sr-users October 2007

sr-users@lists.kamailio.org

130 participants
285 discussions

by Jiri Kuthan

At 18:28 18/10/2007, Stefan.Brozinski(a)materna.de wrote: >Jiri, > >does that mean that there are no plans to support RFC 3841? First: what do you mean by "supporting RFC 3841" ... can't you script it? Secondly: whether as scripting or not, my individual plans are not as important as whether there is some individual on this mailing list willing to do it, there is really not a grand plan.... I think the most reasonable next step would be for someone desiring to have it to script it and share it. >This is sad because 3GPP's CSCFs require these headers, and the >OpenIMSCore project uses SER to implement its CSCFs. Well, I am unfortunately not involved in OpenIMSCore. -jiri >Regards >Stefan > > >> -----Original Message----- >> From: Jiri Kuthan [mailto:jiri@iptel.org] >> Sent: Thursday, October 18, 2007 4:03 PM >> To: Brozinski, Stefan; serusers(a)lists.iptel.org >> Subject: Re: [Serusers] RFC 3841 support? >> >> >> There is no explicit support for callerprefs. I think though >> that most of the scenarios >> they are goof for could be achieved using textops and >> selects+AVP processing. >> >> -jiri >> >> At 15:40 18/10/2007, Stefan.Brozinski(a)materna.de wrote: >> >Hello everybody >> > >> >are there any plans to support RFC 3841 in SER? >> > >> >Specifically, I am looking at the 'Accept-Contact' and the >> >'Request-Disposition' header fields. >> > >> >Thanks >> >Stefan >> >_______________________________________________ >> >Serusers mailing list >> >Serusers(a)lists.iptel.org >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >> >> -- Jiri Kuthan http://iptel.org/~jiri/

17 years, 1 month

[OpenSER-Users] sanitizing sip requests

by William Quan

Hi all, I came across a security alert that basically embeds javascript in the display name of the From to initiate cross-site-scripting (XSS) attacks. Here is an example: From: "<script>alert('hack')</script>""user" <sip:user at domain.com <https://lists.grok.org.uk/mailman/listinfo/full-disclosure>>;tag=002a000c Grammatically , I don't see an issue with this. However, under the right circumstances this could get ugly. Do you see value in having openser take a proactive role to detect these and reject calls? Or is this outside the scope of what a proxy should be doing (leave it to the UA to sanitize) ? Looking to get your thoughts- -will

17 years, 1 month

[OpenSER-Users] OpenSER SIP registrar - third party registration

by Pascal Maugeri

Hi I would like to make OpenSER SIP REGISTRAR to advertise my SIP server that a user just registered successfully to the domain. My server has to make a call to this user as soon as he's registered. First of all is this called "third party registration" ? In practice how does it work ? OpenSER sends a REGISTER to my server and my server replies with 200 OK ? Which script command should I use ? Thanks in advance for your help Pascal

17 years, 1 month

[OpenSER-Users] question regarding record-route and via

by vallimamod abdullah

Hello list, I am trying to understand why the thomson ST2030 hard phone does not work with my openser. I have noticed while ngreping that it seems to not respect the record-route header. I have the following incoming invite: U 2007/10/18 19:31:54.067545 xx.xx.33.119:5060 -> 10.0.0.136:15060 INVITE sip:1920@10.0.0.136:15060;user=phone SIP/2.0. Record-Route: <sip:xx.xx.33.119:5060;nat=yes;lr=on>. Via: SIP/2.0/UDP 192.168.5.119;branch=z9hG4bKae88.af6caa.0. Via: SIP/2.0/UDP 192.168.5.120:5060;branch=z9hG4bK58e93388;rport=5060. From: <sip:6897@192.168.5.120>;tag=as246d1f36. To: <sip:1920@192.168.5.119>. And the phone answers to the IP of the first via instead of the one in record-route: U 2007/10/18 19:31:54.083370 10.0.0.136:15060 -> 192.168.5.119:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.5.119;branch=z9hG4bKae88.af6caa.0. Via: SIP/2.0/UDP 192.168.5.120:5060;branch=z9hG4bK58e93388;rport=5060. From: <sip:6897@192.168.5.120>;tag=as246d1f36. To: <sip:1920@192.168.5.119>. I would like to know the opinion of SIP gourous out there: is this rfc-compliant ? I don't have this issue with other phones so I am wondering if I should ask for a refund... Thank you ! -- Regards, - vma .

17 years, 1 month

Re: [Serusers] RFC 3841 support?

by Jiri Kuthan

There is no explicit support for callerprefs. I think though that most of the scenarios they are goof for could be achieved using textops and selects+AVP processing. -jiri At 15:40 18/10/2007, Stefan.Brozinski(a)materna.de wrote: >Hello everybody > >are there any plans to support RFC 3841 in SER? > >Specifically, I am looking at the 'Accept-Contact' and the >'Request-Disposition' header fields. > >Thanks >Stefan >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

17 years, 1 month

[Serusers] Documentation search problem

by Greger V. Teigre

It just came to my attention that some pages on iptel.org do not function properly, including the doc search tool. While we try to rectify the problem, here's an alternative link: http://siprouter.onsip.org/doc/docsearch/ g-)

17 years, 1 month

Re: [Serusers] Get RealTime Online Users status

by roayu

Hi Jiri! I've captured and analyzed those SIP message dumps. I found that when Sjphone A sent a NOTIFY signal to the SER (after Sjphone A accepted the REFER), SER didn't pass the NOTIFY to the SPA942 to terminate the call between them. Therefore, Sjphone A unable to INVITE Sjphone B to bridge up the connection. May I know what causes this? Thanks for your precious time and reply! Cheers, Roa Yu :) -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Thursday, October 18, 2007 4:28 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Get RealTime Online Users status At 04:59 18/10/2007, roayu wrote: >Hi Jiri! > >Actually, I'm using Sjphone and SPA942 to do the call transfer. I'm calling >using Sjphone A to SPA942, then use the SPA942 to transfer the call to >Sjphone B. > > SJPhone A --> SPA942 --> SJPhone B > >After a short conversation with SJPhone B, then I press the 'xfer' button on >the 2nd time so that both SJPhone A and SJPhone B can talk to each other. >But the call couldn't be transferred to SJPhone B whereas when I used >Asterisk, it's able to do so. > >I've tried to disable the Digest-qop but it still couldn't perform the >transfer properly. What other settings that I need to configure on SER? I'm afraid you've gotta analyze SIP message dumps to figure out what's gone wrong. -jiri >Thanks and really appreciate on your reply. > >Cheers, >Roa Yu > > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Thursday, October 18, 2007 9:16 AM >To: roayu >Cc: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Get RealTime Online Users status > >At 10:37 10/10/2007, roayu wrote: >>Oh, I got it! Thanks Jiri! >> >>Btw, I have another questions. >>1) I would like to know can call transfer being done when the >>connection is P2P? > >you mean without a proxy? sure it can. > >> When I tried to use relay (mediaproxy) to do the call transfer, >it's >>able to transfer the call to the other party whereas when the condition >is >>P2P, it just can't pass the correct signal. > >that's apparently unrelated to use of media proxy. you would have to >check the SIP dumps what's going wrong. Unfortunately I can't promise >I would help to review those -- my to-do-list is endless these days. >Perhaps someone else on the list will. > > >>2) Can SER support SPA942 ? Or is there some settings that I need to >>configure on SER ? > >yes. >well for some SPAs you may need to disable digest-qop since they have a bug. >(don't ask me which though). > >-jiri > > > >>Thanks. >> >>Cheers, >>Roa Yu >> >>-----Original Message----- >>From: Jiri Kuthan [mailto:jiri@iptel.org] >>Sent: Wednesday, October 10, 2007 2:56 PM >>To: roayu; SIP >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Get RealTime Online Users status >> >>yes, set usrloc's database mode to 1. -jiri >> >>At 03:32 10/10/2007, roayu wrote: >> >>>Thanks for your reply. I found that it's only update after around 30 >>>seconds. Is there anyway that I can fasten the update on the database? >>> >>>Thanks. >>> >>>Cheers, >>>Roa Yu >>> >>>-----Original Message----- >>>From: SIP [mailto:sip@arcdiv.com] >>>Sent: Tuesday, October 09, 2007 9:14 PM >>>To: roayu >>>Cc: serusers(a)lists.iptel.org >>>Subject: Re: [Serusers] Get RealTime Online Users status >>> >>>roayu wrote: >>>> >>>> Hi there! >>>> >>>> Can anyone tell me how to get the RealTime Online user status other >>>> than using command '*serctl ul show*'? Or how can I store the realtime >>>> online user status to the MySQL db? >>>> >>>> Thanks. >>>> >>>> Cheers, >>>> >>>> Roa Yu J >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> Serusers(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>The location table in the MySQL database stores what are, in effect, >>>currently online users. It is, however, only an approximation. When a UA >>>registers, it sends an expiration time on its registration and that gets >>>stored in the location table. If the user disconnects without cancelling >>>his registration (some UAs send am expire register message to 'log out' >>>and some don't), then the data may still be in the table until the >>>expire time occurs. Realistically, though, you can get a good idea of >>>currenly online users using the location table and while it may not be >>>100% accurate, it's close enough for government work, as it were. >>> >>>N. >>>_______________________________________________ >>>Serusers mailing list >>>Serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >>-- >>Jiri Kuthan http://iptel.org/~jiri/ >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > > > >-- >Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

17 years, 1 month

[OpenSER-Users] General protection error

by Marcello Lupo

Hi to all, i've got this error in logs and openser crashed: [20670]: force_rtp_proxy2: incorrect port 0 in reply from rtp proxy [20670]: kernel: openser[20670] general protection rip:2b467ccf4150 rsp:7fff2e6d6370 error:0 [20670]: child process 20670 exited by a signal 11 [20670]: core was generated [20630]: INFO: terminating due to SIGCHLD [20640]: INFO: signal 15 received .... .... .... Can you remember me where the core is and if you need it to be analyzed? Openser is 1.2.2 no tls on SLES10 x86_64 . Thanks, Bye, Marcello

17 years, 1 month

[Serusers] RTPProxy question

by Mitya

Hi all, I have a few unclarified questions in my head about RTPProxy. I think I know the basic concept, but I'm curious to know a few details. How does the proxy transfer the packets from A to B ? When it gets a packet from A, does it send it immediately to B (when its possible) without any delay or timing ? Or does it have a buffer (few ms buffer), where it re-orders the packets ? Does it parse the RTP stream ? Does it generate RTCP packets ? Can somebody help me in this ? Thanks, Mitya

17 years, 1 month

[OpenSER-Users] "200 OK" not SDP fixed by RtpProxy... but it works ! ¿?

by Iñaki Baz Castillo

Hi, user_A with STUN calls to user_B behind NAT with no STUN. onreply_route[1] { if (nat_uac_test("1")) fix_nated_contact(); if (isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") force_rtp_proxy("l"); } In the initial request bflag(6) is up because "location" of user_B so RtpProxy is used in the INVITE and 200-OK. But in re-INVITE bflag(6) is down and is not applied force_rtp_proxy("l"); in the 200-OK. In fact I debug the SIP trace in the re-INVITE in OpenSer: ---------------------------------------------------------------------- # user_A -> OpenSer INVITE sip:user_B@112.121.235.28:5061 SIP/2.0 c=IN IP4 212.121.235.18 # OpenSer -> user_B INVITE sip:806@212.121.235.18:5061 SIP/2.0 c=IN IP4 80.94.0.110 <---- RtpProxy applied (OK) # user_B -> OpenSer SIP/2.0 200 OK c=IN IP4 192.168.1.106 [*1] *1: Needs RtpProxy but bflag(6) is down so not applied. # OpenSer -> user_A SIP/2.0 200 OK c=IN IP4 192.168.1.106 <-- RtpProxy NO applied ---------------------------------------------------------------------- But I can confirm that the audio works after re-INVITE in both directions!!! ¿? I've debugged with tcpdump, the RTP is send to RtpProxy from user_A, how is possible?? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

17 years, 1 month

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sr-users October 2007