sr-users September 2007

sr-users@lists.kamailio.org

158 participants
247 discussions

[OpenSER-Users] how to register xlite user for testing with openser

by Live Great

Hi, I used the following command to add xlite user, but xlite failed to register the user in openser server. # openserctl add 1234 1234 sam(a)my.domain.com where 1234 is user id and password. If this is not the correct way to add user for testing from xlite to openser, what are the alternative way I can use? Thanks SW

17 years, 2 months

[OpenSER-Users] OpenSER + FreeRadius error

by Marc LEURENT

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Does this error means something to somebody?? Thanks ERROR:acc:acc_rad_request: radius-ing failed My openser.cfg # -- acc params (with radius )-- modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) ..... ################### SET RADIUS ACCOUTING FLAGS ################################# # Set the acc flags xlog("JUSTE BEFORE SETTING FLAGS FOR RADIUS\r\n"); if(is_method("INVITE") && !has_totag()) { xlog("L_INFO", "I AM SETTING THE FLAGS FOR RADIUS \r\n"); $avp(can_uri) = $ru; # SIP Request's URI $avp(billing_party) = "sip:" + $au + "@" + $ar; # From URI $avp(user_name) = $fu; xlog("!!!! FROM TAG = $ft\r\n"); xlog("SETTING FLAGS 1 & 2 FOR RADIUS \r\n"); setflag(1); # radius_flag setflag(2); # radius_missed_flag }; DEBUG OUTPUT: 3(26026) DEBUG: update_totag_set: new totag 3(26026) DEBUG: add_to_tail_of_timer[2]: 0x2b898c7eb9d0 (12) 3(26026) DEBUG:tm:relay_reply: sent buf=0x61d2e8: SIP/2.0 2..., shmem=0x2b898c7ed7a8: SIP/2.0 2 3(26026) DBG: trans=0x2b898c7eb950, callback type 128, id 0 entered 3(26026) ERROR:acc:acc_rad_request: radius-ing failed 3(26026) DEBUG: cleanup_uac_timers: RETR/FR timers reset 3(26026) DEBUG:tm:UNREF_UNSAFE: after is 0 3(26026) DEBUG:destroy_avp_list: destroying list (nil) 3(26026) receive_msg: cleaning up 1(26024) SIP Request: 1(26024) method: <ACK> 1(26024) uri: <sip:105@82.127.0.79:1030;user=phone;nat=yes> 1(26024) version: <SIP/2.0> 1(26024) parse_headers: flags=2 1(26024) Found param type 232, <branch> = <z9hG4bK2175363103192047157>; state=16 1(26024) end of header reached, state=5 1(26024) parse_headers: Via found, flags=2 1(26024) parse_headers: this is the first via -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG8jcCqjpLE0HiOBYRAlBWAJoD4mlTLi8MCSLJEceI4jUNP2LFSwCgnjQq ytiBJX6Q40CVgt+V0sB0JWk= =BVYV -----END PGP SIGNATURE-----

17 years, 2 months

[Serusers] osp toolkit

by Arya

Hello Does osp toolkit support billing? and does it have a web interface? -- Thank You

17 years, 2 months

[OpenSER-Users] 1.3 release schedule

by Bogdan-Andrei Iancu

Hi, Just as reminder about the 1.3 release schedule: 05.10.2007 - SVN freeze and start testing - please commit all new features or changes till this date 0x.12.2007 - release date (first days of December) Regards, Bogdan

17 years, 2 months

[OpenSER-Users] PA module

by Bogdan-Andrei Iancu

Hi, As we have in openser a new well designed presence support (see presence_* and pua_* modules), I think the PA module needs to be purged (anyhow it is not maintained). Does anybody has any good arguments not to do it? Regards, Bogdan

17 years, 2 months

[Serusers] Monitoring/logging SIP packets

by x-ser＠sidell.org

Sorry for the stupid question... Is there way to (temporarily) monitor and log the contents of all SIP packets being sent and received by my SER server? -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com

17 years, 2 months

[OpenSER-Users] question about force_rtp_proxy()

by Papadopoulos Georgios

Hello all, I have come accross some issue while using nathelper and rtpproxy. The senario is as follows: An INVITE is sent to a NATed client and rtpproxy is used. force_rtp_proxy() rewrites the SDP to something like c=IN IP4 213.5.43.4. m=audio 35034 RTP/AVP 18 100 101. The client replies 486 Busy and the script sends the request to an Asterisk which is used for voicemail. If force_rtp_proxy() is called a second time before sending to Asterisk then the result is a SDP with the following. c=IN IP4 213.5.43.4213.5.43.4. m=audio 3503435034 RTP/AVP 18 100 101. Obviously this can be avoided if I add the proper logic in the script to avoid the second call to force_rtp_proxy(). However this behaviour looks like a bug. As a second question, when the call goes to Asterisk, there is no need for rtpproxy. So how can someone revert the SDP to its original form? How do you usually handle this case? thank you in advance George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

17 years, 2 months

[OpenSER-Users] can not acc failed transactions

by Victor Gamov

Hi All I red ACC module manual many times but still can not understand what I need to do to account failed transactions (408 for example) I make following configuration but failed transactions still not accounted ---------- cut here ------------------ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "db_flag", 1) route[1] { setflag(1); t_on_failure("1"); t_relay(); } failure_route[1] { xlog("L_ERR", "failure_route[1]: METHOD=$rm; rc=$T_reply_code\n"); } ---------- cut here ------------------ Yes I can add some acc_db_request() into failure_route[1] but this function don't set sip_code field in DB. Please advise me before I start to read sources :-) -- CU, Victor Gamov

17 years, 2 months

[Serusers] Cannot open module mysq.so - undefined symbol: log

by Borja Diaz de Teran

*Using Redhat Enterprise *MySQL Version 14.12 Distrib 5.0.27, for pc-linux-gnu (i686) using readline 5.0 1) I've followed the instructions to install SER in conjunction with Mysql and I've even tried to compile in a different way (make group_include="standard standard-dep stable mysql" skip_modules="postgres" all) [root@filipides ser]# ser -f /usr/local/etc/ser/ser.cfg Segment violation I keep getting the same error: Sep 21 13:57:30 filipides ser: ERROR: load_module: could not open module /usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log Sep 21 13:57:30 filipides ser: parse error (81,13-14): failed to load module 2) LOG can be found in the following modules (apart from the ser exe):[root@filipides modules]# grep LOG * Match in cpl-c.so Match in dialog.so Match in mysql.so Match in pa.so Match in xlog.so 3) In order to check the existence of that pattern I did the following search:find . -name "*" -exec grep -l "LOG" {} \; , and it's inside (almost) all the files and objects compiled.implementación. The compilation of the MySQL module seems to be correct. gcc -shared -Wl,-O2 -Wl,-E dbase.o db_con.o db_mod.o my_con.o res.o row.o utils.o val.o -L/usr/lib/mysql -L/usr/local/lib -L/usr/local/lib/mysql -L/usr/local/mysql/lib/mysql/ -L/usr/local/mysql/lib -L/usr/lib64/mysql -lmysqlclient -lz -o mysql.so and it doesn't give any errors. I've tried changing all the LOG calls in all the .c files of the MySQL module to printf. I've compiled and install and nothing changes. I keep getting the same error.Do you have any clue of what I could be doing wrong?Thank you in advance. _________________________________________________________________ ¡Ya está aquí Windows Live Spaces! Ahora podrás crear fácilmente tu propio sitio Web. http://spaces.live.com/signup.aspx

17 years, 2 months

Re: [OpenSER-Users] Missing RTP stream

by Morten Isaksen

The error was in Mediaproxy. In rtphandler.py i changed nonPublicNetworks = [ {'name': '0.0.0.0', 'value': 0x00000000L, 'mask': 0xff000000L}, {'name': '10.0.0.0', 'value': 0x0a000000L, 'mask': 0xff000000L}, {'name': '127.0.0.0', 'value': 0x7f000000L, 'mask': 0xff000000L}, {'name': '172.16.0.0', 'value': 0xac100000L, 'mask': 0xfff00000L}, {'name': '192.168.0.0', 'value': 0xc0a80000L, 'mask': 0xffff0000L}, {'name': '224.0.0.0', 'value': 0xe0000000L, 'mask': 0xf0000000L} ] To nonPublicNetworks = [ {'name': '0.0.0.0', 'value': 0x00000000L, 'mask': 0xff000000L}, {'name': '10.0.0.0', 'value': 0x0a000000L, 'mask': 0xff000000L}, {'name': '127.0.0.0', 'value': 0x7f000000L, 'mask': 0xff000000L}, {'name': '224.0.0.0', 'value': 0xe0000000L, 'mask': 0xf0000000L} ] I think Mediaproxy got confused with the RFC1918 IP's. In my setup there is no NAT between 172.17.0.0/16 and 192.168.0.0/24 - just a router. On 9/21/07, Norman Brandinger <norm(a)goes.com> wrote: > Is there a firewall in the picture ? You have two different subnets and > there probably is a box doing some (NAT) translation / routing between > them. Is is possible the RTP stream is being blocked at the firewall ? > > Norm > > > Morten Isaksen wrote: > > Hi! > > > > I can see in the mediaproxy log the it is initialized to proxy the > > call, but I newer get a "session xxxxx: called signed in from xxx" > > from Asterisk. > > > > session.py shows that the the connection between mediaproxy and > > Asterisk is missing. > > > > I will try to take a look at the sip debug from asterisk and try to > > change the NAT settings in Asterisk. > > > > Thanks for your input. > > > > On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote: > > > >> You stated that you've forced every call through mediaproxy. Are you > >> positive ? > >> > >> Have you taken a look at the mediaproxy logs (and/or sessions.py when > >> the call is up) ? They might provide some useful information to you. > >> > >> Ditto for Asterisk "sip set debug on" (note that the sip debug command > >> format is a moving target). > >> > >> Have you looked at the "nat=" settings in sip.conf as well ? At times, > >> they tie closely with "canreinvite=". > >> > >> Norm > >> > >> > >> Morten Isaksen wrote: > >> > >>> Hi! > >>> > >>> canreinvite is set to no and the OpenSER/mediaproxy is "hiding" the > >>> clients IP-addresses from Asterisk, so I am pretty sure that this is > >>> not the issue. > >>> > >>> On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote: > >>> > >>> > >>>> Hi Morten, > >>>> > >>>> Admittedly, I haven't looked closely at your trace. However, based on > >>>> the description you gave, the first place to look is at the "canrevite" > >>>> setting in Asterisk sip.conf. You might want to try "canreinvite=no" > >>>> after reading up on this particular setting. > >>>> > >>>> Regards, > >>>> Norm > >>>> > >>>> > >>>> Morten Isaksen wrote: > >>>> > >>>> > >>>>> Hi! > >>>>> > >>>>> I have a strange problem with a missing RTP stream between OpenSER and > >>>>> Asterisk. I am not sure if it is OpenSER og Asterisk related. > >>>>> > >>>>> I have this setup > >>>>> > >>>>> Phone A (172.17.96.17) -- > >>>>> \ Openser -- Asterisk > >>>>> -- PSTN > >>>>> / (192.168.0.6) (192.168.0.3) > >>>>> Phone B (172.17.96.10) -- (172.17.64.1) > >>>>> > >>>>> I also have a Mediaproxy running on OpenSER and I force every call to > >>>>> use the Mediaproxy. > >>>>> > >>>>> I call from Phone A or B to the PSTN works fine and from PSTN to Phone > >>>>> A or B it also works. > >>>>> > >>>>> I have the dialplan logic on my Asterisk server so I want calls from > >>>>> Phone A to Phone B to pass the Asterisk server. And this is were I > >>>>> have the problem. When the call is established the RTP stream is > >>>>> missing between Mediaproxy and Asterisk. I only have a RTP stream > >>>>> between the phones and Mediaproxy. As far as I can see the SIP > >>>>> signalling is correct. > >>>>> > >>>>> The SIP traces is listed below. Can you spot the problem in this? > >>>>> > >>>>> I will buy a beer (or 5) at OpenSER training in Rome to anyone who can > >>>>> help me solve this problem. > >>>>> > >>>>> SIP trace between the phones and OpenSER: > >>>>> > >>>>> > >>>>> > >>> > >>> > >> > > > > > > > > -- Morten Isaksen http://www.misak.dk/blog/

17 years, 2 months

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sr-users September 2007