sr-users October 2005

sr-users@lists.kamailio.org

214 participants
458 discussions

[Users] version 1.0.0 released
by Daniel-Constantin Mierla 31 Oct '05

31 Oct '05

Hi everybody, the version 1.0.0 has been released. The branch 'rel_1_0_0' is created and must be used by anybody wanting the get v1.0.0 out from CVS (cvs co -r rel_1_0_0 sip-server). There are a lot of new features in this release, starting from TLS, pseudo-variables in core and continuing with enhancements in avpops and other module. This release is the first open source sip server with TLS support, although there is still to work on it (e.g., multi domain support) it allow secure sip signaling communications between servers and clients. An unified format of pseudo-variables has been adopted in core and many modules, basic functionality for future dialog support is present in RR module, avpops includes functions for string operations (concatenation, prefix, suffix, substitute), arithmetic and bitwise operations, among others. A new module, uac_redirect, has been introduced to handle redirect replies (3xx) on server, with proper accounting and filter rules. See the full list of new features at: http://www.openser.org/release-1.0.0.php A roughly change log can be browsed online at: http://cvs.sourceforge.net/viewcvs.py/openser/sip-server/ChangeLog?view=mar… There are two categories of released packages: - classic openser, without TLS support - the name of the package does not contain the -tls- word - TLS-ed version of openser - the name of the package contain the -tls- word The tarball with the sources can be downloaded from: http://openser.org/pub/openser/1.0.0/src/ The basic binary packages are deployed to the download site: http://openser.org/pub/openser/1.0.0/bin/ http://openser.org/pub/openser/1.0.0/packages/ We are still collecting other packages from contributors which already volunteered, so expect new distributions to be represented there in the next days. The module documentations are up-to-date, the site and tutorials need a quick review in the next days.

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[Serusers] Too many hops
by Philip Jones 31 Oct '05

31 Oct '05

I am using a dedicated server to host SER, when I use the actual IP address of server i/f I connect no problem but, using the domain name, the client times out with "Too many hops". Is there a known solution to this? I would really prefer to use the domain name. Phil

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[Serusers] REFER with Replaces header via FIFO - Invite drops Replaces
by Brian Roy 31 Oct '05

31 Oct '05

Short and sweet: I am attempting to REFER and existing call to another UA using a REFER with an imbedded REPLACES header in the REFER-TO header. FIFO accepts the REFER and the call is set up, however the REPLACES header is dropped between the REFER and the INVITE. 1) Has anyone been able to get a REFER with REPLACES to work with SER? a. Am I doing something completely friggin' stupid?? 2) Is this a SER bug. or a FIFO bug. or neither? See Below: Initial Call: *************************You'll note that replace is supported*************************************** INVITE sip:1999206@maggie-may.home.brianandkelly.ws:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bK5d9b3bcbFDFC6408 From: "Brian Dev" <sip:briandev@maggie-may.home.brianandkelly.ws:5060>;tag=2A63079D-99C40A 74 To: <sip:1999206@maggie-may.home.brianandkelly.ws:5060> CSeq: 1 INVITE Call-ID: 74e15821-cc9958e7-2e639266(a)192.168.0.161 Contact: <sip:briandev@192.168.0.161;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1130777573 1130777573 IN IP4 192.168.0.161 s=Polycom IP Phone c=IN IP4 192.168.0.161 t=0 0 m=audio 2232 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 ACK sip:206@192.168.0.19 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161;branch=z9hG4bKeb569c391EDD1A0 From: "Brian Dev" <sip:briandev@maggie-may.home.brianandkelly.ws:5060>;tag=2A63079D-99C40A 74 To: <sip:1999206@maggie-may.home.brianandkelly.ws:5060>;tag=as1624b103 Route: <sip:192.168.0.24;ftag=2A63079D-99C40A74;lr=on> CSeq: 1 ACK Call-ID: 74e15821-cc9958e7-2e639266(a)192.168.0.161 Contact: <sip:briandev@192.168.0.161;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.0 Max-Forwards: 70 Content-Length: 0 Attempted REFER: ***************************************Note that that Replaces info drops in the INVITE*********************************************************** REFER sip:kellydev@maggie-may.home.brianandkelly.ws SIP/2.0 Via: SIP/2.0/UDP 192.168.0.24;branch=z9hG4bK3915.d2aefa37.0 To: sip:briandev@maggie-may.home.brianandkelly.ws From: sip:kellydev@maggie-may.home.brianandkelly.ws;tag=1f850894c564e190ae8e24 413352beae-4b42 CSeq: 10 REFER Call-ID: 2bbbc774-0(a)192.168.0.24 Content-Length: 0 User-Agent: Sip EXpress router(0.9.4 (i386/linux)) Refer-To: sip:briandev@maggie-may.home.brianandkelly.ws?Replaces=74e15821-cc9958e7 -2e639266(a)192.168.0.161;to-tag=as1624b103;from-tag=2A63079D-99C40A74 INVITE sip:briandev@maggie-may.home.brianandkelly.ws SIP/2.0 Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK880b0a28917F93C5 From: "Kelly Roy - DEV" <sip:kellydev@maggie-may.home.brianandkelly.ws:5060>;tag=519BC0C2-649892 E1 To: <sip:briandev@maggie-may.home.brianandkelly.ws> CSeq: 1 INVITE Call-ID: 801c0ab6-44e87514-ba253cb(a)192.168.0.160 Contact: <sip:kellydev@192.168.0.160:5060;transport=udp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1130775178 1130775178 IN IP4 192.168.0.160 s=Polycom IP Phone c=IN IP4 192.168.0.160 t=0 0 m=audio 2230 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 Brian Roy email: b.t.roy(a)brianandkelly.ws IM - jabber: briantroy(a)brianandkelly.dyndns.org Phone - Gizmo Project: briantroy (1-747-623-4685) Brian Roy email: b.t.roy(a)brianandkelly.ws IM - jabber: briantroy(a)brianandkelly.dyndns.org Phone - Gizmo Project: briantroy (1-747-623-4685)

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[Serusers] REFER with Replaces header via FIFO - Invite drops Replaces
by Brian Roy 31 Oct '05

31 Oct '05

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[Serusers] Windows Messenger 5.1 On-line
by gorand＠dvvti.com 31 Oct '05

31 Oct '05

I am testing Windows Messenger 5.l and it works fine except when I connect other users that are on the SER box, it does not show them as on-line. It displays them as off-line. I am able to make sip calls through the SER Box. Everything works fine in that aspect. I want to use the Windows Messenger as a communication aspect to be able to send instant messages to the users. Please let me know if this possible with this Windows Messenger? Thanks.

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[Users] How to remove fields from SQL accounting?
by Federico Giannici 31 Oct '05

31 Oct '05

I need to customize the data logged with the SQL accounting. The "db_extra" parameter let me ADD new fields but it seems that there is no way to REMOVE fields from the sql command. Infact, the "log_fmt" seems to affect only the syslog accounting. Is there a way to prevent a field to be included into the SQL INSERT command? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Serusers] Handling of REFER messages
by Miroslaw Nawrot 31 Oct '05

31 Oct '05

Hi, I am trying to do a peculiar task and I want to find out if SER can do it. The issue is with call transfers which generate the REFER message. I am connecting a Nortel voip switch to Asterisk via sip. I need to have all call transfers generated by the Nortel switch to not reach Asterisk as all the call transfers are done to dids/extensions not found on the Asterisk PBX. Instead I would like to intercept / re-route them back to my Nortel switch which can handle it. Currently I have a workaround using an Audiocodes gateway with loopback cables. The Audiocodes when sees the transfer request sends it back to the Nortel switch. As this is a wasteful use of expensive hardware I would like to find another method to do this. Can someone let me know if this is possible with SER? And if it is which functions do I use? Any help is greatly appreciated. Mirek The Future of Voice Globalive Communications Corp. T: 416-640-1088 F: 416-640-1089 60 Adelaide St. E, 6th Floor Toronto, ON, M5C 3E4 www.globalive.com

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[Serusers] Error with acc.so and db funtions
by Antonio Membrives 31 Oct '05

31 Oct '05

Hi again i have this lines in ser.cfg modparam("acc", "log_flag", 1) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "miocfsp") for store missing calls in serweb, but when I launch the ser services received 3 errors and see this in /var/log/messages Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_flag> not found in module <acc> Oct 31 13:50:00 dns2 ser: parse error (73,19-20): Can't set module parameter Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_missed_flag> not found in module <acc> Oct 31 13:50:00 dns2 ser: parse error (74,19-20): Can't set module parameter Oct 31 13:50:00 dns2 ser: set_mod_param_regex: parameter <db_url> not found in module <acc> Oct 31 13:50:00 dns2 ser: parse error (76,20-21): Can't set module parameter I have fedora 3+ser 0.9.3 , any idea of this. Thanks. Antonio Membrives

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[Serusers] Error with msilo, dont work
by Antonio Membrives 31 Oct '05

31 Oct '05

Hi to all, I try to use MSILO in Fedora 3 + Ser 0.9.3 and received the next message in /var/log/messages /var/log/tail messages Oct 31 09:54:15 dns2 ser: WARNING: fix_socket_list: could not rev. resolve x.x.x.x Oct 31 09:54:15 dns2 ser[8823]: Maxfwd module- initializing Oct 31 09:54:15 dns2 ser[8823]: bind_dbmod: Module mysql//ser does not export db_use_table function Oct 31 09:54:15 dns2 ser[8823]: init_mod(): Error while initializing module msilo And ser not work, if I coment the line of msilo db_url work again but not work msilo. Any idea? Antonio Membrives. #ser.cfg debug=3 fork=yes log_stderror=no listen=x.x.x.x # INSERT YOUR IP ADDRESS HERE alias=dominio.es alias=x.x.x.x port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("msilo", "registrar", "sip:registar@telintcable.es") modparam("msilo", "db_url", "mysql//ser:heslo@localhost/msilo") #modparam("acc", "log_level", 1) #modparam("acc", "log_flag", 2) #modparam("acc", "log_missed_flag", 3) #modparam("acc", "log_fmt", "fimos") #modparam("acc.", "db_url","mysql://ser:heslo@localhost/ser") #modparam("acc", "db_flag", 2) #modparam("acc", "db_missed_flag", 3) modparam("tm","fr_inv_timer",30) modparam("tm","fr_timer",3) /var/log/tail messages Oct 31 09:54:15 dns2 ser: WARNING: fix_socket_list: could not rev. resolve x.x.x.x Oct 31 09:54:15 dns2 ser[8823]: Maxfwd module- initializing Oct 31 09:54:15 dns2 ser[8823]: bind_dbmod: Module mysql//ser does not export db_use_table function Oct 31 09:54:15 dns2 ser[8823]: init_mod(): Error while initializing module msilo

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[Serusers] where to find serweb tool
by dave cra 31 Oct '05

31 Oct '05

Hi all, I've tried to get the serweb tool off the iptel website and it can not find it. Is there another place to obtain this? I'm running windows 2000. Cheers Dave --------------------------------- Yahoo! FareChase - Search multiple travel sites in one click.

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sr-users October 2005