[OpenSIPS-Users] fixed Re: trouble with mediaproxy and asterisk on callforwarding (not 302)

a > different router. > > > > Uwe Kastens wrote: >> Hi, >> >> I am using opensips 1.5.3 and the latest mediaproxy. I have a strange >> issue with asterisk servers. Maybe somebody might have a hint where to >> look. >> >> The problem is reproducable

[OpenSIPS-Users] trouble with mediaproxy and asterisk on callforwarding (not 302)

Hi, I am using opensips 1.5.3 and the latest mediaproxy. I have a strange issue with asterisk servers. Maybe somebody might have a hint where to look. The problem is reproducable with any asterisk server behind nat (nat=yes). Calls in and out are working without any problem, RTP is present, ringb

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 conf

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 authentica

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 yo

[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 _

[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 t

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.

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

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

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

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,

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 ve

[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 wit

Re: [OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE

> 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 > >> soluti

[OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE

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

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?

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.

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 branc

Re: [OpenSIPS-Users] opensips authentication

gt; 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 server

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:

Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE

we 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, >> >

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, 200

[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://

Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE

='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 th

Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE

anch 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

Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE

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

[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/mailm

[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

Re: [OpenSIPS-Users] howto => mediaproxy on lenny

at 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" wrote: > > Hello, > >

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 wo

[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 ___

[OpenSIPS-Users] solved => Re: explizit handling auf replyto

> 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, >&g

Re: [OpenSIPS-Users] opensips with asterisk => relay REGISTER

k_to()) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > > # auth done -> send it to registrar > consume_credentials(); > $du = "sip:ASTERISK_IP:ASTERISK_PORT"; > t_re

[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

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 zhangc

Re: [OpenSIPS-Users] Newbie To OpenSIPS

der 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:

Re: [OpenSIPS-Users] Newbie To OpenSIPS

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..

Re: [OpenSIPS-Users] Newbie To OpenSIPS

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 &

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

Re: [OpenSIPS-Users] explizit handling auf replyto

atefully 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, >

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/user

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

Re: [OpenSIPS-Users] explizit handling auf replyto

. 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: >>> &g

Re: [OpenSIPS-Users] explizit handling auf replyto

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 et

[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.

[OpenSIPS-Users] solution found Re: opensips register as ua => Re: forward register

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

[OpenSIPS-Users] opensips register as ua => Re: forward register

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, rig

Re: [OpenSIPS-Users] forward register

risk 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

Re: [OpenSIPS-Users] forward register

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

Re: [OpenSIPS-Users] forward register

: > 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 us

[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 inte

Re: [OpenSIPS-Users] register performance with sipp

.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 > poss

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 I

[OpenSIPS-Users] Forcing reinvite for t38 on different pstn gw

Hello list, 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. Has anybody a hint for me? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___

[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)

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...@grou

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 > wrote: >

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

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

[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 ___

[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 t

Re: [OpenSIPS-Users] Error using MySQL with opensips

; > 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-

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 f

Re: [OpenSIPS-Users] Error using MySQL with opensips

Hi, Maybe you have forgotten to create the tables for opensips? opensipsdbctl create .(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 Op

Re: [OpenSIPS-Users] 408 Timeout with X-Lite

Hi, Is your xlite taking the challenge it get from opensips to REGISTER again? The (blah) is just an cut&paste error? BR Uwe Gordon Ross schrieb: > On 02/07/2009 09:49, "Uwe Kastens" wrote: >> Strange so far. I cannot see any wrong configuration on a 1st view. >

Re: [OpenSIPS-Users] 408 Timeout with X-Lite

? BR Uwe Gordon Ross schrieb: > On 02/07/2009 09:13, "Uwe Kastens" 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

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. > > # o

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

Re: [OpenSIPS-Users] 1.6 and mysql => crash Re: DIALOG not deletedon BYE

; >> 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 f

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

Re: [OpenSIPS-Users] mysql database connection error

uot; 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 <mailto:ki...@kiste.org>> wrote: > > Hi Brett, > > Could be broken libs. I would start opensips

Re: [OpenSIPS-Users] mysql database connection error

7;,'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 : &g

Re: [OpenSIPS-Users] mysql database connection error

27;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&#

Re: [OpenSIPS-Users] mysql database connection error

0C28A928E2BE5EFD59FF20CB8705B31ACCF3730','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N',&

Re: [OpenSIPS-Users] mysql database connection error

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]: > ERR

[OpenSIPS-Users] src ip check on Register => Re: How to insert the IP address of user in radius request.

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:srci

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 l

Re: [OpenSIPS-Users] DIALOG not deleted on BYE

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 m

[OpenSIPS-Users] 1.6 and mysql => crash Re: DIALOG not deleted on BYE

ut 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 >> pas

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

Re: [OpenSIPS-Users] DIALOG not deleted on BYE

ule 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, >>>

[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

Re: [OpenSIPS-Users] No RADIUS traffic

en("/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 =>

Re: [OpenSIPS-Users] DIALOG not deleted on BYE

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 asu

Re: [OpenSIPS-Users] DIALOG not deleted on BYE

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

[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

Re: [OpenSIPS-Users] No RADIUS traffic

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 >

[OpenSIPS-Users] update Re: Some results => Re: register performance with sipp

Hello, I modified my setup and put the users in a radius flatfile. With this modifiction I was able to reach 1250 cps with db_mode=3 and 2000 cps with db_mode=2. So the database is the bottleneck. BR Uwe Uwe Kastens schrieb: > Hi, > > In my testing setup I am not able to reach more

Re: [OpenSIPS-Users] No RADIUS traffic

gnoring 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 > > >

[OpenSIPS-Users] Some results => Re: register performance with sipp

tag=4611 To: ;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 lik

Re: [OpenSIPS-Users] register performance with sipp

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

[OpenSIPS-Users] register performance with sipp

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 o

Re: [OpenSIPS-Users] Number portability

Hi Paul, Sounds interesting. Will you always send complete numbers to your operators? Sounds not very complicated if you use dialplan. @Bogdan: What would you use recommend regarding performance? BR Uwe Bogdan-Andrei Iancu schrieb: > Hi Paul, > > More or less yes :) - you need to put transla

Re: [OpenSIPS-Users] Basic call failed

-- > 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 l

Re: [OpenSIPS-Users] Basic call failed

nt 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 &g

Re: [OpenSIPS-Users] Basic call failed

t; > 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

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 wi

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 m

Re: [OpenSIPS-Users] update location table on REGISTER request

ry and how to build > it!! Also would there be all the features consistent that are present in > the 1.5.1 version. > > Thank you again, > > --- Jay > > On Thu, Jun 18, 2009 at 1:17 AM, Uwe Kastens <mailto:ki...@kiste.org>> wrote: > > Hi, > >

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