- sr-users - kamailio.org

[OpenSER-Users] Re: Users Digest, Vol 29, Issue 61
by vallimamod abdullah 20 Oct '07

20 Oct '07

On 10/19/07, users-request(a)openser.org <users-request(a)openser.org> wrote on behalf of Iñaki Baz Castillo <ibc(a)in.ilimit.es>: > > > My openser have a private IP (192.168.5.119) statically natted to > > xx.xx.33.119; so the first via contains the private IP. Is it up to the > > client to figure out how to manage this or is there a way to handle this > > in openser ? > > > Look at the "Via" headres in both cases, they are the same. That's right but the difference is that the second invite reply (from Sipura) arrives to the server and the first one from Thomson don't. > You have an issue because NAT. The Thomson INVITE leaves OpenSer natted but > the Via header added by OpenSer says "192.168.5.119". You should fix the Via > header for messages that will be natted, but sincerely I don't know how. Thank you for this hint. It reminded me a wonderful option in openser core called "advertised_address" where you can set what you want to see in the first Via. So I set my domain name there. It is resolved with the public IP from outside and with the lan IP from the inside and now it works from both sides ! -- Regards, - vma .

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[Users] Re: 200 ok not cought in ONREPLY
by Dan-Cristian Bogos 19 Oct '07

19 Oct '07

Guys, thanks to your support (both Bogdan and Daniel), I found out that there was nothing wrong with the server and t_onreply handling, but the INVITE was taking another routing three than the one I was checking and therefore never hit the t_on_reply section I wanted. This being said, I consider this issue resolved. Cheers, Dan

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[OpenSER-Users] LCR
by Gerson Matiolli 19 Oct '07

19 Oct '07

Hi all I am using Fedora 7 and installed "openser - 1.2.2 - 3.fc7.i386.rpm" and I am using mysql. I need to increase the table LCR, 256 entries for 8000 entries, as do that?

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Re: [OpenSER-Users] Manipulation of the From header username
by Jiri Kuthan 19 Oct '07

19 Oct '07

At 18:30 19/10/2007, samuel wrote: >SIP version 1 (2543) used From header to identify the dialog so it's not recommended to change the From header to be backwards compatible (is there any RFC 2543 SIP endpoint around??...). >SIP Version 2 (RFC 3261) uses tags to identify dialogs so you should be able to change the FRom header except the tag. and mandates backwards compatibility at the same time, which gets you back to square 1. >Recently there was an RFC 4916 that specifies how to identify the connected identity and how to make a change in the From but I think this is far beyond what you are trying to achieve. I'm adraid there are too many fascinating extensions in such quantity that many have stopped taking care of those (sometimes for a good reason) >There are additional headers (Remote_PartyID, and related) that can be used to specify an identify of the calling party.. asserted identity being most adequate nowadays, I guess. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Get RealTime Online Users status
by Michal Matyska 19 Oct '07

19 Oct '07

Looks you have SIP ALG enabled on your NAT, which breaks the call transfer (no surprise, I haven't seen working SIP ALG yet). Check OPTIONS request sent by SJphone and received by SER, check NOTIFY sent by SJphone and received by SER. Disable the SIP ALG and try again. If you can't disable it, change the NAT box itself. Michal On Fri, 2007-10-19 at 14:42 +0800, roayu wrote: > Hi Michal, > > I've attached the config file as well as the network capture so that you can > help me to find out the problem. But I'm not sure whether I will gonna guide > u to the answer 42 or not. :p > > Thanks for your help! > > > Cheers, > Roa Yu > > > -----Original Message----- > From: Michal Matyska [mailto:michal@iptel.org] > Sent: Friday, October 19, 2007 1:17 PM > To: roayu > Cc: Jiri Kuthan; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Get RealTime Online Users status > > Hi, > > unless you post the network capture and your config file (but make sure > you don't disclose anything you want to keep secret) there is nobody who > can help you. > > The only answer is: your script is broken, except the answer 42, but I > don't think your question was the ultimate question about life, universe > etc. ;-) > > Michal > > P.S. If you haven't read "Hitchhikers guide to the universe" don't panic > and just forget the answer 42. > > On Fri, 2007-10-19 at 11:35 +0800, roayu wrote: > > 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/ > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

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[OpenSER-Users] segfault in presence moudle of openser-1.2 branch
by Klaus Darilion 19 Oct '07

19 Oct '07

Hi Anca! #0 0x00e030e3 in update_subscription (msg=0x8162b68, subs=0xbfe24ef0, rtag=0xbfe24e08, to_tag_gen=0) at subscribe.c:359 359 record_route.len= strlen(record_route.s); (gdb) (gdb) (gdb) (gdb) bt #0 0x00e030e3 in update_subscription (msg=0x8162b68, subs=0xbfe24ef0, rtag=0xbfe24e08, to_tag_gen=0) at subscribe.c:359 #1 0x00e065f8 in handle_subscribe (msg=0x8162b68, str1=0x0, str2=0x0) at subscribe.c:1294 #2 0x080528d0 in do_action (a=0x815d518, msg=0x8162b68) at action.c:891 #3 0x08054b6c in run_action_list (a=0x815d438, msg=0x8162b68) at action.c:131 #4 0x08054258 in do_action (a=0x815d5c8, msg=0x8162b68) at action.c:815 #5 0x08054b6c in run_action_list (a=0x815ba20, msg=0x8162b68) at action.c:131 #6 0x08054d9f in run_top_route (a=0x815ba20, msg=0x3) at action.c:111 #7 0x0807cf23 in receive_msg ( buf=0x81253e0 "SUBSCRIBE sip:12.34.56.246:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 80.75.44.244;branch=z9hG4bK5eb5 .405f258.0\r\nVia: SIP/2.0/UDP 10.241.131.64:23308;received=194.48.128.75;branch=z9hG4bK-d87543-537e85345d6f75 26-"..., len=601, rcv_info=0xbfe25720) at receive.c:156 #8 0x080a6935 in udp_rcv_loop () at udp_server.c:451 #9 0x0806b966 in main_loop () at main.c:834 #10 0x0806da0e in main (argc=7, argv=0xbfe25904) at main.c:1393 The crash happened when processing a SUSBCRIBE without Record-Route header: SUBSCRIBE sip:12.34.56.246:5060 SIP/2.0 Via: SIP/2.0/UDP 12.34.56.244;branch=z9hG4bK5eb5.405f258.0 Via: SIP/2.0/UDP 10.123.123.64:23308;received=123.45.123.75;branch=z9hG4bK-d87543-537e85345d6f7526-1--d87543-;rport=11609 Max-Forwards: 68 Contact: <sip:a.foobar@123.45.123.75:11609;transport=udp> To: <sip:a.foobar@mydomain.at>;tag=mango.1441.1192697556.19 From: "foobarfoob"<sip:a.foobar@mydomain.at>;tag=5224f96c Call-ID: ZGU0ZGE4NjQzYzRjZGYxZWNhNjM5N2U2MDY0MTZlY2E. CSeq: 4 SUBSCRIBE Expires: 0 User-Agent: foobarfoobarfoobarfoobarfoobarfoobar Event: presence.winfo Content-Length: 0 regards Klaus

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[OpenSER-Users] Strange behavior for openser 1.3 in mhomed mode (automatic mangling of Contact header and c= line of SDP)
by Ovidiu Sas 19 Oct '07

19 Oct '07

Hi all, I was playing with openser 1.3 in a multihomed setup. In this setup, I noticed that by default, openser is mangling the Contact header and the sdp. I stripped down the config to bare minimum to see if it was something related to my config, but it seems that this is not related to the config. Here's the stripped down config that I used: # ----------- global configuration parameters ------------------------ debug=3 fork=yes log_stderror=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) auto_aliases=no listen=udp:br0 listen=udp:eth1 children=1 mhomed=1 disable_tcp=yes disable_dns_blacklist=true # ------------------ module loading ---------------------------------- mpath="/opt/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" # -- tm params -- modparam("tm", "fr_inv_timer", 100) modparam("tm", "fr_inv_timer_avp", "$avp(i:26)") modparam("tm", "fr_timer", 10) modparam("tm", "fr_timer_avp", "$avp(i:25)") modparam("tm", "noisy_ctimer", 1) modparam("tm", "onreply_avp_mode", 1) loadmodule "rr.so" # ------------------------- request routing logic ------------------- # main routing logic route{ if (!method=="REGISTER") { record_route(); } if (loose_route()) { route(1); }; if (!uri==myself) { route(1); }; if (uri==myself) { sl_send_reply("404", "Not Found"); exit; }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } Here's the network setup: <asterisk box> --- < openser > --- <SIP server> <192.168.2.20> --- <192.168.2.1 : EXTERNAL_IP> --- <SIP_SERVER> And here's an ngrep of the initial INVITEs: 23:53:31.439678 IP 192.168.2.20.sip > 192.168.2.1.sip: SIP, length: 819 INVITE sip:callee@SIP_SERVER SIP/2.0 Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK4f70ea70 From: "Ovi Sas" <sip:caller@SIP_SERVER>;tag=as048c5bed To: <sip:callee@SIP_SERVER> Contact: <sip:caller@192.168.2.20> Call-ID: 28aedce563bc57dc2cac78967e69c141@SIP_SERVER CSeq: 102 INVITE User-Agent: asterisk Max-Forwards: 70 Date: Thu, 18 Oct 2007 04:23:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 8560 8560 IN IP4 192.168.2.20 s=session c=IN IP4 192.168.2.20 t=0 0 m=audio 34028 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 23:53:31.454727 IP EXTERNAL_IP.sip > SIP_SERVER.sip: SIP, length: 1011 INVITE sip:callee@SIP_SERVER SIP/2.0 Record-Route: <sip:EXTERNAL_IP;r2=on;lr;ftag=as048c5bed> Record-Route: <sip:192.168.2.1;r2=on;lr;ftag=as048c5bed> Via: SIP/2.0/UDP EXTERNAL_IP:5060;branch=z9hG4bKea4e.7aacc885.0 Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK4f70ea70 From: "Ovi Sas" <sip:caller@SIP_SERVER>;tag=as048c5bed To: <sip:callee@SIP_SERVER> Contact: <sip:caller@EXTERNAL_IP:5060> Call-ID: 28aedce563bc57dc2cac78967e69c141@SIP_SERVER CSeq: 102 INVITE User-Agent: asterisk Max-Forwards: 70 Date: Thu, 18 Oct 2007 04:23:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 286 v=0 o=root 8560 8560 IN IP4 192.168.2.20 s=session c=IN IP4 EXTERNAL_IP t=0 0 m=audio 34028 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Please note the Contact header and the c= line inside SDP. Is there a way to bypass this behaviour? Using the nathelper module doesn't seem to help (the Contact header will still carry the EXTERNAL_IP. Regards, Ovidiu Sas

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[OpenSER-Users] Problem with python-application when installing OpenXCAP
by Antoine Fressancourt 19 Oct '07

19 Oct '07

Hello, I try to install OpenXCAP on a Ubuntu server Feisty machine. I am up to the point of installing the python-application 1.0.9, but when I try to install it with python setup.py install, I get the following error : byte-compiling /usr/lib/python2.5/site-packages/application/python/queue.py to queue.pyc File "/usr/lib/python2.5/site-packages/application/python/queue.py", line 9 v v v v v v v ^ SyntaxError: invalid syntax As I have absolutely no experience with Python, I would like to know how I could work around this error. Thank you in advance for your help, Antoine

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[OpenSER-Users] New OpenXCAP release 0.9.7
by Adrian Georgescu 19 Oct '07

19 Oct '07

Hello, There is a new OpenXCAP release available. openxcap (0.9.7) unstable; urgency=low * Prevented Twisted from printing SQL login info in the log * Added some info to INSTALL about tested versions openxcap (0.9.6) unstable; urgency=low * Fixed trusted peer authentication bug against the latest Twisted svn trunk. openxcap (0.9.5) unstable; urgency=low * Changed watcher application to reflect new database column names of OpenSER revision 2923 * Changed default SQL table name from "xcap" to "xcap_xml" openxcap (0.9.4) unstable; urgency=low * Added "watchers" application to query OpenSER presence watchers. * Include debian directory in the source archive. openxcap (0.9.3) unstable; urgency=low * Adjusted OpenXCAP to use python-application >= 1.0.9 * Set TLS mode based on http/https scheme of the root URI. * document_validation defaults to Yes in config.ini.sample openxcap (0.9.2) unstable; urgency=low * Changed the name and structure of xcap_xml table. * Added the 'type' param to refreshWatchers OpenSER command. * Added configuration option to disable XML schema validation. * Updated XML scheme to allow EyeBeam resource lists to pass validation. * Update debian dependencies. The software can be downloaded from: http://openxcap.org Kind regards, Adrian Georgescu

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Re: [Serusers] RFC 3841 support?
by Jiri Kuthan 19 Oct '07

19 Oct '07

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/

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[OpenSER-Users] sanitizing sip requests
by William Quan 19 Oct '07

19 Oct '07

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

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[OpenSER-Users] OpenSER SIP registrar - third party registration
by Pascal Maugeri 19 Oct '07

19 Oct '07

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

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[OpenSER-Users] question regarding record-route and via
by vallimamod abdullah 19 Oct '07

19 Oct '07

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 .

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Re: [Serusers] RFC 3841 support?
by Jiri Kuthan 19 Oct '07

19 Oct '07

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/

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[Serusers] Documentation search problem
by Greger V. Teigre 19 Oct '07

19 Oct '07

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-)

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Re: [Serusers] Get RealTime Online Users status
by roayu 19 Oct '07

19 Oct '07

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/

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[OpenSER-Users] General protection error
by Marcello Lupo 18 Oct '07

18 Oct '07

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

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[Serusers] RTPProxy question
by Mitya 18 Oct '07

18 Oct '07

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

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[OpenSER-Users] "200 OK" not SDP fixed by RtpProxy... but it works ! ¿?
by Iñaki Baz Castillo 18 Oct '07

18 Oct '07

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

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[OpenSER-Users] received registration reply
by Asim Riaz 18 Oct '07

18 Oct '07

Hi, I am relaying all registration to another proxy server using t_relay, any way to save the reply from that proxy to my "location" if the authentication is successful. Regards Riaz

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[Serusers] SER 0.9.6 and MySQL connection failed
by Mitya 18 Oct '07

18 Oct '07

Hi all, I'm running SER 0.9.6 on a Debian Etch, compiled it from source, using etch's libmysqlclient5 to connect a mysql server 5.0.20 (running on a different server, but on the same LAN) >From my ser.cfg: modparam("usrloc", "db_mode", 1) modparam("usrloc", "timer_interval", 10) modparam("usrloc", "use_domain", 1) modparam("usrloc", "db_url", "mysql://ser:xxxxx@x.x.x.x:3306/ser") When I start SER, everything seems to be fine for a while. After about one day, when I unplug my UAC and the registration expires, SER can not remove it from the database because it says: Oct 18 14:47:07 sip ser[21813]: Binding '1111111(a)1.2.3.4','sip:1111111@2.3.4.5:1164' has expired Oct 18 14:47:07 sip ser[21813]: submit_query(): mysql_ping failed Oct 18 14:47:07 sip ser[21813]: submit_query(): MySQL server has gone away Oct 18 14:47:07 sip ser[21813]: db_delete: Error while submitting query Oct 18 14:47:07 sip ser[21813]: db_del_ucontact(): Error while deleting from database Oct 18 14:47:07 sip ser[21813]: wt_timer(): Error while deleting contact from database If I restart SER, everything works fine (for a while) again. I'm sure that _NO_ONE_ was restarted the MySQL server during the test period. I can reproduce it at any time. Is it a known issue, or do I have a special case ? Every comments are welcome. Thanks, Mitya

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[Serusers] RFC 3841 support?
by Stefan.Brozinski＠materna.de 18 Oct '07

18 Oct '07

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

1 0

[OpenSER-Users] Regarding on Implementation the Music On Hold With OpenSER as SIP Proxy Server.
by raviprakash sunkara 18 Oct '07

18 Oct '07

---------- Forwarded message ---------- From: RaviPrakash <ravi.prakash(a)troopsoftware.com> Date: Oct 18, 2007 3:25 PM Subject: Regarding on Implementation the Music On Hold With OpenSER as SIP Proxy Server. To: adamlinford(a)oralnet.co.uk Cc: sunkara.raviprakash.feb14(a)gmail.com Hello Regarding on Implementation the Music On Hold with Opener as SIP Proxy. As I know OpenSER is not Media Proxy Server. But I want to implement below call flow, that Bob has to Send the REFER method To Media Server ( Asterisk ). But I Don't want this ( Alice --invite_bob---> OpenSER ---- invite_bob-->Asterisk-----invite_bob---> openser---> Bob ) this scenario will work .. That User Agent has to send the REFER method to OpenSER Proxy through --> Asterisk. Alice Bob Media Proxy.. *Plz let me know, If anything is missed or something is else...* *Thanks & Regards* *Ravi Prakash Sunkara* *Troop Software* -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead

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Re: [Serusers] ser 2_0_0 lookup_user($tu.uid, @ruri) and handling of user names with unescaped semicolon.
by Jiri Kuthan 18 Oct '07

18 Oct '07

no, it is saying that "sip:alice;day=<mailto:tuesday@atlanta.com>tuesday(a)atlanta.com" is the username (consistiing of semicolon, equal sign, etc.) A strange but a valid one. The only deficit to my knowledge is not expanding % escapes. (perhaps you want to report it in JIRA?) -jiri p.s. some of the questions you had asked previously are addressed in the new oob.cfg -- feel free to check it out and give feeback. At 11:36 18/10/2007, Tomasz Zieleniewski wrote: >Hi, > >According to rfc 3261: > > > >19.1.3 Example SIP and SIPS URI > > > > sip:alice;day=<mailto:tuesday@atlanta.com> >tuesday(a)atlanta.com > > The last sample URI above has a user field value of > "alice;day=tuesday". The escaping rules defined above allow a > semicolon to appear unescaped in this field. For the purposes of > > this protocol, the field is opaque. The structure of that value is > only useful to the SIP element responsible for the resource. > > >in such case shouldn't the lookup_user($tu.uid, @ruri) method analyze only the username part before semicolon >for the lookup?? > >Waiting for Your Feedback >Bests >Tomasz >_______________________________________________ >Serusers mailing list >Serusers@lists.iptel.orghttp://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] : Regarding on Implementation the Music On Hold With OpenSER as SIP Proxy Server.
by raviprakash sunkara 18 Oct '07

18 Oct '07

Hello Regarding on Implementation the Music On Hold with Opener as SIP Proxy. As I know OpenSER is not Media Proxy Server. But I want to implement below call flow, that Bob has to Send the REFER method To Media Server ( Asterisk ). But I Don't want this ( Alice --invite_bob---> OpenSER ---- invite_bob-->Asterisk-----invite_bob---> openser---> Bob ) this scenario will work .. That User Agent has to send the REFER method to OpenSER Proxy through --> Asterisk. Alice Bob Media Proxy.. *Plz let me know, If anything is missed or something is else...* *Thanks & Regards* *Ravi Prakash Sunkara* VoIP Development Tech Lead

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