-- Incoming call created |0 |0 OutGoing call created |0 | 5000 Total Call created | | 5000 Current Call |0 | -+---+-- Successful call|4 | 5000 Failed call|0 |0 -+---+-- Call Length| 00:00:00:137 | 00:00:00:015 -- Test Terminated 2009-06-22 09:21:27: Discarding message which can't be mapped to a known SIPp call: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.20.139.62:5060;branch=z9hG4bK-18810-4611-2 From: sip:aizeejenupigeeghe...@xxx.xxx.de;tag=4611 To: sip:aizeejenupigeeghe...@xxx.xxx.de;tag=a0ed7041c178b85c9908fb3ff756ca71.2356 Call-ID: 4611-18...@10.20.139.62 CSeq: 2 REGISTER WWW-Authenticate: Digest realm Uwe Kastens schrieb: Hi, Thanks for the pointer. I should mabye learn to search on the mailing list :) Do you remember what your hardware setup looked like? Esp I would be very interested in the hardware you used for the sipp requests. Did you see any errors like this on your tests Discarding message which can't be mapped to a known SIPp call? I was wondering if this is caused by the opensips server or by some pkg sipp is loosing. I think I would like to document some of the tests in the opensips wiki. BR Uwe Iñaki Baz Castillo schrieb: El Domingo, 21 de Junio de 2009, Uwe Kastens escribió: Hello, Anybody experience with measuring REGISTER performance with sipp? I made some tests and I was wondering how many requests should be possible with opensips/sipps (radius against mysql). It looks like, that I can handle easily 500 REGISTER requests per sec on a XEN Domain (one for sipp and one for opensips), database is on dedicated quad-core server. I found out that my freeradius config caused some trouble (max_requests). What could I expect with that setup? Hi, I did some tests about it (with no radius however): http://lists.opensips.org/pipermail/users/2008-December/002074.html -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Basic call failed
Hi, looks like your opensips will handle @40.0.0.164 as not local, so it will not ask your location db. Maybe you should entry the domain 40.0.0.164 in your domain table in mysql BR Uwe XIN Xiuhe schrieb: Hi, Please find attached logfile. thanks -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: 2009年6月19日 14:33 To: XIN Xiuhe Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Basic call failed Hi, that looks ok. Could you please make a call with opensips at debuglevel 9 at post the logfile? BR uwe XIN Xiuhe schrieb: mysql select * from location; ++--++--+--+--+-+---+-+--+-+---++-+-+-+ | id | username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | cflags | user_agent | socket | methods | ++--++--+--+--+-+---+-+--+-+---++-+-+-+ | 11 | 0004 | NULL | sip:0...@40.0.0.165:5060 | NULL | NULL | 2009-06-19 13:41:26 | -1.00 | 1569339300-27981165 |2 | 2009-06-19 12:41:26 | 0 | 0 | Alcatel-Lucent ISAM | udp:40.0.0.164:7060 | 3199 | | 12 | 0003 | NULL | sip:0...@40.0.0.165:5060 | NULL | NULL | 2009-06-19 13:41:28 | -1.00 | 1570548700-27981169 |2 | 2009-06-19 12:41:28 | 0 | 0 | Alcatel-Lucent ISAM | udp:40.0.0.164:7060 | 3199 | ++--++--+--+--+-+---+-+--+-+---++-+-+-+ 2 rows in set (0.00 sec) mysql -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: 2009年6月19日 12:47 To: XIN Xiuhe Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Basic call failed Hi, Are you working with a database? So what will the location table entries look like? eq select * from location where user=0004 BR Uwe XIN Xiuhe schrieb: Hi, uwe Please find attached opensips.cfg. For this question: What will the contact from the location service show you for Users? I don't know what you mean, do you mean user A and B's contact info? user A: contact info is 0...@40.0.0.165, uri is 0...@40.0.0.164 user B: contact info is 0...@40.0.0.165 , uri is 0...@40.0.0.164 Thanks for your help! xxh -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: 2009年6月19日 12:15 To: XIN Xiuhe Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Basic call failed Hi, What will the contact from the location service show you for Users? Could you post your opensips.cfg? BR Uwe XIN Xiuhe schrieb: Hi, I tried to use opensips to make a basic call, but failed. user A: 0...@40.0.0.165 user B: 0...@40.0.0.165 Both of them registered with opensips(ip address: 40.0.0.164) successfully. User A off hook and call user B, after opensips received the invite message, it should send it to 40.0.0.165, but from the trace (attached basiccall.pcap), it send it to 40.0.0.164. What is the root cause, can somebody give some ideas? Thanks ! - - -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] extract expires header value
Hi, This could match: http://www.opensips.org/Resources/DocsCoreVar#toc17 3.14 Contact instance $ct - reference to contact instance/body from the contact header. A contact instance is display_name + URI + contact_params. As a Contact header may contain multiple Contact instances and a message may contain multiple Contact headers, an index was added to the $ct variable: * $ct -first contact instance from message * $(ct[n]) - the n-th contact instance form the beginning of message, starting with index 0 * $(ct[-n]) - the n-th contact instance form the end of the message, starting with index -1 (the last contact instance) 3.15 Fields of a contact instance $ct,fields() - reference to the fields of a contact instance/body (see above). Supported fields are: * name - display name * uri - contact uri * q - q param (value only) * expires - expires param (value only) * methods - methods param (value only) * received - received param (value only) * params - all params (including names) Examples: * $ct.fields(uri) - the URI of the first contact instance * $(ct.fields(name)[1]) - the display name of the second contact instance BR uwe Jayesh Nambiar schrieb: Hello, Thanks Uwe for the pointers. Is there a way to extract the expires value from the CONTACT header that comes in the REGISTER message. Thanks, --- Jay -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Basic call failed
Hi, What will the contact from the location service show you for Users? Could you post your opensips.cfg? BR Uwe XIN Xiuhe schrieb: Hi, I tried to use opensips to make a basic call, but failed. user A: 0...@40.0.0.165 user B: 0...@40.0.0.165 Both of them registered with opensips(ip address: 40.0.0.164) successfully. User A off hook and call user B, after opensips received the invite message, it should send it to 40.0.0.165, but from the trace (attached basiccall.pcap), it send it to 40.0.0.164. What is the root cause, can somebody give some ideas? Thanks ! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Basic call failed
Hi, Are you working with a database? So what will the location table entries look like? eq select * from location where user=0004 BR Uwe XIN Xiuhe schrieb: Hi, uwe Please find attached opensips.cfg. For this question: What will the contact from the location service show you for Users? I don't know what you mean, do you mean user A and B's contact info? user A: contact info is 0...@40.0.0.165, uri is 0...@40.0.0.164 user B: contact info is 0...@40.0.0.165 , uri is 0...@40.0.0.164 Thanks for your help! xxh -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: 2009年6月19日 12:15 To: XIN Xiuhe Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Basic call failed Hi, What will the contact from the location service show you for Users? Could you post your opensips.cfg? BR Uwe XIN Xiuhe schrieb: Hi, I tried to use opensips to make a basic call, but failed. user A: 0...@40.0.0.165 user B: 0...@40.0.0.165 Both of them registered with opensips(ip address: 40.0.0.164) successfully. User A off hook and call user B, after opensips received the invite message, it should send it to 40.0.0.165, but from the trace (attached basiccall.pcap), it send it to 40.0.0.164. What is the root cause, can somebody give some ideas? Thanks ! -- -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No RADIUS traffic
Leon, mysql.so in opensips is not needed for the radius authentication. Shared secrets for radius are correct? Anyway you should see some traffic on the radius server. Could you please test echo Message-Authenticator = 0x00 | radclient 127.0.0.1:1812 status shared secret You should see then traffic on radiusd -X If yes I would start checking permissions again BR uwe Leon Li schrieb: Hi Ashwini, I have added param for aut_radius, but no luck. L Why do I need mysql.so if the radius server will host all users credential? Regards, Leon *From:* ASHWINI NAIDU [mailto:ashwini.na...@gmail.com] *Sent:* Monday, 15 June 2009 2:52 PM *To:* Leon Li *Cc:* Uwe Kastens; users@lists.opensips.org *Subject:* Re: [OpenSIPS-Users] No RADIUS traffic On Mon, Jun 15, 2009 at 10:19 AM, ASHWINI NAIDU ashwini.na...@gmail.com mailto:ashwini.na...@gmail.com wrote: hi leon, But i do not see your openser communicating with radiusclient. modparam(auth_radius, radius_config, /etc/radiusclient-ng/radiusclient.conf) mention the path of radiusclient.conf properly. Your mysql support is also commented. *loadmodule mysql.so* On Mon, Jun 15, 2009 at 5:13 AM, Leon Li leon...@aarnet.edu.au mailto:leon...@aarnet.edu.au wrote: Here it is. ### Global Parameters # debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ debug=6 fork=no log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = /usr/local/etc/openser/tls/user/user-cert.pem #tls_private_key = /usr/local/etc/openser/tls/user/user-privkey.pem #tls_ca_list = /usr/local/etc/openser/tls/user/user-calist.pem listen=202.158.197.134 port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 http://192.168.1.2:5060 ### Modules Section #set module path mpath=/usr/local/lib/openser/modules/ /* uncomment next line for MySQL DB support */ #loadmodule mysql.so loadmodule sl.so loadmodule tm.so loadmodule rr.so loadmodule maxfwd.so loadmodule usrloc.so loadmodule registrar.so loadmodule textops.so loadmodule mi_fifo.so loadmodule uri_db.so loadmodule uri.so loadmodule xlog.so loadmodule acc.so /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule auth.so loadmodule auth_radius.so #loadmodule auth_db.so /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule alias_db.so /* uncomment next line for multi-domain support NOTE: a DB (like mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see multi-module params section ) */ #loadmodule domain.so /* uncomment the next two lines for presence server support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule presence.so #loadmodule presence_xml.so # - setting module-specific parameters --- # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/openser_fifo) # - rr params - # add value to ;lr param to cope with most of the UAs modparam(rr, enable_full_lr, 1) # do not append from tag to the RR (no need for this script) modparam(rr, append_fromtag, 0) # - rr params - modparam(registrar, method_filtering, 1) /* uncomment the next line to disable parallel forking via location */ # modparam(registrar, append_branches, 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam(registrar, max_contacts, 10) # - uri_db params - /* by default we disable the DB support in the module as we do not need
Re: [OpenSIPS-Users] update location table on REGISTER request
Hi,
You could solve this by working with AVP and sql. The idea could be:
if (method==REGISTER){
Authentication goes here
$avp(s:user)=$aU;
avp_db_query(select count() from location where
username='$avp(s:user)',var(x));
if var(x) 0 {
avp_db_query(delete from location where username='$avp(s:user)');
}
save(location);
Untested!
BR
Uwe
Jayesh Nambiar schrieb:
Hi All,
I had a requirement of allowing only one registration per user in a
particular scenario. I did not want to use the max_contacts parameter of
registrar module as it wont work right !!! The drawback was:
If device A had registered successfully and for some reason got
disconnected from the internet, the device won't unregister itself.
Opensips still has the record in the location table for that device, now
if the internet comes back and when the device tries to register again,
opensips will not allow since it already has the record in the location.
The device will have to wait until the earlier registration expires in
the opensips.
The idea was to have a way of updating the location table if same user
is trying to REGISTER from same location or different location. Meaning
if user A is registered from location A and someone else using same
credentials of user A tries to register from location B, the location
table should only update the earlier record to location B and not keep
location A and location B both for user A.
Is there a way to do this. Any help will be highly appreciiated.
Thanks in advance.
--- Jay
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Re: [OpenSIPS-Users] update location table on REGISTER request
Hi,
ok. good point in a high traffic env.
The same thing may work if you build a binary which works with the
ul_delete_urecord(domain, aor)
(http://www.opensips.org/html/docs/modules/1.5.x/usrloc.html) and call
it each time a user registers.
BR
uwe
Jayesh Nambiar schrieb:
Hi Uwe,
This will not work well if i use db_mode as 2 in the usrloc module.
db_mode 1 does lot of DB queries !!!
--- Jay
On Wed, Jun 17, 2009 at 4:33 PM, Uwe Kastens ki...@kiste.org
mailto:ki...@kiste.org wrote:
Hi,
You could solve this by working with AVP and sql. The idea could be:
if (method==REGISTER){
Authentication goes here
$avp(s:user)=$aU;
avp_db_query(select count() from location where
username='$avp(s:user)',var(x));
if var(x) 0 {
avp_db_query(delete from location where username='$avp(s:user)');
}
save(location);
Untested!
BR
Uwe
Jayesh Nambiar schrieb:
Hi All,
I had a requirement of allowing only one registration per user in a
particular scenario. I did not want to use the max_contacts
parameter of
registrar module as it wont work right !!! The drawback was:
If device A had registered successfully and for some reason got
disconnected from the internet, the device won't unregister itself.
Opensips still has the record in the location table for that
device, now
if the internet comes back and when the device tries to register
again,
opensips will not allow since it already has the record in the
location.
The device will have to wait until the earlier registration expires in
the opensips.
The idea was to have a way of updating the location table if same user
is trying to REGISTER from same location or different location.
Meaning
if user A is registered from location A and someone else using same
credentials of user A tries to register from location B, the location
table should only update the earlier record to location B and not keep
location A and location B both for user A.
Is there a way to do this. Any help will be highly appreciiated.
Thanks in advance.
--- Jay
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[OpenSIPS-Users] better solution possible? = lookup and routing with 2 location databases
Hi,
Perhaps somebody has a better idea to solve the following issue. I have
two opensips servers. UA can register at both opensips with different
domains and credentials.
opensips1 will get the calls from pstn and should route them to
opensips2 , if the UA is registered as well with opensips2.
I would like to have parallel invites to all registered UAs. But the
invites should come from that opensips, with which the UA is registered.
1st I was wondering, why a call is forked to opensips2 as well, even if
no UA is registered there. I think it might be caused by the
rewritehostport(domain2.de) which is working on the current URI?
route[1] {
if (!loose_route()){
rewritehostport(domain1.de);
if lookup(locinternal) {
append_branch();
}
rewritehostport(domain2.de);
if (registered(location)) {
append_branch($...@domain2.de);
}
}
if (!t_relay()) {
sl_reply_error();
};
t_on_reply(1);
}
BR
Uwe
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Re: [OpenSIPS-Users] better solution possible? = lookup and routing with 2 location databases
Hi Bogdan,
Thanks for your answer.
If I understood you scripting correctly, the lookup(locinternal); will
have the value of the 1st lookup, if there is an entry in locinternal.
If the 2nd if (registered(location ...) will match, than I will have
the entry from location only, correct?
BR
Uwe
Bogdan-Andrei Iancu schrieb:
Hi Uwe,
I think the problem is in the branch management. If I get is right, you
get the records from locinternal table and if records are present in
location, you fork a branch to the other server, right ?
If so, you do something like :
route[1] {
if (!loose_route()){
rewritehostport(domain1.de);
$var(x) = $rU+@domain2.de;
lookup(locinternal);
if (registered(location,$var(x))) {
$rb = $var(x);
}
}
See:
http://www.opensips.org/html/docs/modules/1.5.x/registrar.html#id271315
Regards,
Bogdan
Uwe Kastens wrote:
Hi,
Perhaps somebody has a better idea to solve the following issue. I have
two opensips servers. UA can register at both opensips with different
domains and credentials.
opensips1 will get the calls from pstn and should route them to
opensips2 , if the UA is registered as well with opensips2.
I would like to have parallel invites to all registered UAs. But the
invites should come from that opensips, with which the UA is registered.
1st I was wondering, why a call is forked to opensips2 as well, even if
no UA is registered there. I think it might be caused by the
rewritehostport(domain2.de) which is working on the current URI?
route[1] {
if (!loose_route()){
rewritehostport(domain1.de);
if lookup(locinternal) {
append_branch();
}
rewritehostport(domain2.de);
if (registered(location)) {
append_branch($...@domain2.de);
}
}
if (!t_relay()) {
sl_reply_error();
};
t_on_reply(1);
}
BR
Uwe
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Re: [OpenSIPS-Users] No RADIUS traffic
Hi,
This is strange. Could you post your opensips.cfg or send it to me directly?
BR
Uwe
Leon Li schrieb:
The port is 1812, and specify them in radiusclient.conf with
Authserver127.0.0.1:1812
Run radius -X returns nothing when SIP client trying to register. :(
Regards,
Leon
-Original Message-
From: Uwe Kastens [mailto:ki...@kiste.org]
Sent: Thursday, 11 June 2009 5:02 PM
To: Leon Li
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] No RADIUS traffic
That is strange. Could you make sure that your radiusports are correct?
And could you start your radius-server with -X to have debug output on
stdout?
BR
Uwe
Leon Li schrieb:
I started the testing with RADIUS server on the same box as OpenSIPs.
There is nothing hit RADIUS logs.
Regards,
Leon
-Original Message-
From: Uwe Kastens [mailto:ki...@kiste.org]
Sent: Wednesday, 10 June 2009 4:29 PM
To: Leon Li
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] No RADIUS traffic
Hi,
Looks like that.
Jun 10 00:47:29 [24576] ERROR:auth_radius:radius_authorize_sterman:
rc_auth failed
Your radius server is a remote server? Could you try to start sniffing
at the opensips server?
BR
Uwe
Jun 10 00:47:29 [24576] ERROR:auth_radius:radius_authorize_sterman:
rc_auth failed
Leon Li schrieb:
My radiusclient.conf is almost the same as this, except the different
directory. I turned on debug as below and run OpenSIPs as root (-u
root)
but still nothing shows it try to send authentication to radius?
Jun 10 00:47:29 [24576] DBG:core:parse_msg: SIP Request:
Jun 10 00:47:29 [24576] DBG:core:parse_msg: method: REGISTER
Jun 10 00:47:29 [24576] DBG:core:parse_msg: uri:
sip:202.158.197.134
Jun 10 00:47:29 [24576] DBG:core:parse_msg: version: SIP/2.0
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=2
Jun 10 00:47:29 [24576] DBG:core:parse_to: end of header reached,
state=10
Jun 10 00:47:29 [24576] DBG:core:parse_to: display={test},
ruri={sip:t...@202.158.197.134}
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: To [35];
uri=[sip:t...@202.158.197.134]
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: to body [test
sip:t...@202.158.197.134
]
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: cseq CSeq: 1
REGISTER
Jun 10 00:47:29 [24576] DBG:core:parse_via_param: found param type
232,
branch = z9hG4bK-3ab88355ef-DL; state=16
Jun 10 00:47:29 [24576] DBG:core:parse_via: end of header reached,
state=5
Jun 10 00:47:29 [24576] DBG:core:parse_headers: via found, flags=2
Jun 10 00:47:29 [24576] DBG:core:parse_headers: this is the first via
Jun 10 00:47:29 [24576] DBG:core:receive_msg: After parse_msg...
Jun 10 00:47:29 [24576] DBG:core:receive_msg: preparing to run
routing
scripts...
Jun 10 00:47:29 [24576] DBG:maxfwd:is_maxfwd_present: value = 70
Jun 10 00:47:29 [24576] DBG:uri:has_totag: no totag
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=78
Jun 10 00:47:29 [24576] DBG:tm:t_lookup_request: start searching:
hash=44313, isACK=0
Jun 10 00:47:29 [24576] DBG:tm:matching_3261: RFC3261 transaction
matching failed
Jun 10 00:47:29 [24576] DBG:tm:t_lookup_request: no transaction found
Jun 10 00:47:29 [24576] DBG:core:grep_sock_info: checking if
host==us:
15==15 [202.158.197.134] == [202.158.197.134]
Jun 10 00:47:29 [24576] DBG:core:grep_sock_info: checking if port
5060
matches port 5060
Jun 10 00:47:29 [24576] DBG:core:parse_to_param: tag=DLdda82ca71a
Jun 10 00:47:29 [24576] DBG:core:parse_to_param: epid=09C9A6A0
Jun 10 00:47:29 [24576] DBG:core:parse_to: end of header reached,
state=29
Jun 10 00:47:29 [24576] DBG:core:parse_to: display={test},
ruri={sip:t...@202.158.197.134}
REGISTER for (test) sip:202.158.197.134
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=4000
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: content_length=0
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: found end of header
Jun 10 00:47:29 [24576] DBG:auth:pre_auth: credentials with given
realm
not found
Proxy Authentication Required (Digest)
Jun 10 00:47:29 [24576] DBG:auth:build_auth_hf: 'WWW-Authenticate:
Digest realm=202.158.197.134,
nonce=4a2f03cd2aecc81d4dd620f4ff564bed1d4d6963
'
Jun 10 00:47:29 [24576] DBG:core:parse_headers:
flags=
Jun 10 00:47:29 [24576] DBG:core:check_via_address: params
202.158.213.91, 202.158.213.91, 0
Jun 10 00:47:29 [24576] DBG:core:destroy_avp_list: destroying list
(nil)
Jun 10 00:47:29 [24576] DBG:core:receive_msg: cleaning up
Jun 10 00:47:29 [24576] DBG:core:parse_msg: SIP Request:
Jun 10 00:47:29 [24576] DBG:core:parse_msg: method: REGISTER
Jun 10 00:47:29 [24576] DBG:core:parse_msg: uri:
sip:202.158.197.134
Jun 10 00:47:29 [24576] DBG:core:parse_msg: version: SIP/2.0
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=2
Jun 10 00:47:29 [24576] DBG:core:parse_to: end of header reached,
state=10
Jun 10 00:47:29 [24576] DBG:core:parse_to: display={test},
ruri={sip:t
[OpenSIPS-Users] put value of header field into AVP
Hi, How can I put the value of a header field in one AVP? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] put value of header field into AVP
Hi,
Thanks, this is very cool.
Is there a way to split one avp without perl in a handfull of differnt
avps or is better to work with different header files?
Maybe some more background to that question. My pstn gw should send some
information for later usage in routing etc.pp. (main number and ddi
number, in which place should the number with ddi be sent to the UA).
So I was wondering if I use one header and split the variables on the
opensips or let the pstn gw insert more than one header field (if
needed). ATM I would prefer to work with different headers.
BR
Uwe
Jeff Pyle schrieb:
Generically, avp(s:oneavp) = $hdr(SIP-Header)
String translations also work here, such as:
$avp(s:oneavp) = $(hdr(P-Charge-Info){uri.user})
- Jeff
On 6/12/09 9:49 AM, Uwe Kastens ki...@kiste.org wrote:
Hi,
How can I put the value of a header field in one AVP?
BR
Uwe
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Re: [OpenSIPS-Users] put value of header field into AVP
Hi Brett,
Yeah you can do this. You should really review the wiki for such
questions, it's all there :)
Believe it or not, I searched a long time - but rather at the wrong place.
Try the select transformation:
http://www.opensips.org/Resources/DocsCoreTran15#toc6
$var(x) = 12,34,56;
$(var(x){s.select,1,,}) = 34 ;
Exactly what I wanted to have
Thanks a lot
BR
uwe
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Re: [OpenSIPS-Users] No RADIUS traffic
That is strange. Could you make sure that your radiusports are correct?
And could you start your radius-server with -X to have debug output on
stdout?
BR
Uwe
Leon Li schrieb:
I started the testing with RADIUS server on the same box as OpenSIPs.
There is nothing hit RADIUS logs.
Regards,
Leon
-Original Message-
From: Uwe Kastens [mailto:ki...@kiste.org]
Sent: Wednesday, 10 June 2009 4:29 PM
To: Leon Li
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] No RADIUS traffic
Hi,
Looks like that.
Jun 10 00:47:29 [24576] ERROR:auth_radius:radius_authorize_sterman:
rc_auth failed
Your radius server is a remote server? Could you try to start sniffing
at the opensips server?
BR
Uwe
Jun 10 00:47:29 [24576] ERROR:auth_radius:radius_authorize_sterman:
rc_auth failed
Leon Li schrieb:
My radiusclient.conf is almost the same as this, except the different
directory. I turned on debug as below and run OpenSIPs as root (-u
root)
but still nothing shows it try to send authentication to radius?
Jun 10 00:47:29 [24576] DBG:core:parse_msg: SIP Request:
Jun 10 00:47:29 [24576] DBG:core:parse_msg: method: REGISTER
Jun 10 00:47:29 [24576] DBG:core:parse_msg: uri:
sip:202.158.197.134
Jun 10 00:47:29 [24576] DBG:core:parse_msg: version: SIP/2.0
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=2
Jun 10 00:47:29 [24576] DBG:core:parse_to: end of header reached,
state=10
Jun 10 00:47:29 [24576] DBG:core:parse_to: display={test},
ruri={sip:t...@202.158.197.134}
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: To [35];
uri=[sip:t...@202.158.197.134]
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: to body [test
sip:t...@202.158.197.134
]
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: cseq CSeq: 1
REGISTER
Jun 10 00:47:29 [24576] DBG:core:parse_via_param: found param type
232,
branch = z9hG4bK-3ab88355ef-DL; state=16
Jun 10 00:47:29 [24576] DBG:core:parse_via: end of header reached,
state=5
Jun 10 00:47:29 [24576] DBG:core:parse_headers: via found, flags=2
Jun 10 00:47:29 [24576] DBG:core:parse_headers: this is the first via
Jun 10 00:47:29 [24576] DBG:core:receive_msg: After parse_msg...
Jun 10 00:47:29 [24576] DBG:core:receive_msg: preparing to run routing
scripts...
Jun 10 00:47:29 [24576] DBG:maxfwd:is_maxfwd_present: value = 70
Jun 10 00:47:29 [24576] DBG:uri:has_totag: no totag
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=78
Jun 10 00:47:29 [24576] DBG:tm:t_lookup_request: start searching:
hash=44313, isACK=0
Jun 10 00:47:29 [24576] DBG:tm:matching_3261: RFC3261 transaction
matching failed
Jun 10 00:47:29 [24576] DBG:tm:t_lookup_request: no transaction found
Jun 10 00:47:29 [24576] DBG:core:grep_sock_info: checking if host==us:
15==15 [202.158.197.134] == [202.158.197.134]
Jun 10 00:47:29 [24576] DBG:core:grep_sock_info: checking if port 5060
matches port 5060
Jun 10 00:47:29 [24576] DBG:core:parse_to_param: tag=DLdda82ca71a
Jun 10 00:47:29 [24576] DBG:core:parse_to_param: epid=09C9A6A0
Jun 10 00:47:29 [24576] DBG:core:parse_to: end of header reached,
state=29
Jun 10 00:47:29 [24576] DBG:core:parse_to: display={test},
ruri={sip:t...@202.158.197.134}
REGISTER for (test) sip:202.158.197.134
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=4000
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: content_length=0
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: found end of header
Jun 10 00:47:29 [24576] DBG:auth:pre_auth: credentials with given
realm
not found
Proxy Authentication Required (Digest)
Jun 10 00:47:29 [24576] DBG:auth:build_auth_hf: 'WWW-Authenticate:
Digest realm=202.158.197.134,
nonce=4a2f03cd2aecc81d4dd620f4ff564bed1d4d6963
'
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=
Jun 10 00:47:29 [24576] DBG:core:check_via_address: params
202.158.213.91, 202.158.213.91, 0
Jun 10 00:47:29 [24576] DBG:core:destroy_avp_list: destroying list
(nil)
Jun 10 00:47:29 [24576] DBG:core:receive_msg: cleaning up
Jun 10 00:47:29 [24576] DBG:core:parse_msg: SIP Request:
Jun 10 00:47:29 [24576] DBG:core:parse_msg: method: REGISTER
Jun 10 00:47:29 [24576] DBG:core:parse_msg: uri:
sip:202.158.197.134
Jun 10 00:47:29 [24576] DBG:core:parse_msg: version: SIP/2.0
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=2
Jun 10 00:47:29 [24576] DBG:core:parse_to: end of header reached,
state=10
Jun 10 00:47:29 [24576] DBG:core:parse_to: display={test},
ruri={sip:t...@202.158.197.134}
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: To [35];
uri=[sip:t...@202.158.197.134]
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: to body [test
sip:t...@202.158.197.134
]
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: cseq CSeq: 2
REGISTER
Jun 10 00:47:29 [24576] DBG:core:parse_via_param: found param type
232,
branch = z9hG4bK-4e1197fe1e-DL; state=16
Jun 10 00:47:29 [24576] DBG:core:parse_via: end of header reached,
state=5
Jun 10 00:47:29 [24576
Re: [OpenSIPS-Users] No RADIUS traffic
] DBG:tm:matching_3261: RFC3261 transaction
matching failed
Jun 10 00:47:29 [24576] DBG:tm:t_lookup_request: no transaction found
Jun 10 00:47:29 [24576] DBG:core:grep_sock_info: checking if host==us:
15==15 [202.158.197.134] == [202.158.197.134]
Jun 10 00:47:29 [24576] DBG:core:grep_sock_info: checking if port 5060
matches port 5060
Jun 10 00:47:29 [24576] DBG:core:parse_to_param: tag=DLdda82ca71a
Jun 10 00:47:29 [24576] DBG:core:parse_to_param: epid=09C9A6A0
Jun 10 00:47:29 [24576] DBG:core:parse_to: end of header reached,
state=29
Jun 10 00:47:29 [24576] DBG:core:parse_to: display={test},
ruri={sip:t...@202.158.197.134}
REGISTER for (test) sip:202.158.197.134
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=4000
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: content_length=0
Jun 10 00:47:29 [24576] DBG:auth:check_nonce: comparing
[4a2f03cd2aecc81d4dd620f4ff564bed1d4d6963] and
[4a2f03cd2aecc81d4dd620f4ff564bed1d4d6963]
Jun 10 00:47:29 [24576] ERROR:auth_radius:radius_authorize_sterman:
rc_auth failed
Proxy Authentication Required (Digest)
Jun 10 00:47:29 [24576] DBG:auth:build_auth_hf: 'WWW-Authenticate:
Digest realm=202.158.197.134,
nonce=4a2f03cd2aecc81d4dd620f4ff564bed1d4d6963
'
Jun 10 00:47:29 [24576] DBG:core:parse_headers: flags=
Jun 10 00:47:29 [24576] DBG:core:get_hdr_field: found end of header
Jun 10 00:47:29 [24576] DBG:core:check_via_address: params
202.158.213.91, 202.158.213.91, 0
Jun 10 00:47:29 [24576] DBG:core:destroy_avp_list: destroying list (nil)
Jun 10 00:47:29 [24576] DBG:core:receive_msg: cleaning up
Regards,
Leon
-Original Message-
From: Uwe Kastens [mailto:ki...@kiste.org]
Sent: Tuesday, 9 June 2009 5:27 PM
To: Leon Li
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] No RADIUS traffic
Hi,
cat radiusclient.conf |grep -v ^#|grep -v ^$
auth_orderradius,local
login_tries 4
login_timeout 60
nologin /etc/nologin
issue /etc/radiusclient-ng/issue
authserverlocalhost
acctserverlocalhost
servers /etc/radiusclient-ng/servers
dictionary/etc/radiusclient-ng/dictionary
login_radius /usr/sbin/login.radius
seqfile /var/run/opensips/radius.seq
mapfile /etc/radiusclient-ng/port-id-map
default_realm
radius_timeout10
radius_retries3
bindaddr localhost
login_local /bin/login
BR
Uwe
--
kiste lat: 54.322684, lon: 10.13586
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Re: [OpenSIPS-Users] No RADIUS traffic
Hi, cat radiusclient.conf |grep -v ^#|grep -v ^$ auth_order radius,local login_tries 4 login_timeout 60 nologin /etc/nologin issue /etc/radiusclient-ng/issue authserver localhost acctserver localhost servers /etc/radiusclient-ng/servers dictionary /etc/radiusclient-ng/dictionary login_radius/usr/sbin/login.radius seqfile /var/run/opensips/radius.seq mapfile /etc/radiusclient-ng/port-id-map default_realm radius_timeout 10 radius_retries 3 bindaddr localhost login_local /bin/login BR Uwe Leon Li schrieb: Hi, What is your radiusclient.conf like? Regards, Leon -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: Friday, 5 June 2009 7:28 PM To: Leon Li Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] No RADIUS traffic Hi, I do not have that login.radius on my system - I think its not used with opensips. I would say there might be an permissions issue. I can remember I had lots of trouble, cause I don't wanted to run everything as root:root. My setup looks like that seqfile /var/run/opensips/radius.seq with -rw-r--r-- 1 opensips opensips and drwxr-xr-x opensips opensips /etc/radiusclient-ng BR Uwe Leon Li schrieb: There is no such a file in the directory. Will it be generated by radiusclient-ng? Also, the radiusclient.conf shows: # program to call for a RADIUS authenticated login login_radius/usr/local/sbin/login.radius I checked /usr/local/sbin/login.radius, but it is only a dummy script. How it can be changed? Thanks, Leon -Original Message- From: Uwe Kastens [mailto:ki...@kiste.org] Sent: Thursday, 4 June 2009 5:12 PM To: Leon Li Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] No RADIUS traffic Hi, If I remember it correctly I had the same problem some day and it was caused by wrong permissions on /var/run/radius.seq. Just a guess BR Uwe Leon Li schrieb: Hi, I am try to use RADIUS server. However, after configuration, I found there is no RADIUS traffic at all. Log shows: Jun 4 06:45:59 /usr/local/sbin/openser[396]: rc_avpair_new: unknown attribute 5 Jun 4 06:45:59 /usr/local/sbin/openser[396]: ERROR:auth_radius:radius_authorize_sterman: rc_auth failed But nothing on RADIUS server end. OpenSIPs + radiusclient-ng on one box and RADIUS is on another. My radiusclient.conf is like: # General settings # specify which authentication comes first respectively which # authentication is used. possible values are: radius and local. # if you specify radius,local then the RADIUS server is asked # first then the local one. if only one keyword is specified only # this server is asked. auth_order radius,local # maximum login tries a user has login_tries 4 # timeout for all login tries # if this time is exceeded the user is kicked out login_timeout 60 # name of the nologin file which when it exists disables logins. # it may be extended by the ttyname which will result in # a terminal specific lock (e.g. /etc/nologin.ttyS2 will disable # logins on /dev/ttyS2) nologin /etc/nologin # name of the issue file. it's only display when no username is passed # on the radlogin command line issue /usr/local/etc/radiusclient-ng/issue # RADIUS settings # RADIUS server to use for authentication requests. this config # item can appear more then one time. if multiple servers are # defined they are tried in a round robin fashion if one # server is not answering. # optionally you can specify a the port number on which is remote # RADIUS listens separated by a colon from the hostname. if # no port is specified /etc/services is consulted of the radius # service. if this fails also a compiled in default is used. authserver 202.158.212.103:1812 # RADIUS server to use for accouting requests. All that I # said for authserver applies, too. # acctserver 202.158.212.103:1813 # file holding shared secrets used for the communication # between the RADIUS client and server servers /usr/local/etc/radiusclient-ng/servers # dictionary of allowed attributes and values # just like in the normal RADIUS distributions dictionary /usr/local/etc/radiusclient-ng/dictionary # program to call for a RADIUS authenticated login login_radius/usr/local/sbin/login.radius # file which holds sequence number for communication with the # RADIUS server seqfile /var/run/radius.seq # file which specifies mapping between ttyname and NAS-Port attribute mapfile /usr/local/etc/radiusclient-ng/port-id-map # default authentication realm to append to all usernames if no # realm was explicitly specified by the user # the radiusd directly form Livingston doesnt use any realms, so leave # it blank then default_realm
