- sr-users - kamailio.org

[Serusers] Performance counters
by Parminder.Nayyar＠relianceada.com 04 Aug '07

04 Aug '07

Hi all Anyone here has any idea as to how we can keep the performance counters - ie the number of INVITEz recieved, the number of successfull responses to INVITE rcvd n so on... so that a performance metric cud be calculated as to how many calls got thru.. hwo many got thru voice, how many thru signalling only, n other such things??? Dint find ne module so far that cud help out in this... Has neone got ne idea?? Thanks... Parminder Nayyar

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[Serusers] Problem with TM initialialization
by Fernando Schmitt 04 Aug '07

04 Aug '07

Guys, I'm receiving the following output when running ser: 0(5705) Aug 03 06:58:26 DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(5705) Aug 03 06:58:26 DEBUG: MD5 calculated: 103d8e7e9fc1a3538c1143a96b22bab4 0(5705) Aug 03 06:58:26 DEBUG: init_mod: tm 0(5705) Aug 03 06:58:26 TM - initializing... 0(5705) Aug 03 06:58:26 Call-ID initialization: '79dff3eb' 0(5705) Aug 03 06:58:26 DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(5705) Aug 03 06:58:26 DEBUG: register_fifo_cmd: new command (t_hash) registered 0(5705) Aug 03 06:58:26 DEBUG: lock_initialize: lock initialization started 0(5705) Aug 03 06:58:26 init_mod(): Error while initializing module tm ERROR: error while initializing modules Do you know why? Thanks!

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Re: [OpenSER-Users] MediaProxy 2.0 roadmap
by Alexandre dos Reis Ferreira 03 Aug '07

03 Aug '07

Hi Adrian, I'd like a feature which allow get in real time the MOS of each call in the media sessions interface. Regards, Areis. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.11.2/933 - Release Date: 2/8/2007 14:22

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[OpenSER-Users] openser+asterisk question
by Radu Spineanu 03 Aug '07

03 Aug '07

Hi, When deploying a multi-location openser+asterisk architecture what is the best way of storing the user accounts? I thought about separating each location with a different prefix and having a mysql DB for each one. Then tell openser to route according to the prefixes. I like this version because it means if one location fails it still works locally. And the other servers are not affected. But this brings a problem with centralizing CDR. And if I want to interconnect with other VoIP networks I would have to pick a central location, right? Thanks, Radu Spineanu

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[OpenSER-Users] Adding an attribute to SDP
by Kuddusi 03 Aug '07

03 Aug '07

Hi, How can I add a header to SDP? Actually what I want to do is, whenever an INVITE with SDP comes, search for a ptime and if ptime can not be found, add a default line to SDP body. Like if (has_body()){ if (!search_body("ptime") { # like below à # append_body("a=ptime:20"); } } Thanks, Kuddusi CIFTCIBASI Next Generation Networks Teletek Telecommunication Services Corp. Ayazma Dere Cad. Aksit Plaza No:12/1 Fulya, Besiktas , 34349, www.teletek.net Tel : +90 212 227 7030 Direkt: +90 212 310 2233 Fax : +90 212 227 8700 Email: kuddusi.ciftcibasi(a)teletek.net Skype : realkudu MSN : kuddusic(a)hotmail.com Yahoo: kuddusi.

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[OpenSER-Users] error loading presence_xml module
by Babon Frederic 03 Aug '07

03 Aug '07

Hello, I have a problem when i try to add "presence_xml" module : ERROR : load_module: could not open module (/lib/openser/modules/presence_xml.so) : /lib/openser/modules/presence_xml.so : undefined symbol : add_event Do you have any idea about this error? Here is an extract of my openser configuration file: loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # -- presence params -- modparam("presence", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence_xml", "xcap_table", "xcaps") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'a') #modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence_xml", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:localhost:5065") modparam("presence_xml", "pidf_manipulation", 1) Thanks for your help. Frederic.

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[Serusers] loose_route() - loose_route: no routing target is local??
by tzieleniewski 03 Aug '07

03 Aug '07

Hi, When I invoke the loose_route() function I get the following info: loose_route: no routing target is local Even though the message contains the Route header fields the functions returns false. Please point me what do I miss here?? Thanks in advance. Cheers Tomasz

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[OpenSER-Users] STUN working!!
by Marc LEURENT 03 Aug '07

03 Aug '07

Hello everybody, To remind you, I'm trying to set a SIP architecture for NATTED SIP users without using any rtp proxy (because of lack of ressource). STUN is working very well between 2 users in 2 differents NATs. But when they are behing the same NAT, they cannot reach the other.. In the INVITE and 200Ok SDP fields, they have put their public address with a port reserved with STUN: it's not working! I think that it is because it's difficult for them to reach the public IPaddress/port where they should send the RTP stream from inside the NAT. So my idea was: Why not rewriting SDP fields to put their private address when they are behind the same NAT But I don't know how to rewrite the SDP fields for both users. (I have the private address of both users in the location database) Help would be very grateful... If you have any idea! Best Regards

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[OpenSER-Users] Not possible variables or AVP's in alias_db_lookup() ?
by Iñaki Baz Castillo 03 Aug '07

03 Aug '07

I want OpenSer to dinamically look for alias in a table for each domain, so I've in my database tables with name as: dbaliases_sip1_domain_org dbaliases_sip2_domain_com dbaliases_sip3_domain_net And I do (thanks to Andreas Granig): $avp(s:dom) = $rd; avp_subst("$avp(s:dom)", "/\./_/g"); alias_db_lookup("dbaliases_" + $avp(s:dominio)); but it fails because the SQL query is literally: select username,domain from dbaliases_$avp(s:dominio) where ... I've tryed putting variables and more things into the function parameter, but it's never processed and the SQL query is always done literally. I use OpenSer 1.2.1. What can I do for solving this? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] NAT + STUN
by Marc LEURENT 03 Aug '07

03 Aug '07

Do you have any idea to solve this matter: In the BYE request: 0(2569) found end of header 0(2569) find_next_route: No next Route HF found 0(2569) after_loose: No next URI found So the BYE message is not forwarded by openser Please find below INVITE and BYE requests 0(2632) SIP Request: 0(2632) method: <INVITE> 0(2632) uri: <sip:103@sd-7501.dedibox.fr:5060> 0(2632) version: <SIP/2.0> 0(2632) parse_headers: flags=2 0(2632) Found param type 232, <branch> = <z9hG4bK4747925369759203710>; state=16 0(2632) end of header reached, state=5 0(2632) parse_headers: Via found, flags=2 0(2632) parse_headers: this is the first via 0(2632) After parse_msg... 0(2632) preparing to run routing scripts... 0(2632) parse_headers: flags=100 0(2632) DEBUG:parse_to:end of header reached, state=10 0(2632) DBUG:parse_to: display={}, ruri={sip:103@sd-7501.dedibox.fr:5060;user=phone} 0(2632) DEBUG: get_hdr_field: <To> [46]; uri=[sip:103@sd-7501.dedibox.fr:5060;user=phone] 0(2632) DEBUG: to body [<sip:103@sd-7501.dedibox.fr:5060;user=phone> ] 0(2632) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(2632) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2632) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 10 0(2632) ROUTE2: STARTING NAT DETECTION 0(2632) !!!!!!!!! NAT UAC TEST 19 SUCEDEED 0(2632) parse_headers: flags=80 0(2632) ROUTE2: SETFLAG 3 0(2632) DEBUG: add_param: tag=c0a80101-b67ff5 0(2632) DEBUG:parse_to:end of header reached, state=29 0(2632) DBUG:parse_to: display={"101"}, ruri={sip:101@sd-7501.dedibox.fr:5060;user=phone} 0(2632) parse_headers: flags=200 0(2632) DEBUG: get_hdr_body : content_length=269 0(2632) found end of header 0(2632) find_first_route: No Route headers found 0(2632) loose_route: There is no Route HF 0(2632) DEBUG: has_totag: no totag 0(2632) I AM SETTING THE FLAGS FOR RADIUS 0(2632) SETTING FLAGS 1 & 2 FOR RADIUS 0(2632) CHECKING IF URI <> myself 0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91] 0(2632) grep_sock_info - checking if port 5060 matches port 5060 0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91] 0(2632) grep_sock_info - checking if port 5060 matches port 5060 0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91] 0(2632) grep_sock_info - checking if port 5060 matches port 5060 0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91] 0(2632) grep_sock_info - checking if port 5060 matches port 5060 0(2632) rewrite_uri: Rewriting Request-URI with 'sip:103@82.127.0.79:1028;user=phone' 0(2632) parse_headers: flags=ffffffffffffffff 0(2632) STARTING ROUTE 1 0(2632) subst_run: running. r=1 0(2632) subst_str: no match 0(2632) FLAG 3 OK GOTO ROUTE 3 0(2632) !!!!!!!!! ON ROUTE 3 FOR NATTED CONTACT 0(2632) DEBUG: t_newtran: T on entrance=0xffffffff 0(2632) parse_headers: flags=ffffffffffffffff 0(2632) parse_headers: flags=78 0(2632) t_lookup_request: start searching: hash=12532, isACK=0 0(2632) DEBUG: RFC3261 transaction matching failed 0(2632) DEBUG: t_lookup_request: no transaction found 0(2632) DBG: trans=0xb5c08fa8, callback type 1, id 1 entered 0(2632) trace_onreq_in: trace off... 0(2632) DBG: trans=0xb5c08fa8, callback type 1, id 0 entered 0(2632) parse_headers: flags=78 0(2632) DEBUG: noisy_timer set for accounting 0(2632) DEBUG:rr:is_direction: param ftag not found 0(2632) parse_headers: flags=ffffffffffffffff 0(2632) check_via_address(82.127.0.79, 82.127.0.79, 0) 0(2569) SIP Request: 0(2569) method: <BYE> 0(2569) uri: <sip:101@82.127.0.79:1312> 0(2569) version: <SIP/2.0> 0(2569) parse_headers: flags=2 0(2569) Found param type 232, <branch> = <z9hG4bK2074253192092946047>; state=16 0(2569) end of header reached, state=5 0(2569) parse_headers: Via found, flags=2 0(2569) parse_headers: this is the first via 0(2569) After parse_msg... 0(2569) preparing to run routing scripts... 0(2569) parse_headers: flags=100 0(2569) DEBUG: add_param: tag=c0a80101-b31387 0(2569) DEBUG:parse_to:end of header reached, state=29 0(2569) DBUG:parse_to: display={}, ruri={sip:101@sd-7501.dedibox.fr:5060;user=phone} 0(2569) DEBUG: get_hdr_field: <To> [66]; uri=[sip:101@sd-7501.dedibox.fr:5060;user=phone] 0(2569) DEBUG: to body [<sip:101@sd-7501.dedibox.fr:5060;user=phone>] 0(2569) get_hdr_field: cseq <CSeq>: <1> <BYE> 0(2569) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2569) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 10 0(2569) ROUTE2: STARTING NAT DETECTION 0(2569) !!!!!!!!! NAT UAC TEST 19 SUCEDEED 0(2569) parse_headers: flags=80 0(2569) DEBUG: get_hdr_body : content_length=0 0(2569) found end of header 0(2569) ROUTE2: SETFLAG 3 0(2569) DEBUG: add_param: tag=c0a80101-2376fc2 0(2569) DEBUG:parse_to:end of header reached, state=29 0(2569) DBUG:parse_to: display={}, ruri={sip:103@sd-7501.dedibox.fr:5060;user=phone} 0(2569) parse_headers: flags=200 0(2569) is_preloaded: No 0(2569) grep_sock_info - checking if host==us: 11==12 && [82.127.0.79] == [88.191.45.91] 0(2569) grep_sock_info - checking if port 5060 matches port 1312 0(2569) grep_sock_info - checking if host==us: 11==12 && [82.127.0.79] == [88.191.45.91] 0(2569) grep_sock_info - checking if port 5060 matches port 1312 0(2569) DEBUG:check_self: host != me 0(2569) grep_sock_info - checking if host==us: 12==12 && [88.191.45.91] == [88.191.45.91] 0(2569) grep_sock_info - checking if port 5060 matches port 5060 0(2569) after_loose: Topmost route URI: 'sip:88.191.45.91;lr=on;ftag=c0a80101-b31387' is me 0(2569) parse_headers: flags=200 0(2569) found end of header 0(2569) find_next_route: No next Route HF found 0(2569) after_loose: No next URI found 0(2569) DBG:rr:run_rr_callbacks: callback id 0 entered with <lr=on;ftag=c0a80101-b31387> Dan-Cristian Bogos a écrit : > That's because for INVITE your will take routing decisions but BYE u > will just proxy out. > > DanB > > On 8/2/07, Marc LEURENT <lftsy(a)free.fr> wrote: >> I've compared the INVITE and BYE method.... >> And the uri in the INVITE method is >> 0(2632) SIP Request: >> 0(2632) method: <INVITE> >> 0(2632) uri: <sip:103@sd-7501.dedibox.fr:5060> >> 0(2632) version: <SIP/2.0> >> >> whereas in the BYE method >> >> 0(2569) SIP Request: >> 0(2569) method: <BYE> >> 0(2569) uri: <sip:101@82.127.0.79:1312> >> 0(2569) version: <SIP/2.0> >> >> >> so the DEBUG:check_self: host != me >> I'm going to try without accoounting, but it should'nt change anything... >> >> Best Regards >> >> >> >> >> >>

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Re: [OpenSER-Users] how does OpenSER run on IPV6?
by Daniel-Constantin Mierla 03 Aug '07

03 Aug '07

Hello, On 08/03/07 08:41, null_neo(a)sina.com wrote: > Hi > > I am glad to recieve a mail to help me! > I have worked out today.Just force OpenSER only listening on IPV6 address. yep, maybe should be stated clear that you should force listen=ipv6_address to have openser managing ipv6 traffic. Daniel > And I use Linphone as UA ,they worked well. > HeHe,I'm very happy because I finished this job(without reward) and I > can have my holidays some days later. > Thank you. > > > > ----- Original Message ----- > From:Daniel-Constantin Mierla > To:null_neo@sina.com > Subject:Re: [OpenSER-Users] how does OpenSER run on IPV6? > Date:07-08-02 18:11:37 > > Hello, > > the "result was bad" does not give us hints to help you. > Please paste > the error messages you get. > > Cheers, > Daniel > > > On 07/31/07 04:29, null_neo(a)sina.com wrote: > > > > Dear sir: > > > > First,I am a Chinese,and my English is poor.^_^ > > > > We are testing kinds of applications on IPV6 including VOIP. And we > > use Linphone as our UA. If I don't use domain name,how can I add a > > account? On IPV4,I add account " # openserctl add test test > > test@ipv4_address" .And in the file "openser.cfg" ,I > directly write my > > SIP server ipv4_address . Can I directly replace > "ipv4_address" by > > "ipv6_address"? I has tried,and the resault is bad. > > > > I am very sorry that I am amateurish for VOIP ,because Our work is > > only to test. > > > > Thank you! > > > > Li Zhonglei > > > > > > > > ------------------------------------------------------------------- > > �� ҵ��ܵ��( > > > http://d1.sina.com.cn/sina/limeng3/mail_zhuiyu/2007/mail_zhuiyu_20070723.ht… > > > ) > > > > =================================================================== > > ע��2G��䣨 http://mail.sina.com.cn/chooseMode.html �� > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > ------------------------------------------------------------------- > �� ҵ��ܵ��( > http://d1.sina.com.cn/sina/limeng3/mail_zhuiyu/2007/mail_zhuiyu_20070723.ht… > ) > > =================================================================== > ע��2G��䣨 http://mail.sina.com.cn/chooseMode.html ��

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[OpenSER-Users] SCTP support
by Daniel-Constantin Mierla 03 Aug '07

03 Aug '07

Hello, courtesy of Glenn Marmon, Connecticut College Class of 2007, who submitted the patch, OpenSER includes now support for SCTP (Stream Control Transmission Protocol). It completes the list of the possible IP transport layers that could be used with SIP in OpenSER. You can find details about SCTP at: http://www.sctp.org/ Basically, it is a reliable transport protocol, like TCP, but provides a number of functions that are critical for telephony signaling transport (read more about the benefits: http://www.ietf.org/rfc/rfc3286.txt). If you have the chance to own a SCTP-aware SIP phone and can test, please provide us feedback on the mailing lists -- also, let us know the name and type of any SIP phone with SCTP support that you know. Hope SCTP will gain more and more space, as it seems to be the right protocol for VoIP/IM&P signaling, where UDP cannot cope very well with huge messages and TCP adds too much overhead. Cheers, Daniel

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[OpenSER-Users] Logs to /var/log/openser
by Iñaki Baz Castillo 03 Aug '07

03 Aug '07

Hi, I've configured: log_facility=LOG_LOCAL0 and added in "syslog.conf": # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/ser.log local0.* -/var/log/ser.log as it appears in http://docs.huihoo.com/openser/tutorials/ser-syslog/ar01s06.html OpenSer then logs in /var/log/openser, true, but it still logs in /var/log/syslog. ¿?¿ Obviously I don't want OpenSer logs appearing in /var/log/syslog. I use Debian Etch, any idea of what can be wrong? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] avp_subst and regex
by Zahid Mehmood 02 Aug '07

02 Aug '07

Hi, I am trying to rewrite a 5 digit number to a 11 digit number by adding a prefix "121285". To achieve that I use the following: avp_subst("$avp(s:fromuri)","/^(sip:)([134][0-9]{4}@.*)/\1+121285\2/"); in openser.log new fromuri = sip:+12128512345@host.columbia.edu It works fine because of the "+" after \1 This avp_subst does not work properly if the "+" is taken out. How can I use "\1" followed by another digit? avp_subst("$avp(s:fromuri)","/^(sip:)([134][0-9]{4}@.*)/\1121285\2/"); in openser.log: new fromuri = J128512345(a)host.columbia.edu What am i doing wrong? Thanks in advance for your help. -- Zahid

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[OpenSER-Users] OpenSER looping routes back to itself.
by Taylor Carpenter 02 Aug '07

02 Aug '07

Running OpenSER v1.1.1 with internal ip for sip client connections and ext ip to connect to sip providers. I am getting loops in my calls sometimes... They stop after reaching the max hops trying to send the BYE message to it self. I am wondering if it is the record- route that I see that includes the internal IP. The 483 error is U 2007/07/30 00:49:56.133157 192.168.18.22:5060 -> 192.168.18.22:5060 SIP/2.0 483 Too Many Hops. From: <sip:15126461516@192.168.18.1:5060>;tag=nu-481d-6c62. To: <sip:18065436468@192.168.18.22:5060>;tag=6738. Call-ID: 421ac0c6-1dd2-11b2-8093-df998124da20(a)192.168.18.91. CSeq: 11269143 BYE. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.7fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.6fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.5fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.4fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.3fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.2fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.1fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.0fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.fee832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.eee832a5.0. Via: SIP/2.0/UDP 192.168.18.91:5060;rport=5060. Server: OpenSer (1.1.1-notls (i386/linux)). Content-Length: 0. Any ideas?

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[OpenSER-Users] benchmark module
by Daniel-Constantin Mierla 02 Aug '07

02 Aug '07

Hello, starting from the original patch submitted by Bastian Friedrich and incorporating the code used in the core for past benchmarking, the devel version has a new module "benchmark". It can be used to log the duration of a configuration part execution, allowing in this way to detect slow elements of routing script. I am sure is a important tool, at least in preparing production systems, so development will continue to make more flexible the printed statistics. The readme file will give you more details: http://www.openser.org/docs/modules/1.3.x/benchmark.html The module exports an internal API, so developers can do benchmarking inside the C code. Feedback is appreciated. Cheers, Daniel

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Re: [OpenSER-Users] Replacing "." of a domain with "_" and store the result in a variable
by Iñaki Baz Castillo 02 Aug '07

02 Aug '07

El Thursday 02 August 2007 17:31:23 escribió: > Hi, > > $avp(s:dom) = $rd; > avp_subst("$avp(s:dom)", "/\./_/g"); Great, didn't realize of that :) Thanks a lot. > > Andreas > > Iñaki Baz Castillo wrote: > > Hi, I just need to take the requested domain $rd and store in a variable > > that domain but with . replaced by _: > > > > sip.domain.org -> sip_domain_org > > > > This is because I want tables in MySQL containing the domain name but > > MySQL don't allow . in table name. > > > > So now, what should I do for that? I really thought "textops" could help > > me but I don't see a function to replace text and store it in a variable. > > > > I see "replace()" and "subst()" but they seem to exist for replacing text > > in the SIP message "on the fly", that is not that I'm looking for. > > > > Any suggestion? Thanks a lot. -- ilimit... *Iñaki Baz Castillo* ibc(a)in.ilimit.es ÀREA SISTEMES 0034 937 333 375 VOLTA 1, PIS 5 08224 TERRASSA.BCN Aquest enviament és confidencial i està destinat únicament a la persona a qui s'ha enviat. Pot contenir informació privada sotmesa al secret professional, la distribució de la qual està prohibida per la legislació vigent.

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[OpenSER-Users] Replacing "." of a domain with "_" and store the result in a variable
by Iñaki Baz Castillo 02 Aug '07

02 Aug '07

Hi, I just need to take the requested domain $rd and store in a variable that domain but with . replaced by _: sip.domain.org -> sip_domain_org This is because I want tables in MySQL containing the domain name but MySQL don't allow . in table name. So now, what should I do for that? I really thought "textops" could help me but I don't see a function to replace text and store it in a variable. I see "replace()" and "subst()" but they seem to exist for replacing text in the SIP message "on the fly", that is not that I'm looking for. Any suggestion? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] OPENSERAMIN:SocketError in loginController#login
by zze-WU Yin RD-MAPS-LAN 02 Aug '07

02 Aug '07

Hello everybody, I haved installed a openseradmin. And I can have a page for login in the http://localhost:3000. But when i loggined with: login: openser password:openserrw I have got a page of error : SocketError in loginController # login Do anybody know why? Thank you very much! Yin

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[OpenSER-Users] NAT + STUN
by Marc LEURENT 02 Aug '07

02 Aug '07

Hello evrybody, I need a piece of advice if you have time.... Thanks I have some trouble with my configuration file, I'm trying to set a SIP network working through NAT using STUN. REGISTRATION is working, INVITE are working, I can set a communication with 2 working phones but sip pings an BYE messages are not forwared... Any idea?? Thanks # U 88.191.45.91:5060 -> 82.127.0.79:1312 OPTIONS sip:82.127.0.79:1312 SIP/2.0. Via: SIP/2.0/UDP 88.191.45.91:5060;branch=0. From: sip:pinger@sip.leurent.eu;tag=804466c1. To: sip:82.127.0.79:1312. Call-ID: 23ce8297-c09c8eb7-a(a)88.191.45.91. CSeq: 1 OPTIONS. Content-Length: 0. . # U 82.127.0.79:1312 -> 88.191.45.91:5060 SIP/2.0 481 CallLeg/Transaction Does Not Exist. Via: SIP/2.0/UDP 88.191.45.91:5060;branch=0. From: <sip:pinger@sip.leurent.eu>;tag=804466c1. To: <sip:82.127.0.79:1312>;tag=c0a80101-4bf1d8. Call-ID: 23ce8297-c09c8eb7-a(a)88.191.45.91. CSeq: 1 OPTIONS. Content-Length: 0. . # U 82.127.0.79:1312 -> 88.191.45.91:5060 BYE sip:103@82.127.0.79:1027 SIP/2.0. Via: SIP/2.0/UDP 82.127.0.79:1313;branch=z9hG4bK8030359792092547043. From: "101"<sip:101@sip.leurent.eu:5060;user=phone>;tag=c0a80101-4c5eed. To: <sip:103@sip.leurent.eu:5060;user=phone>;tag=c0a80101-1d0bb0d. Call-ID: 66464a0-c0a80101-0-1f(a)192.168.95.4. CSeq: 2 BYE. Max-Forwards: 70. Route: <sip:88.191.45.91:5060;lr=on;ftag=c0a80101-4c5eed>. User-Agent: THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-AF-AE. Content-Length: 0. . # # $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) children=4 #alias=sip.leurent.eu port=5060 #server_signature=yes #tos=IPTOS_LOWDELAY avp_aliases="day=i:101;time=i:102;can_uri=i:800;s_ip=i:801;billing_party=i:802;from_header=i:803;sip_proxy_ip=i:804" #;pstnuser=i:805;pstnpassword=i:806:pstnrealm=i:807" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" # Stateless Module loadmodule "tm.so" # Transaction Module loadmodule "rr.so" # Record-Route and Route Module loadmodule "maxfwd.so" # Max-Forward processor Module loadmodule "usrloc.so" # User Location Implementation Module loadmodule "registrar.so" # SIP Registrat Implementation Module (need usrloc) loadmodule "textops.so" # Text Operation Module loadmodule "mi_fifo.so" # FIFO transport layer implementation for Management Interface loadmodule "acc.so" # Accounting Module loadmodule "avpops.so" # AVP Operation Module (user preference) loadmodule "uri.so" # Generic URI operation Module loadmodule "auth.so" # Authentification Module #loadmodule "auth_db.so" # Database-backend Authentication mMdule loadmodule "auth_radius.so" # RADIUS-backend Authentication Module loadmodule "group_radius.so" # User-groups Module with RADIUS-backend #loadmodule "avp_radius.so" # RADIUS-backend for AVP loading Module #loadmodule "presence.so" # Presence server Module #loadmodule "pua.so" # Common API for presence user agent client loadmodule "options.so" # OPTIONS server replier Module loadmodule "xlog.so" # Advanced Logger Module loadmodule "nathelper.so" # NAT Traversal Helper Module #loadmodule "dispatcher.so" # Dispatcher (load-balancer) Module loadmodule "uac.so" # User Agent Client loadmodule "siptrace.so" # SipTrace module (storage of SIP requests) #loadmodule "exec.so" # Allows to start an external command from a OpenSER script # ----------------- setting module-specific parameters --------------- # -- maxfwd params -- modparam("maxfwd", "max_limit", 10) # Default is 256 | 10 in the functions # -- sl params -- #modparam("sl", "enable_stats", 1) # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- # Uncomment this if you want to use SQL database modparam("usrloc", "db_mode", 1) # Write instantaneously in the DB modparam("usrloc", "db_url", "mysql://openser:test@127.0.0.1/openser") modparam("usrloc", "timer_interval", 10) modparam("usrloc", "nat_bflag" , 3) # -- rr params -- modparam("rr", "enable_full_lr", 1) # add value to ;lr param to make some broken UAs happy # -- siptrace params -- modparam("siptrace", "db_url", "mysql://openser:test@127.0.0.1/openser") modparam("siptrace", "table", "sip_trace") # Default value "sip_trace" modparam("siptrace", "trace_on", 1) # -- registrar params -- modparam("registrar", "default_expires", 1800) modparam("registrar", "received_avp", "$avp(i:42)") # -- nathelper params -- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "sipping_bflag", 5) modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:42)") # Same Value as the registrar module modparam("nathelper", "sipping_from", "sip:pinger@sip.leurent.eu") # -- auth params -- #modparam("auth", "secret", "johndoessecretphrase") # Default is random => don't set it #modparam("auth", "nonce_expire", 300) # Time before nounce expiration modparam("auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "use_domain", 1) # username@domain will be used for lookup # -- avp_radius parameter -- #modparam("avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- acc params (with radius )-- modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "early_media", 1) modparam("acc", "report_cancels", 1) #modparam("acc", "report_ack", 0) modparam("acc", "detect_direction", 1) #modparam("acc", "log_flag", 1) # number of the flag which will be used to mark messages for accounting #modparam("acc", "log_level", 1) # Set the reporting log level #modparam("acc", "log_missed_flag", 2) # #modparam("acc", "failed_transaction_flag", 2) modparam("acc", "service_type", 15) # Radius service type used for accounting : 15 = (SIP) #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") # ATTENTION: DO NOT PUT ; at the end of the radius_extra attribute modparam("acc", "radius_extra", "Sip-Src-IP=$si; Sip-Src-Port=$sp; Canonical-URI=$avp(can_uri); Billing-Party=$avp(billing_party); SIP-Proxy-IP=$avp(sip_proxy_ip); User-Agent=$ua ") #Billing-Party=$avp(billing_party) #From-Header=$hdr(from); #User-Name=$fU; #From-Header=$avp(from_header); #Digest-Realm=$fd #Sip-From-Tag=$avp(from_header); #SIP-Method=$rm; # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # NAT detection route(2); # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") { record_route(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); } route(1); }; # Set the acc flags 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) = $fu; # From URI $avp(from_header) = $fU; # From URI username $avp(sip_proxy_ip) = $Ri; # Received IP address setflag(1); # radius_flag setflag(2); # radius_missed_flag }; # Functions when calling other domains if (!uri==myself) { # check if user is allowed to do voip calls to other domains # if(is_method("INVITE|MESSAGE")) { # if (radius_is_user_in("From", "voip")) { # sl_send_reply("403", "Forbidden VoIP"); # exit; # }; # }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { sip_trace(); xlog("L_INFO", "$fU IS TRYING TO REGISTER \r\n"); if (!radius_www_authorize("sip.leurent.eu")) { www_challenge("sip.leurent.eu", "0"); # qop set to 1 xlog("L_INFO", "WWW_CHALLENGE of $si FAILED \r\n"); exit; }; #if (isflagset(5)) { if (isbflagset(3)) { #setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); # Deprecated setbflag(5); # Set Flag for SIP PINGING }; save("location"); xlog("L_INFO", "SAVE LOCATION OF $si \r\n"); exit; }; if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); xlog("L_DBG", "ACC RADIUS: 404 NOT FOUND FOR $si \r\n"); sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } ## Generic Forward route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ #setflag(6); # Deprecated, for version 1.1 setbflag(3); }; #if (isflagset(5)||isflagset(6)) { if (isbflagset(3)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } # NAT Detection route[2]{ force_rport(); if (nat_uac_test("19")) { xlog("!!!!!!!!! NAT UAC TEST 19 SUCEDEED \r\n"); if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; #setflag(5); Deprecated setbflag(3); }; } ## Route for natted contact route[3] { if (is_method("BYE|CANCEL")) { # Ajout Maison acc_rad_request("200 ok"); acc_log_request("200 ok"); #unforce_rtp_proxy(); } else if (is_method("INVITE")){ #force_rtp_proxy(); t_on_failure("1"); }; #if (isflagset(5)) if (isbflagset(3)){ search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } t_on_reply("1"); } ## Failure Route 1 failure_route[1] { xlog("!!!!!!!!! ON FAILURE ROUTE \r\n"); #if (isflagset(6) || isflagset(5)) { if (isbflagset(3)) { #unforce_rtp_proxy(); } } ## Reply route onreply_route[1] { xlog("!!!!!!!!! ON REPLY ROUTE \r\n"); #if ((isflagset(5) || isflagset(6)) && status=~"(183)|(2[0-9][0-9])") { if (isbflagset(3) && status=~"(183)|(2[0-9][0-9])") { #force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); #if (isflagset(6)) { if (isbflagset(3)) { xlog("!!!!!!!!! ON REPLY ROUTE / FIX NATED CONTACT \r\n"); fix_nated_contact(); } exit; }

3 7

[OpenSER-Users] openser to solve SIP+RTP+NAT problems in typical SOHO setup
by Jens Thiele 02 Aug '07

02 Aug '07

Hi, I am evaluating software to solve SIP+RTP+NAT problems in typical SOHO setups: Scenarios: Typical scenarios are (A)DSL "dial-up / dynamic IP" connections with one public IP and a router doing NAT for some LANs. 1) WAN multiple LANs internet <-> NAT router <-> SIP phones sip registrar SIP proxy <-> SIP phones with NAT logic : 2) WAN multiple LANs internet <-> NAT router <-> SIP proxy with NAT logic, SIP phones sip registrar <-> SIP phones : Notes: - the SIP phones use the SIP proxy with NAT logic as outbound proxy - the proxy is notified if the public IP changes The closest match so far has been siproxd [1]. But I would prefer to use openser+nathelper+rtpproxy. Now it seems nathelper is designed for a different scenario (SIP provider): WAN/internet sip registrar <-> ser+nathelper <-> NAT router <-> multiple LANs with SIP phones and uses the packet source address in fix_nated_contact. Now I stumbled across milkfish [2] and its patches [3] to openser (nathelper). It seems the patch allows to explicitly set the address used by fix_nated_contact to rewrite the contact header (instead of using the packet source address). Searching the mailing list archives I did not find a better solution. Closest match I found was the thread: "[Devel] Extended nathelper" started by Walter Schober http://www.mail-archive.com/devel@openser.org/msg01930.html Ending in the suggestion to use add_rvc_param(). But add_rcv_param seems to use the packet source address, too. But also from this thread it seems one might use the mangle module to rewrite the contact header field, but then I am not sure about the state of this module: "This is a module to help with SDP mangling. Still in testing." I am a bit surprised that there is no straight forward solution to such a common setup. Am I missing something really obvious? Thanks Greetings Jens Thiele [1] siproxd <http://siproxd.sourceforge.net/> [2] milkfish <http://wiki.milkfish.org> [3] milkfish openser patches <http://packages.milkfish.org/boozy/Milkfish_Sources_for_OpenWrt-SDK/OpenWrt…>

1 0

[OpenSER-Users] openser+radius
by ram 02 Aug '07

02 Aug '07

Hi iam installing openser SVN and Freeradius 1.16 followed the link as mentioned in document when i run radiusd -X iam getting below problem 1. rlm_sql (sql): Connected new DB handle, #4 2. Module: Instantiated sql (sql) 3. Module: Loaded Acct-Unique-Session-Id 4. acct_unique: key = "User-Name, Acct-Session-Id, NAS-IP-Address, Client-IP-Address, NAS-Port" 5. rlm_acct_unique: Cannot find attribute 'NAS-IP-Address' in dictionary 6. radiusd.conf[1159]: acct_unique: Module instantiation failed. 7. radiusd.conf[1753] Unknown module "acct_unique". 8. radiusd.conf[1747] Failed to parse preacct section. any suggestiong ram

2 1

[OpenSER-Users] PRACK handling
by Zahid Mehmood 02 Aug '07

02 Aug '07

Hello everyone, What is the preferred way of handling PRACKS? simply relay them if they are in loose_route? should they be authenticated? What, if any, are the potential risks of not authenticating PRACKS? Thanks in advance. -- Zahid

2 1

[OpenSER-Users] disable media/nathelper
by Aymeric Moizard 02 Aug '07

02 Aug '07

Hi, I'm facing an issue with openser. I checked the latest SVN code and it seems my issue is not solved even in the svn version. My UA is sending an SDP with 2 m lines: one for audio and one for video. In a re-INVITE, I want to disable video and I'm sending m=video 0 RTP/AVP.. My openser is currently modifying the port number for the video line even if it's 0 which means it's disabled... This behavior would of course happen even for audio lines that ones wants to disable. I was not able to find a clean fix in nathelper.c and hope to get some help for proposing one... I guess it would be something close to: if (oldport==0) skip altering ip/port for media and continue; but I don't want to insert any mem leak or lose a session... tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

2 1

[OpenSER-Users] how does OpenSER run on IPV6?
by null_neo＠sina.com 02 Aug '07

02 Aug '07

Dear sir: First,I am a Chinese,and my English is poor.^_^ We are testing kinds of applications on IPV6 including VOIP. And we use Linphone as our UA. If  I don't use domain name,how can I add a account? On IPV4,I add account " # openserctl add test test test@ipv4_address" .And in the file "openser.cfg" ,I directly write my SIP server ipv4_address . Can I directly replace "ipv4_address" by "ipv6_address"?  I has tried,and the resault is bad. I am very sorry that I am amateurish for VOIP ,because Our work is only to test. Thank you!                                                                                                 Li  Zhonglei ------------------------------------------------------------------- 吉贝克 ——商业智能的领军者( http://d1.sina.com.cn/sina/limeng3/mail_zhuiyu/2007/mail_zhuiyu_20070723.ht… ) =================================================================== 注册新浪2G免费邮箱（ http://mail.sina.com.cn/chooseMode.html ）

2 1

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