- sr-users - kamailio.org

[Serusers] SER pre-release - Presence and NOTIFY
by SER LIST 14 Nov '06

14 Nov '06

Hi, I have SER Proxy and PRESENCE running on different machines. REGISTERS are being forwarded by SER Proxy to PRESENCE and PRESENCE is storing the location information in "location" db. What I am seeing is that when the PRESENCE Server sends NOTIFYs to watchers, it sends them to the SER PROXY (destination IP Is that of SER PROXY) but the URI in NOTIFY is that of the watchers LOCATION INFO. Is this correct? I would like the NOTIFY URI to contain the SIP DOMAIN and not the contact info. … [View More]

3 3

[Serusers] Presence Client
by Ajay Kolambekar 14 Nov '06

14 Nov '06

Hi, To test the presence capability of SER which presence client application I can use? Regards, Ajay DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Pvt. Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this … [View More]

2 1

Re: [Serusers] Any resolution to the presence core dump?
by SER LIST 14 Nov '06

14 Nov '06

Yes. It seems to be fixed in the CVS version. I am running the CVS version since four days and I have not had a core dump. I am testing further and will post any issues I find. Thanks >From: Vaclav Kubart <vaclav.kubart(a)iptel.org> >To: SER LIST <sergrp(a)hotmail.com> >CC: serusers(a)iptel.org >Subject: Re: [Serusers] Any resolution to the presence core dump? >Date: Mon, 13 Nov 2006 14:25:16 +0100 > >It could be corrected in CVS version, but it is not sure. … [View More]

2 1

[Serusers] Problem with SER modules, mediaproxy.so and domain.so
by jk 14 Nov '06

14 Nov '06

Hello, I am totally confused, Installed ser rpm on FC5, with all the modules, If I start ser with out mediaproxy.so and domain.so modules every thing works great. But When I add these two mosules I start getting in problem and it does not start. Here is the error, 1(0) DEBUG: init_mod_child (-1): tm 1(0) DEBUG: callid: '74fe3fe7-0(a)xxx.xxx.xxx.xxx' 1(0) DEBUG: init_mod_child (-1): usrloc 1(0) get_connection(): Connection not found in the pool 2(0) INFO: fifo process starting: 11768 2(… [View More]0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '74fe3fe7-0(a)xxx.xxx.xxx.xxx' 2(0) DEBUG: init_mod_child (-2): usrloc 2(0) get_connection(): Connection not found in the pool 0(11766) DEBUG: init_mod_child (1): tm 0(11766) DEBUG: callid: '74fe3fe7-11766(a)xxx.xxx.xxx.xxx' 0(11766) DEBUG: init_mod_child (1): usrloc 0(11766) get_connection(): Connection not found in the pool 1(11767) DEBUG: init_mod_child (-1): auth_db 0(11766) DEBUG: init_mod_child (1): auth_db 1(11767) get_connection(): Connection found in the pool 0(11766) get_connection(): Connection found in the pool 1(11767) DEBUG: init_mod_child (-1): uri_db 0(11766) DEBUG: init_mod_child (1): uri_db 1(11767) get_connection(): Connection found in the pool 0(11766) get_connection(): Connection found in the pool 1(11767) DEBUG: init_mod_child (-1): domain 0(11766) 1(11767) DEBUG: init_mod_child (1): domain DEBUG: init_mod_child (-1): nathelper 0(11766) get_connection(): Connection not found in the pool 1(11767) DEBUG: init_mod_child (-1): avpops 1(11767) get_connection(): Connection found in the pool 1(11767) DEBUG: init_mod_child (-1): permissions 2(11768) DEBUG: init_mod_child (-2): auth_db 2(11768) get_connection(): Connection found in the pool 2(11768) DEBUG: init_mod_child (-2): uri_db 2(11768) get_connection(): Connection found in the pool 2(11768) DEBUG: init_mod_child (-2): domain 2(11768) DEBUG: init_mod_child (-2): nathelper 2(11768) DEBUG: init_mod_child (-2): avpops 2(11768) get_connection(): Connection found in the pool 2(11768) DEBUG: init_mod_child (-2): permissions 2(11768) get_connection(): Connection found in the pool * 0(11766) new_connection(): Access denied for user 'serro'@'localhost' (using password: YES) I have not idea why it is using serror@localhost to connect to db. I will post my config files (ser.cfg and serctl) below. * 0(11766) db_init(): Could not create a connection * 0(11766) ERROR:domain_db_init: cannot initialize database connection 0(11766) ERROR: domain:child_init(): Unable to connect to the database 0(11766) init_mod_child(): Error while initializing module domain 0(11766) main_dontfork: init_child failed 2(11768) INFO: signal 15 received 2(11768) Memory status (pkg):* 2(11768) 1(11767) INFO: signal 15 received 1(11767) Memory status (pkg): 1(11767) fm_status (0x811c800): 1(11767) heap size= 1048576 Top part of ser.cfg file [root@ ser]# more ser.cfg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ #listen=192.168.2.201 listen=209.XXX.XXX.XXX check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" a fifo_db_url="mysql://username:password@host/ser_db" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/avpops.so" loadmodule "/usr/lib/ser/modules/permissions.so" #loadmodule "/usr/lib/ser/modules/domain.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://username:password@host/ser_db") modparam("auth_db", "calculate_ha1", 0) 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","/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://username:password@host/ser_db") modparam("avpops", "avp_table", "usr_preferences") # main routing logic route { SERCTL file # configuration for starting/stopping ser PID_FILE=/var/run/ser.pid SYSLOG=1 # 0=output to console, 1=output to syslog STARTOPTIONS= # for example -dddd DIR=`dirname $0` SERBIN=$DIR/ser AWK=awk # ser's FIFO server if [ -z "$SER_FIFO" ]; then SER_FIFO=/tmp/ser_fifo fi # period in which stats are reprinted if [ -z "$WATCH_PERIOD" ] ; then WATCH_PERIOD=2 fi # SQL config if [ -z "$SQL_DB" ] ; then SQL_DB=ser_DB fi if [ -z "$SQL_HOST" ] ; then SQL_HOST=SER_HOST //not localhost fi if [ -z "$SQL_USER" ] ; then SQL_USER=username fi # the read-only user for whom password may be stored here if [ -z "$RO_USER" ] ; then RO_USER=usename fi if [ -z "$RO_PW" ] ; then RO_PW=password fi # binaries GENHA1='gen_ha1' MYSQL='mysql' SER='sr' LAST_LINE='tail -1' # ACL name verification VERIFY_ACL=1 ACL_GROUPS="local ld int voicemail free-pstn prepaid" VERSION='$Revision: 1.83.2.2 $' NOTE: If I dont add mediaproxy.so and domain.so then everything works lgreate Ser verision [root@ ser]# rpm -qa | grep ser- ser-serweb-0.9.6-6.fc5 ///This also did not work, problem with classkit and runkit extension. If any one could make it work on FC5 with PHP 5.1, Please share. ser-mysql-0.9.6-6.fc5 ser-0.9.6-6.fc5 I am tired like dead, Any idea will be help. Thank you, -JK [View Less]

1 0

[Serusers] Please help with radius-ser mysql setup
by Lokesh Kumar 13 Nov '06

13 Nov '06

Hello List, I had setup ser-0.9.6 and freeradius-1.1.3. I am able to authenticate sip users via radius using entries in mysql database, but unable to do group membership checking with mysql database, but if I use users text file for group checking it works fine. The mentioned below link have the logs and entries in mysql database. HYPERLINK "http://pastebin.ca/247613"http://pastebin.ca/247613 As Mr. Greger suggested in previous post not to challenge options message (second … [View More]

1 0

[Serusers] Help needed with radius-mysql and group authorization checking
by Lokesh Kumar 13 Nov '06

13 Nov '06

Hello List, I have problem with the group membership checking of ser calls with radius using mysql module. Well group checking is working fine with the users file but when I try to do that checking with mysql entries it always getting failed. Below mentioned is section of my users file 211069020(a)mydomain.pt Sip-Group == "int", Auth-Type := Accept Reply-Message = "Authorized" What should be the entries corresponding to the above mentioned in mysql tables of … [View More]

2 1

[Users] The Summit picx are So Good,
by raviprakash sunkara 13 Nov '06

13 Nov '06

Hello Users , Thanks to *Bogdan-Andrei Iancu* *Daniel-Constantin Mierla* *Dmitry Isakbayev* *Elena-Ramona Modroiu* *Glenn Dalgliesh* *Klaus Darilion* *Mike Williams* *Norman Brandinger I'M very glad to see in U in Summit pics on Nov 7-9 , * -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

3 2

[Serusers] SER IPV6
by ali daher 13 Nov '06

13 Nov '06

Hi all I am new with Ser i installed Ser 0.9.3 on a linux suse 10.1 machine i have two clients running supporting ipv6 . the 2 clients and the linux are connected through a switch. i want to send SIP messages between clients over IPV6. i can not register clients on SER using the IPV6. i was told to change the ser.cfg file daoes anyone have idea how to make ser work on ipv6, and support SIP messages on IPV6? thanks in advance for your help regards … [View More]

2 1

[Users] OpenSER Summit: photos and slides
by Daniel-Constantin Mierla 13 Nov '06

13 Nov '06

Hello, first photo album from the OpenSER summit days is available at: http://www.openser.org/events/2006-OpenSER-Summit/photos/Nov.7-8/ Also, the slides are collected in their last version and uploaded to: http://www.openser.org/events/2006-OpenSER-Summit/slides/ We like to thank to all people attending the event, the number of participants was a pleasant surprise even for VoN organizers. During Wednesday we were able to meet lot of community members and listen to very interesting … [View More]

1 1

[Serusers] Pseudo-variables
by Benito Caracuel 13 Nov '06

13 Nov '06

Hello All, I would like to do: exec_msg("cat /tmp/test.txt > %ci"); But %ci has always the same value!! How Can I change the value of a pseudo-variable? Thank you. Ben, _________________________________________________________________ Prueba algunos de los nuevos servicios en línea que te ofrece Windows Live Ideas: tan nuevos que ni siquiera se han publicado oficialmente todavía. http://ideas.live.com

1 0

[Serusers] (no subject)
by ali daher 13 Nov '06

13 Nov '06

Hi I want to post to the list my email is "daher_ali(a)hotmail.com" regards _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[Serusers] added sipgrep utility
by Vaclav Kubart 13 Nov '06

13 Nov '06

There was added sipgrep utility written by Alexandr Dubovikov <shurik(a)start4.info> into SER's source tree (utils/sipgrep). It is a wrapper on ngrep which * filters SIP messages according number in To or From header fields * displays From tag, To tag, Call-ID and branch in different colours, thus it is possible to trace dialogs or transactions by "one look into message" ... this is really great! ;-) * It can store received messages into a file and show them (together) For a … [View More]

1 0

[Serusers] Answering with music?
by Evert 13 Nov '06

13 Nov '06

Hi all! For testing purposes I want certain extensions being answered with streamed music. I guess this requires certain routing in SER (and perhaps a SIP client?) Has anyone done something similar before? Where/how should I start? Regards, Evert

3 3

[Serusers] Any resolution to the presence core dump?
by SER LIST 13 Nov '06

13 Nov '06

I was wondering whether a patch or a new release is available to the issue of presence core dumping on a regular basis that was reported short while back? Thanks in advance. >The version 0.10.99 of presence snapshot core dumps on a regular basis after running for a certain time. Doing a gdb shows that the core dump occurs at msg_queue_destroy: >#0 0x001b87f2 in msg_queue_destroy (q=0xb6293e98) > at /usr/local/src/ser-0.10.99-dev35-pa-4.1/lib/../fastlock.h:147 >147 … [View More]

3 2

[Serusers] XCAP server
by Jayantheesh Srimushnam - TLS, Chennai 13 Nov '06

13 Nov '06

Hi All, I have downloaded the SER and trying to run XCAP server. I have gone through the README file for the XCAP server and understood that it is server simulation. Still I didn't get how to install and run it. Could anyone please through some light on this? Thanks, Jayantheesh DISCLAIMER The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. It shall not attach any liability on the originator or HCL or its affiliates. … [View More]

2 1

[Serusers] Registering SIP UA from the internet
by "tzieleniewski" 13 Nov '06

13 Nov '06

Hi All!! According to the SER -Getting Started Manual SER is located in the local network and has an local IP adres. Is it some how possible to register the SIP UA from the outside of the local net?? I mean can I register them without giving SER an public IP adress?? If yes then what do I have to specify (what adress) in the softphones configuration in the SIP Proxy section?? Thanks 4 any help! Bests

2 2

[Serusers] Change SDP
by Benito Caracuel 13 Nov '06

13 Nov '06

Hello All, Basic Concept: I have to change the Path MSRP into SDP field in an INVITE and relay the new INVITE. SDP EXAMPLE: v=0 o=686873200 3372497915 3372497916 IN IP4 80.27.211.205 s=SipSession c=IN IP4 80.27.211.205 t=0 0 m=message 49152 TCP/MSRP * a=direction:active a=accept-types:text/plain a=path:msrp://80.27.211.205:49152/63331682315468750;tcp When I receive the SDP, I have to change "msrp://80.27.211.205:49152", so I execute an script and I obtain the new SDP into file.txt. Now,… [View More]

1 0

[Serusers] Presence - PA AND RLS
by SER LIST 13 Nov '06

13 Nov '06

Hi, I am not sure whether PA module access the XCAP server or not? I have users presence rules configured on XCAP server, but all my clients are getting a NOTIFY w/ pending status. What function call in PA accesses the XCAP server? It seems to me that the RLS accesses the XCAP. Thanks in advance _________________________________________________________________ Buy, Load, Play. The new Sympatico / MSN Music Store works seamlessly with Windows Media Player. Just Click PLAY. http://musicstore.sympatico.msn.ca/content/viewer.aspx?cid=SMS_Sept192006

2 1

[Serusers] Presence / Eyebeam Question
by SER LIST 13 Nov '06

13 Nov '06

I have the following issue: I have two clients (eyebeam softphones). If I modparam(pa, auth, none) then I can see the status of each other. But when modparam(pa, auth, xcap) I cannot see each other's status. I do get a POPUP window on each softphone asking me whether I want to allow the other party to see my status. Even after I say "Allow", I still can't see the status. I am able to write the presence-rules to XCAP server from the eyebeam client. Eyebeam writes the file presrules.xml … [View More]

2 1

[Users] exec module : why not in onreply_route ?
by Jean-François SMIGIELSKI 13 Nov '06

13 Nov '06

Hello, On 11/13/06 Daniel-Constantin Mierla wrote: [...] > basically should work in onreply route as well, I haven't tried it. You > will not have access to transaction's avps, otherwise you can play and > see if you encounter any problem. Just edit the modules/exec/exec_mod.c, > add ONREPLY_ROUTE to exported functions' flags, recompile and reinstall. > If everything is ok, then we can include it in the trunk. [...] That is what I did last week and till now, I did not *yet* … [View More]

1 0

[Users] update documentation
by Klaus Darilion 13 Nov '06

13 Nov '06

Hi Daniel,Bogdan! Please update the 1.2.x documentation (http://openser.org/docs/modules/1.2.x/): - add domainpolicy module - add new ENUM module features thanks Klaus -- Klaus Darilion nic.at

2 1

[Users] Errors whie doing openser integration with radius server
by Sanjeev Manoli 13 Nov '06

13 Nov '06

Hi, I am getting following errors as seen in /var/log/message while running openser with radius integration, ***************************************************************************************** Nov 6 20:33:45 lx-dev monit[13565]: 'openser' start: /etc/init.d/openser Nov 6 20:33:45 lx-dev monit[13565]: 'openser' failed to start Nov 6 20:33:45 lx-dev openser: init_tcp: using epoll_lt as the io watch method (auto detected) Nov 6 20:33:45 lx-dev openser: INFO: statistics manager … [View More]

2 1

[Users] which is better: avp_db_load with scheme or avp_db_query?
by Alan Crosswell 13 Nov '06

13 Nov '06

Hi, I want to pull the First+Last names of the subscriber for the $ru so I can stick an RPID in a 180 or 183 response (Polycom phones display the called party name when you do this). Is it better practice to define two schemes and use avp_db_load to get the first and last name columns out of subscriber table or to use avp_db_query and do it in one shot? Thanks. /a

2 1

[Serusers] Asterisk + SIP integration for SIPLE. Newb needs advice.
by Ryan Churches 13 Nov '06

13 Nov '06

Ive administrate several Asterisk servers, and sometimes one company will have multiple servers with their own respective users. Unfortunately users of one server are unable to see the presence state of a user on another server. I understand that SER can (a) be my presence agent or (b) can pass SUBSCRIBErs to a respective asterisk box, but doesnt that result in (a) clients subscribing to SER cannot see the presence states of local channels on asterisk (i.e. a parking spot), or (b) clients not … [View More]

1 0

[Users] CANCEL/ACK processing
by Mark Kent 13 Nov '06

13 Nov '06

Observant readers will recognize from the following that this question is related to earlier posts from me... that is: there is something fundamental that I just don't get. If someone is eager to answer all my questions, then please search ahead for the question marks (?). Also, what is the bigger picture that I am missing? Please use as many words as necessary to explain. Thanks, -mark The players: CUSTOMER-IP: the UserAgent OPENSER-IP: my openser box OPENSER.FQDN: the fully … [View More]qualified domain name of my openser box PSTNGW-IP: the PSTN gateway My openser box gets this CANCEL: U 2006/11/06 18:40:10.689785 CUSTOMER-IP:5060 -> OPENSER-IP:5060 CANCEL sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> Call-ID: CALLID@CUSTOMER-IP CSeq: 2 CANCEL Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Max-Forwards: 70 Supported: 100rel,replaces Content-Length: 0 It ends up in my standard stateful relay handler where I t_relay() it (should I?) and this gets generated: # U 2006/11/06 18:40:10.691301 OPENSER-IP:5060 -> PSTNGW-IP:5060 CANCEL sip:445551212@PSTNGW-IP;user=phone SIP/2.0 Record-Route: <sip:011445551212@OPENSER-IP;lr=on;ftag=TAG> From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> Call-ID: CALLID@CUSTOMER-IP CSeq: 2 CANCEL Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0 Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Max-Forwards: 69 Supported: 100rel,replaces Content-Length: 0 and this also happens: # U 2006/11/06 18:40:10.691342 OPENSER-IP:5060 -> CUSTOMER-IP:5060 SIP/2.0 200 canceling From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=TAG2 Call-ID: CALLID@CUSTOMER-IP CSeq: 2 CANCEL Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Content-Length: 0 How was that last packet generated, where was it done? It's not in my openser.cfg, should I have sent back just such a message? If so, would this automagically sent message still have been sent? Next I get these two packets: # U 2006/11/06 18:40:10.761582 PSTNGW-IP:5060 -> OPENSER-IP:5060 SIP/2.0 200 OK Call-ID: CALLID@CUSTOMER-IP From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> Content-Length: 0 CSeq: 2 CANCEL Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0,SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Contact: sip:445551212@PSTNGW-IP:5060;user=phone Record-Route: <sip:011445551212@OPENSER-IP;lr=on;ftag=TAG> # U 2006/11/06 18:40:10.761616 PSTNGW-IP:5060 -> OPENSER-IP:5060 SIP/2.0 487 Request Terminated Call-ID: CALLID@CUSTOMER-IP From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 Content-Length: 0 CSeq: 2 INVITE Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0,SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH and my openser box replies with : # U 2006/11/06 18:40:10.762238 OPENSER-IP:5060 -> PSTNGW-IP:5060 ACK sip:445551212@PSTNGW-IP;user=phone SIP/2.0 Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0 From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG Call-ID: CALLID@CUSTOMER-IP To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 CSeq: 2 ACK Content-Length: 0 and sends on this to the UAC: # U 2006/11/06 18:40:10.762377 OPENSER-IP:5060 -> CUSTOMER-IP:5060 SIP/2.0 487 Request Terminated Call-ID: CALLID@CUSTOMER-IP From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 Content-Length: 0 CSeq: 2 INVITE Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Now, how was that generated? Is this part of the "tm" module? And then when I get this ACK: # U 2006/11/06 18:40:10.772875 CUSTOMER-IP:5060 -> OPENSER-IP:5060 ACK sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 Call-ID: CALLID@CUSTOMER-IP CSeq: 2 ACK Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Max-Forwards: 70 Contact: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP> Content-Length: 0 What should I do with it? I do nothing... if I should do nothing is there any special way I should do it? Or is this just handled by the tm module? What I don't get is the CANCEL is directed AT my openser box: CANCEL sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> just like the ACK: ACK sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 and I t_relay() the CANCEL but sit on the ACK... [View Less]

2 1

[Users] Some problem in Hung up ...... help me.........
by raviprakash sunkara 13 Nov '06

13 Nov '06

Hello Users, When I keep my Sip Server into firewall_router is not hung up, But with Direct Modem to My Sip Server then it hung ups finely, Sip server listen on 192.168.2.90 with public ip xx.xx.xx.xx.xx. See the Loation tables +----------+--------+----------------------------+------------------------+------+---------------------+-------+-----------------------------------------------+-------+---------------------+-------+----------------------+-----------------------+---------+ | … [View More]username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | user_agent | socket | methods | +----------+--------+----------------------------+------------------------+------+---------------------+-------+-----------------------------------------------+-------+---------------------+-------+----------------------+-----------------------+---------+ | 102 | | sip:102@59.144.72.234:5060 | sip:59.144.72.234:5060 | NULL | 2006-11-07 03:17:42 | -1.00 | 414D9B3694024D7D94F21AF17E63460E(a)xx.xx.xx.xx | 61160 | 2006-11-07 02:47:42 | 1 | X-Lite release 1103m | udp:192.168.2.90:5060 | NULL | | 101 | | sip:101@61.17.248.68:5060 | sip:61.17.248.68:5060 | NULL | 2006-11-07 03:17:04 | -1.00 | 532BD0D57F89495CB42AFA7B7D760BE2(a)xx.xx.xx.xx | 9830 | 2006-11-07 02:47:04 | 1 | X-Lite release 1103m | udp:192.168.2.90:5060 | NULL | +----------+--------+----------------------------+------------------------+------+---------------------+-------+-----------------------------------------------+-------+---------------------+-------+----------------------+-----------------------+---------+ my openser nat configure is like route[2] { if(nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_resgiter(); force_rport(); setflag(6); } else { if (method == "INVITE") { fix_nated_sdp("2") }; fix_nated_contact() force_rport() setflag(6); }; } In X-lite log..... After the INVITE Method got response from server ( xx.xx.xx.xx.xx) .... ACK is sending like That ..... and also Bye method After the INvite is posseed Ack Send >> 192.168.2.90:5060 ... But CANCEL method send properly............ Help me -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com [View Less]

2 1

[Users] exec module : why not in onreply_route ?
by Jean-François SMIGIELSKI 13 Nov '06

13 Nov '06

Hello, all. I would know why we should avoid using the exec module exported functions outside out the request_route and on_failure_route blocks? (and so ... why it is forbidden). Answers carry useful informations (e.g. SDP) that we could want to parse with an external command. Maybe is there a more appropriate way to achieve this, and I mail you for this reason. Thanks once again, JF Smigielski ________________________________________________________________________ iBELGIQUE, exprimez-… [View More]

3 2

[Serusers] off-topic: Sofphones with Redirect Suport
by Felipe Louback 13 Nov '06

13 Nov '06

Hi all, I need a softphone that supports redirect. Let me explain: In all free softphones I download, I have an "outbound proxy" field. So, if my server returns and 302 Moved Temporarily, the softphone will get the Contact header field and try to call the user again using the outbound proxy, and the outbound will reply with an 302 and I will get an loop. I need a softphone who will use a location server to query the location of a user and after that the softphone will make the call directly, … [View More]

2 1

[Serusers] SER pre-release ser_ctl
by SER LIST 13 Nov '06

13 Nov '06

Hello, When I try to run the ser_ctl command to do something like ser_ctl version or any other command, I get the error ser_ctl: (111, 'Connection refused'): error SER is running while I type the command. Any hints will be appreciated. _________________________________________________________________ Buy, Load, Play. The new Sympatico / MSN Music Store works seamlessly with Windows Media Player. Just Click PLAY. http://musicstore.sympatico.msn.ca/content/viewer.aspx?cid=SMS_Sept192006

3 3

Re: [Serusers] SER pre-release - PRESENCE ISSUES
by SER LIST 12 Nov '06

12 Nov '06

Thank you. settig the xcap_root for the xcap module fixed the two errors relating to xcap_root. BUT I am still getting the two errors relating to the lookup_domain command, namely: ------------------------------------- ser: parse error (192,24-25): unknown command, missing loadmodule? ser: parse error (261,26-27): unknown command, missing loadmodule? ser: ERROR: bad config file (2 errors) ser startup failed ---------------------------------- I am using the presence example file. The … [View More]excerpt from my ser.cfg file is (I have marked the LINES where the error occurs): ---- start --- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.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/domain.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/dialog.so" loadmodule "/usr/local/lib/ser/modules/rls.so" loadmodule "/usr/local/lib/ser/modules/pa.so" loadmodule "/usr/local/lib/ser/modules/xcap.so" loadmodule "/usr/local/lib/ser/modules/presence_b2b.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/fifo.so" loadmodule "/usr/local/lib/ser/modules/xmlrpc.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" { LINES TRUNCATED} if (uri==myself) { if (!lookup_domain("To")) { <-------------------------- LINE 192 xlog("L_ERR", "Unknown domain to: %tu from: %fu\n"); route(1); break; } if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); break; }; (lines truncated) if (lookup_domain("From")) { <---------------------- LINE 261 if (lookup_user("From")) { if (is_simple_rls_target("$uid-list")) { # log(1, "it is simple subscription!\n"); # takes From UID and makes XCAP query for user's # list named "default" if (!query_resource_list("default")) { t_reply("404", "No such user list"); break; } } } } ----------- END --------------------- Any suggestions?? THanks in advance >From: samuel <samu60(a)gmail.com> >To: "SER LIST" <sergrp(a)hotmail.com> >CC: serusers(a)iptel.org, serdev(a)iptel.org >Subject: Re: [Serusers] SER pre-release - PRESENCE ISSUES >Date: Fri, 10 Nov 2006 10:14:17 +0100 > >XCAP parameters where removed from rls and pa and a new module, called >xcap contains the xcap root so you have to delete pa and rls xcap >parameter, load the new module xcap.so and set the xcap_root parameter >of xcap module to the directory having the doc structure. > >About the lookup_domain, there can be a typo. Can you post the config >file syntax?? > >Hope it helps! > >Samuel > >2006/11/10, SER LIST <sergrp(a)hotmail.com>: >>I installed the SER pre-release along w/ all modules. The problem is that >>presence now does not run (it was working w/ the presence-release-10.99). >>The errors I get are: >>---------------------------------------------------------------------------------------------------------- >>set_mod_param_regex: parameter <xcap_root> not found in module <rls> >>ser: parse error (82,20-21): Can't set module parameter >>set_mod_param_regex: parameter <auth_xcap_root> not found in module <pa> >>parse error (97,20-21): Can't set module parameter >>ser: parse error (186,24-25): unknown command, missing loadmodule? >>parse error (255,26-27): unknown command, missing loadmodule? >>ERROR: bad config file (4 errors) >>------------------------------------------------------------------------------------------------------------- >>I checked the pa module code and also the rls code and the parameters >>auth_xcap_root and xcap_root are NOT defined !!!! >> >>The error on line 185 and 255 refer to the "Lookup_domain" function. I >>verified that the domain module has been compiled and included in the >>ser.cfrg file _________________________________________________________________ Ready for the world's first international mobile film festival celebrating the creative potential of today's youth? Check out Mobile Jam Fest for your a chance to WIN $10,000! www.mobilejamfest.com [View Less]

3 8

[Users] Forwarding requests
by Harry Harcrow 12 Nov '06

12 Nov '06

I have my SER server running on a Linux machine and I am attempting to forward some requests to a Cisco SIP server. I have been running X-lite clients on windows and the same Linux box that the SER server is running on. I have no problem completing calls between the X-lite clients OR with either X-lite client calling the Cisco SIP server directly. However, when I try to make a call from either X-lite client through the SER server I eventually get timeout methods. I am trapping those calls and … [View More]

2 1

[Serusers] Suggestion
by Eddie Johnson Jr 12 Nov '06

12 Nov '06

I am new to setting up SER and this is the test setup. 1. DHCP running from router 2. SJ Phone on laptop 192.168.X.X 3. Grandstream Budgetone 101 192.168.X.X 4. SIP Express Router 192.168.X.X 5. Asterisk Server 192.168.X.X attached to tdm400p card is an analog phone Will this work internally? There is no connection to a telco. The SJ phone is showing the following: nat/firewall: Port Restricted Cone NAT SIP calls use outbound proxy: 192.168.X.X (SER) Will I have to … [View More]

1 0

[Serusers] SER not sending 486 Busy Here
by Ravi ***** 11 Nov '06

11 Nov '06

Hi Users , I am Using SER-0.96 , When I make a call from SER to Pstn it works fine I mean is all "INV"BYE"CAN". But when I recieved a call from Pstn I cannot cancel the call because X-lite is sending to pstn gateway 480 temp. service not available instead of 486 Busy Here. I am using default config. but I dont understand where it went wrong . Any Idea or help greatly appreciated … [View More]

1 0

Re: [Serusers] Re: How to change Caller ID in FROM field ?
by lawrence k.y. lin 10 Nov '06

10 Nov '06

Hi Kristian. Thanks for your email. I am using ser 0.9.6 and did not find uac_replace_from function. Then I did a search and found that it is in openser. I did install openser and it worked. Do you or anyone know if this function will be added into ser release ? Please advise, thanks, Larry >------------------------------ >Message: 20 >Date: Wed, 25 Oct 2006 13:11:06 +0200 >From: Kristian Larsson <kristian(a)netatonce.se> >Subject: Re: [Serusers] Re: How to … [View More]

1 0

[Serusers] fr_inv_timer + avp not working under high load
by Marcus Hunger 10 Nov '06

10 Nov '06

Hi list, I am trying to realise dynamic invite-timeout-timers loading timer-settings per call from my database using avp. It seems to work perfectly on my development-system but fails nearly completly in my high loaded life-environment. According to my logs, timer values are set correctly (eg.: reply_received: FR_INV_TIMER = 15, see modules/tm/t_reply.c:1371), but most of the times the timers have no effect. My setup is similar to http://siprouter.onsip.org/doc/gettingstarted/ch09.html, … [View More]

4 6

[Users] Problem with TCP INVITE being bridged by OpenSER to UDP
by Russ Daigle 10 Nov '06

10 Nov '06

I have a scenario where user Alice registered with OpenSER with UDP port 5070. Then, I have user Bob sending an invite to Alice via the OpenSER using TCP. Hence, OpenSER is bridging my TCP SIP requests to UDP (and vice versa for the responses). Here is the sequence: Alice ----- TCP Invite -----> OpenSER ---------- UDP Invite ------ > Alice Alice responds back with 180 ringing, then 200 Ok. Upon getting the 200 OK, Bob sends an ACK (to complete the INVITE), followed … [View More]by an Immediate BYE. THE TROUBLE IS...... the ACK and the BYE that Bob sends are dropped by OpenSER, and never forwarded onto Alice!!!! I see nothing wrong with the messages. WHat is happening? sip:~ russdaigle$ sudo tcpdump -i en0 -A -q -s 2000 port 5060 Password: tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on en0, link-type EN10MB (Ethernet), capture size 2000 bytes The following shows an sip-invite-bye from mutator using TCP to mjsip (via OPENSER). Everything after the ACK is dropped and not sent on. Then, is a similar but UDP exchange. In that case, we get a 404 not found in response to the BYE!!! 01:17:59.553909 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 588 INVITE sip:alice@siptest2.com:5060 SIP/2.0 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Contact: <sip:bob@10.10.2.6>;transport=TCP Max-Forwards: 70 Content-Type: application/sdp Content-Length: 195 v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 10.10.2.6 t=0 0 m=audio 5000 RTP/AVP 0 4 97 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:97 telephone-event/8000 01:17:59.554929 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 536 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Server: OpenSer (1.1.0-notls (i386/darwin)) Content-Length: 0 Warning: 392 10.10.1.234:5060 "Noisy feedback tells: pid=11962 req_src_ip=10.10.2.6 req_src_port=50135 in_uri=sip:alice@siptest2.com: 5060 out_uri=sip:alice@10.10.2.6:5070 via_cnt==1" 01:17:59.555035 IP 10.10.1.234.sip > 10.10.2.6.5070: UDP, length 814 INVITE sip:alice@10.10.2.6:5070 SIP/2.0 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Contact: <sip:bob@10.10.2.6>;transport=TCP Max-Forwards: 69 Content-Type: application/sdp Content-Length: 195 P-hint: usrloc applied v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 10.10.2.6 t=0 0 m=audio 5000 RTP/AVP 0 4 97 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:97 telephone-event/8000 01:17:59.667962 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 364 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:17:59.699680 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 505 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:17:59.699683 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 728 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 01:17:59.700273 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 440 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:17:59.700299 IP 10.10.1.234.sip > 10.10.2.6.50135: tcp 663 SIP/2.0 200 OK Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 01:17:59.804566 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 435 ACK sip:alice@siptest2.com:5060 SIP/2.0 Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Route: <sip:10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK59972 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 CSeq: 1 ACK Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Max-Forwards: 70 01:17:59.839493 IP 10.10.2.6.50135 > 10.10.1.234.sip: tcp 435 BYE sip:alice@siptest2.com:5060 SIP/2.0 Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Route: <sip:10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK68444 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 CSeq: 2 BYE Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com Max-Forwards: 70 01:18:00.199306 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 728 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bK9316.e5ab8ce2.0;i=b1 Via: SIP/2.0/TCP 10.10.2.6:50135;branch=z9hG4bK85478043 Record-Route: <sip:10.10.1.234;r2=on;lr=on;ftag=fromhackblah> Record-Route: <sip: 10.10.1.234;transport=tcp;r2=on;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=7c452a3254b2e183 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: SYRGNUXWKNZNXXPXUAMUTJXXPPMXWKVJFAXUNUUTGG(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 [View Less]

2 3

[Serusers] IM and silo table in MySQL
by davor jovanovic 10 Nov '06

10 Nov '06

Hi all, I have a problem with silo table in MySQL. messages are stored in the table when user is offline, and when comes back online messages are delivered to the client. Problem is that the messages aren't erased from table after delivery . Am I missing something in my ser.cfg? I'm using ser-0.9.6 and XLite 3.0. I will appreciate any help. thanks /davor loadmodule "/usr/lib/ser/modules/msilo.so" modparam("auth_db|uri_db|usrloc|permissions|group|acc_db|msilo", "db_url", "mysql://… [View More]ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo", "check_time", 10) modparam("msilo", "clean_period", 3) modparam("msilo", "use_contact", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; ... } else if (method=="MESSAGE") { route(8); break; }; ... route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!www_authorize("","subscriber")) { www_challenge("","0"); break; ... m_dump(); } route[8] { ########################## # Message Handler # ########################### if (!lookup("location")) { if (method == "MESSAGE") { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("1")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; }; }; route(1); t_on_failure("1"); } ......... failure_route[1] { if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; [View Less]

1 0

[Serusers] Problem with radius-sql and ser-0.9.6
by Lokesh Kumar 10 Nov '06

10 Nov '06

Hello, I am running old ser version 0.9.6, where I am authenticating on radius and keeping the users record in default sql database of radius. But it is not authenticating, the logs are mentioned below. But it worked absolutely fine with radius users files. I have the entry for the user in radcheck file but still it is saying user not found. Can anyone give any hint where I am doing wrong. Going to the next request --- Walking the entire request list --- Waking up in 1 … [View More]seconds... rad_recv: Access-Request packet from host 127.0.0.1:54086, id=241, length=262 User-Name = "211069020(a)voip.nortenet.pt" Digest-Attributes = 0x0a0b323131303639303230 Digest-Attributes = 0x0112766f69702e6e6f7274656e65742e7074 Digest-Attributes = 0x022a3435353436346634343937623539656362346335633261323364656436656532393934 3565316432 Digest-Attributes = 0x04167369703a766f69702e6e6f7274656e65742e7074 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303031 Digest-Attributes = 0x08103132373935383532383139343033 Digest-Response = "2ae0ba094f508b9dff7bb56d96649875" Service-Type = Sip-Session Sip-Uri-User = "211069020" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 1 modcall[authorize]: module "preprocess" returns ok for request 1 modcall[authorize]: module "chap" returns noop for request 1 modcall[authorize]: module "mschap" returns noop for request 1 rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 1 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 1 radius_xlat: '211069020(a)voip.nortenet.pt' rlm_sql (sql): sql_set_user escaped user --> '211069020(a)voip.nortenet.pt' radius_xlat: 'SELECT id, UserName, Attribute, Value, op FROM radcheck WHERE Username = '211069020(a)voip.nortenet.pt' ORDER BY id' rlm_sql (sql): Reserving sql socket id: 13 rlm_sql (sql): User 211069020(a)voip.nortenet.pt not found in radcheck radius_xlat: 'SELECT radgroupcheck.id,radgroupcheck.GroupName,radgroupcheck.Attribute,radgroupche ck.Value,radgroupcheck.op FROM radgroupcheck,usergroup WHERE usergroup.Username = '211069020(a)voip.nortenet.pt' AND usergroup.GroupName = radgroupcheck.GroupName ORDER BY radgroupcheck.id' radius_xlat: 'SELECT radgroupreply.id,radgroupreply.GroupName,radgroupreply.Attribute,radgrouprep ly.Value,radgroupreply.op FROM radgroupreply,usergroup WHERE usergroup.Username = '211069020(a)voip.nortenet.pt' AND usergroup.GroupName = radgroupreply.GroupName ORDER BY radgroupreply.id' rlm_sql (sql): User 211069020(a)voip.nortenet.pt not found in radgroupcheck rlm_sql (sql): Released sql socket id: 13 rlm_sql (sql): User not found modcall[authorize]: module "sql" returns notfound for request 1 modcall: leaving group authorize (returns ok) for request 1 rad_check_password: Found Auth-Type DIGEST auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 1 rlm_digest: Configuration item "User-Password" or Digest-HA1 is required for authentication. modcall[authenticate]: module "digest" returns invalid for request 1 modcall: leaving group authenticate (returns invalid) for request 1 auth: Failed to validate the user. Login incorrect: [211069020(a)voip.nortenet.pt] (from client localhost port 5060) Delaying request 1 for 1 seconds Finished request 1 Going to the next request --- Walking the entire request list --- Waking up in 1 seconds... --- Walking the entire request list --- Sending Access-Reject of id 240 to 127.0.0.1 port 54085 Waking up in 1 seconds... --- Walking the entire request list --- Sending Access-Reject of id 241 to 127.0.0.1 port 54086 Waking up in 3 seconds... --- Walking the entire request list --- Cleaning up request 0 ID 240 with timestamp 455463c8 Waking up in 1 seconds... --- Walking the entire request list --- Cleaning up request 1 ID 241 with timestamp 455463c9 Nothing to do. Sleeping until we see a request. Thanks very much Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.14.1/527 - Release Date: 11/9/2006 [View Less]

2 1

[Users] Problems with TCP REGISTER, UDP REGISTER works fine
by Russ Daigle 10 Nov '06

10 Nov '06

For some reason, I am having problems with having TCP REGISTER messages being accepted by openser. The following is a REGISTER I send to Open SER. It is never responded to. Do you see anything wrong with it? It should be registering port 5060 TCP. 00:19:42.157347 IP 10.10.2.6.50083 > 10.10.1.234.sip: tcp 345 REGISTER sip:siptest2.com:5060 SIP/2.0^M Via: SIP/2.0/TCP 10.10.2.6:50083;branch=z9hG4bK45^M From: Alice <sip:alice@siptest2.com>;tag=fromhackblah^M To: Alice <sip:… [View More]

4 4

[Users] Problem with BYE method getting a 404 not found response
by Russ Daigle 10 Nov '06

10 Nov '06

I can reproduce this easily. Do you see anything wrong with the below packet traces? Here is the sequence: Bob ----- UDP Invite -----> OpenSER ---------- UDP Invite ------ > Alice Alice responds back with 180 ringing, then 200 Ok. Upon getting the 200 OK, Bob sends an ACK (to complete the INVITE), followed by an Immediate BYE. THE TROUBLE IS...... the ACK that Bob sends is dropped (never forwarded to Alice) and the BYE that Bob sends is responded by OpenSER with a … [View More]404 not found, and never forwarded onto Alice!!!! I see nothing wrong with the messages. WHat is happening? Below is the message exchanges. Followed by "openserctl ul show" to show Alice is still registered. 01:18:19.801230 IP 10.10.2.6.5080 > 10.10.1.234.sip: UDP, length 597 INVITE sip:alice@siptest2.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Contact: <sip:bob@10.10.2.6:5080>;transport=UDP Max-Forwards: 70 Content-Type: application/sdp Content-Length: 195 v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 10.10.2.6 t=0 0 m=audio 5000 RTP/AVP 0 4 97 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:97 telephone-event/8000 01:18:19.802801 IP 10.10.1.234.sip > 10.10.2.6.5080: UDP, length 539 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Server: OpenSer (1.1.0-notls (i386/darwin)) Content-Length: 0 Warning: 392 10.10.1.234:5060 "Noisy feedback tells: pid=11955 req_src_ip=10.10.2.6 req_src_port=5080 in_uri=sip:alice@siptest2.com: 5060 out_uri=sip:alice@10.10.2.6:5070 via_cnt==1" 01:18:19.802885 IP 10.10.1.234.sip > 10.10.2.6.5070: UDP, length 734 INVITE sip:alice@10.10.2.6:5070 SIP/2.0 Record-Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bKcbde.5492991.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060> CSeq: 1 INVITE Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Contact: <sip:bob@10.10.2.6:5080>;transport=UDP Max-Forwards: 69 Content-Type: application/sdp Content-Length: 195 P-hint: usrloc applied v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 10.10.2.6 t=0 0 m=audio 5000 RTP/AVP 0 4 97 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000/1 a=rtpmap:97 telephone-event/8000 01:18:19.891076 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 362 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bKcbde.5492991.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:18:19.957284 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 420 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bKcbde.5492991.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 Record-Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060> From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com CSeq: 1 INVITE Server: mjsip stack 1.6 Content-Length: 0 01:18:19.957363 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 643 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bKcbde.5492991.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 Record-Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 01:18:19.957995 IP 10.10.1.234.sip > 10.10.2.6.5080: UDP, length 584 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 Record-Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 01:18:20.047797 IP 10.10.2.6.5080 > 10.10.1.234.sip: UDP, length 365 ACK sip:alice@siptest2.com:5060 SIP/2.0 Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK0 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 CSeq: 1 ACK Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Max-Forwards: 70 01:18:20.047803 IP 10.10.2.6.5080 > 10.10.1.234.sip: UDP, length 370 BYE sip:alice@siptest2.com:5060 SIP/2.0 Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK767774 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 CSeq: 2 BYE Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Max-Forwards: 70 01:18:20.049706 IP 10.10.1.234.sip > 10.10.2.6.5080: UDP, length 561 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK767774 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 CSeq: 2 BYE Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com Server: OpenSer (1.1.0-notls (i386/darwin)) Content-Length: 0 Warning: 392 10.10.1.234:5060 "Noisy feedback tells: pid=11955 req_src_ip=10.10.1.234 req_src_port=5060 in_uri=sip: 10.10.1.234;lr=on;ftag=fromhackblah out_uri=sip: 10.10.1.234;lr=on;ftag=fromhackblah via_cnt==2" 01:18:20.421033 IP 10.10.2.6.5070 > 10.10.1.234.sip: UDP, length 643 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.234;branch=z9hG4bKcbde.5492991.0 Via: SIP/2.0/UDP 10.10.2.6:5080;branch=z9hG4bK15 Record-Route: <sip:10.10.1.234;lr=on;ftag=fromhackblah> To: Alice <sip:alice@siptest2.com:5060>;tag=d418a42d9b1c6819 From: Bob <sip:bob@siptest2.com>;tag=fromhackblah Call-ID: TUNQQQNWAYJDYCXGYAIWQEMRGOLILIBMMJMLNTFVLIBIBUORMDALF(a)musecurity.com CSeq: 1 INVITE Contact: <sip:alice@192.168.0.3:5070> Server: mjsip stack 1.6 Content-Length: 135 Content-Type: application/sdp v=0 o=- 817933771 817933775 IN IP4 10.10.2.6 s=darkness c=IN IP4 192.168.0.3 t=0 0 m=audio 21000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 'alice' sip:~ russdaigle$ sudo openserctl ul show WARINIG: no database engine found - tried '' Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: 0x2191b90 d_ll { n : 0 first: 0x0 last : 0x0 } ---/Domain--- ---Domain--- name : 'location' size : 512 table: 0x218fa08 d_ll { n : 1 first: 0x2193c90 last : 0x2193c90 } ...Record(0x2193c90)... domain: 'location' aor : 'alice' ~~~Contact(0x2193cd0)~~~ domain : 'location' aor : 'alice' Contact : 'sip:alice@10.10.2.6:5070' Expires : 794 q : Call-ID : 'XQQZRLTJDMQZUMPKAJZBQEVGSPRAFOXTCGRQQCSIWPKMJLTGHFC(a)musecurity.com' CSeq : 1 User-Agent: 'n/a' received : '' Path : '' State : CS_NEW Flags : 0 Sock : 10.10.1.234:5060 (0xfa6e8) Methods : 4294967295 next : 0x0 prev : 0x0 ~~~/Contact~~~~ .../Record... ---/Domain--- ===/Domain list=== [View Less]

2 1

[Users] certificate recommendations
by Mark Price 10 Nov '06

10 Nov '06

Can I have some recommendations about what company and what package to go with for a certificate to work with openser? I have a cert from godaddy, and it seems that it won't work with openser because of the intermediate certificate that they require you to use. Godaddy issues a certificate, a private key and an intermediate certificate (the intermediate certificate So openser loads just fine if I set: tls_certifcate=cert.pem tls_preivate_key=cert.key but the phone still fails to validate the … [View More]

2 1

[Users] Multiple CA
by Gregoire 10 Nov '06

10 Nov '06

Hi everybody! I am using OpenSER 1.1 with TLS. I have generate the client and server certificate with the scripts gen_rootCA.sh and gen_usercert.sh. Everything works fine, but I have generate certificate for my UA with another CA and I have added this CA to the file user-cacert.pem. When I try to connect with my UA, OpenSER logs an error like: "tls_error: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca" My file user-cacert.pem looks like: -------BEGIN CERTIFICATE------ … [View More]

2 4

[Users] Measuring QoS with OpenSER
by Andreas Granig 10 Nov '06

10 Nov '06

Hi, I just came across http://www.golem.de/0602/43572.html (sorry, German). It states that AVM contributed GPL'ed source code to OpenSER, which allows an UAC to send QoS measurement samples to the SIP proxy after finishing a call. The goal is to have steady QoS monitoring without changing or extending the network topology. Are these extensions somewhere in CVS or isn't it intended to be released to the public? Andy

4 7

[Serusers] Change INVITE and Replace
by Benito Caracuel 10 Nov '06

10 Nov '06

Hello all, I need help about this: When an INVITE packet is received by SER, a Perl script changes the SDP of this INVITE and the result of this script (new INVITE with new SDP) is saved in a txt file. My question is: How can I do that SER sends the new INVITE, which is in a txt file. I have tried to do this with REPLACE command but I think is not possible because It doesn`t accept "regular expression" as substitute and I don´t know how save this new SDP in variables which can be … [View More]

4 3

[SerUsers]: Mysql DB at different machine
by Kamal.Mann＠t-systems.com 10 Nov '06

10 Nov '06

Hi All I am trying to implement ser database (mysql part) on different machine. Ser is installed on machine say 'A' and Mysql DB is on 'B'. In ser.cfg I changed url to connect to database everywhere in ser.cfg file with machine B's ip (eg. modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@10.25.117.86/ser") n in rest too). But I am getting the following error when I checked ser logs: Nov 10 14:12:40 vm01 /usr/sbin/ser[12979]: get_connection(): Connection not found … [View More]

2 4

[Serusers] SER pre-release - PRESENCE ISSUES
by SER LIST 10 Nov '06

10 Nov '06

I installed the SER pre-release along w/ all modules. The problem is that presence now does not run (it was working w/ the presence-release-10.99). The errors I get are: ---------------------------------------------------------------------------------------------------------- set_mod_param_regex: parameter <xcap_root> not found in module <rls> ser: parse error (82,20-21): Can't set module parameter set_mod_param_regex: parameter <auth_xcap_root> not found in module <pa… [View More]> parse error (97,20-21): Can't set module parameter ser: parse error (186,24-25): unknown command, missing loadmodule? parse error (255,26-27): unknown command, missing loadmodule? ERROR: bad config file (4 errors) ------------------------------------------------------------------------------------------------------------- I checked the pa module code and also the rls code and the parameters auth_xcap_root and xcap_root are NOT defined !!!! The error on line 185 and 255 refer to the "Lookup_domain" function. I verified that the domain module has been compiled and included in the ser.cfrg file >From: "Greger V. Teigre" <greger(a)teigre.com> >To: Jiri Kuthan <jiri(a)iptel.org> >CC: SER developer mailing list <serdev(a)iptel.org>, serusers(a)iptel.org >Subject: Re: [Serusers] new major SER pre-release, Ottendorf,is out for >testing >Date: Tue, 07 Nov 2006 15:37:17 +0100 > >Just a bit more info on this pre-release adding to Jiri's long post (if you >can take it ;-): > >* The mentioned http://www.iptel.org/ser/doc/010whatsnew will be updated as >we go with new documentation. Any registered user on iptel.org can add to >or modify documentation and all help is welcome. If you are uncertain about >how to proceed, send me an email > >* As Jiri mentioned, any input and help is appreciated. If you want to >contribute, but don't know what to do, have a look at the to-dos page: >http://www.iptel.org/todos > >* This pre-release is, without basic documentation, a bit for the die-hard. >This is an overview of what we are working (and need help) on to make it >easier accessible: - A tar source package ready for simple update from >CVS (similar to the ONsip.org source package) > - Better "kickstart" instructions, including more details in What's new >0.10 (above link) > - Updated SER - Getting Started configuration files and updated SER - >Getting Started document > - Database migration script (contact me if you want to try out an >existing python migration script and test/improve) > >* Also, in addition to the ser.cfg and Nils recently commited ser-basic.cfg >in CVS, take a look at an example config file: >http://www.iptel.org/basic_changes_in_configuration_file_0 > >*An optimized NAT-handled config file can be found here: >http://www.iptel.org/ser/howtos/optimizing_the_use_of_rtp_proxy > >Happy testing! >g-) > >Jiri Kuthan wrote: >>Ladies and gentlemen, girls and boys, >> >>the new testing pre-release of the best SIP proxy server ever, SER, is >>out! >> >>This major pre-release, codenamed Ottendorf, includes fundamental >>improvements to the software. The improvements allow SER users to achieve >>significantly higher managability and reliability by better data mode, >>interop features, security, etc. They recollect voluminous feedback we >>have received from iptel customers and open source users. In particular, >>the new features include but are not limited to: >>- improved data model which eliminates any possible ambiguities. >>- key under-the-hood changes with impact on performance: timer subsystem >> overhaul (now fast, with fine resolution, high accuracy and >>variability), TCP processing overhaul (higher scalability) >>- key reliability changes: DNS cache and IP blacklisting >>- key interop changes: support for STUN-bis, presence server aligned to >>latest >> IETF development, sanity module for early detection of invalid messages >>- more management: management features are now available via XML-RPC >>remotely, >> as well as via an alternative binary protocol >>- more security: ratelimit module, more check types in permissions module >>- scripting features: select operator in combination with revised AVPs, >> named flags and routes, more types of routes (onsend, timer, branch) >>- applications: presence has been greatly updated to keep pace with >>development in the IETF/SIMPLE (includes asynchronous subscription >> authorization, extended PIDF); web interface includes many new >>managibility >> features (multidomain, templates and multilanguage to name some). New >> scripting flexibility allows safely for features like individual >>ringing >> timers and direct access to SIP message elements. >>- new tools: Nils's script execution tree tool >> http://lists.iptel.org/pipermail/serdev/2006-October/008124.html >> and refurbished ser_ctl >>see the following links for some additional details: >>- http://www.iptel.org/ser/doc/010whatsnew - >>http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/NEWS?rev=HEAD >> >> >>Accompanying software, SEMS and SERWEB have been greatly overhauled as >>well. >>SEMS now features improved performance (single-threaded model now), >>service >>composition via python-IVR-scripting and B2BUA, adaptive playout buffer >>for >>better voice quality, and more. SERWEB features new multilevel >>administrative >>access system, multidomain support, per-domain customization, remote >>xml-rpc >>access, multilanguage support, and more. The new command-line utility, >>SER_CTL, provides also many great improvements over the previous tool. >> >>Interop >>------- >>Shortly, we have been keeping the bar high through our participation in >>the IETF >>and SIPITs. SER implements latest interop changes from the IETF (STUN-bis, >>presence >>+ XCAP, to name the most important ones) and both new features and legacy >>features were extensively and successfully tested middle of October in >>SIPit. >> >>Next Steps >>----------- >>Keep in mind, this is a pre-release for testing purposes. It is now call >>on >>users to test the software hard and provide feedback. In the meantime, the >>codebase remains frozen and we only commit patches fixing various bugs. >>Once >>we get good confidence that Ottendorf has stabilized and is being >>adequately >>documented, we will declare the final release. No date has been determined >>yet. >>Also, we consider changing CVS to SVN. >> >>Again, and this is very important: the final release will be as good as >>the >>feedback we obtain for this pre-release. Please do spend your time sending >>solid feedback back to [serusers]. >> >> >>Roadmap >>------- >>We keep rolling. We are still debating with our generous contributors what >>is the next possible good things to work on, the topics include but are >>not limited >>to as of now: >>- documentation. >>- databases are important. LDAP and BDB, DB-API revision. >>- interop: identity, TLS revision >>- advanced scripting (scripting engine, vars returns, AVPs) >>- uncountable improvements (futex, dev/rtc, tm/refcnts, tcp fd cache, >>.....) >> >>Trivia >>-------- >>The release name Ottendorf is coming from a place in Saxonian Switzerland, >>where many iptel fellows and their friends spent a weekend during this >>Indian summer. >>http://www.iptel.org/jan/gallery/saxonian_switzerland_2006/ >>http://www.iptel.org/~andi/2006-09-22-Ottendorf/ >>http://www.iptel.org/~vlada/saxon_switzerland/index.html >>http://www.ohlmeier.de/albums/ottendorf_2006/index.html >> >> >>Acknowledgments >>--------------- >>This grand achievement is result of extremely many folks who did lot >>of hard work. Out of folks affiliated with iptel, Andrei and Jan spent >>immense effort on moving SER forward, with great support from Hallik >>(command-line tool), Michal, Miklos, Nils, Tomas, Vaclav (presence), Vlada >>(new STUN). In this context, I would like to thank to our company for the >>great support. Recently, Bogdan Pintea greatly helped to make the upcoming >>release better and more stable under sponsorship of his company, iptego. >>transnexus provided osp support. >> >>SEMS has been greatly improved thanks to Stefan and Raphael. >>Credits for serweb go to Karel. >> >>Same importantly, SER has been immensely benefitting of contributors, >>to name in alphabetic order at least the 'top-of-the-iceberg' fellows: >>Bogdan Pintea, Cesc Santa, Greg Fausak, Greger Teigre, Hendrik Scholz, >><http://tutpro.com/>Juha Heinänen, Klaus Darilion, Maxim Sobolev, Peter >>Nixon, Uli Abend. >> >>Great support on the mailing list has been provided by Greger V. Teigre, >>Java Rockx, Klaus Darilion, Zeus Ng, and more. >> >>Simply, this list is too short and incomplete to capture all the >>generous contributions that came in -- my appologies for everyone >>else I have missed. >> >> >>Warnings and Disclaimers >>------------------------ >>This is a TESTING version. We are encouraging people to give it intense >>testing and provide us with feedback, so that we can properly assess >>what it takes to declare it final. The volume and number of changes >>is big, caution is advised. >> >>All GPL warnings and disclaimers apply. >> >>This is SER, which is not the derivate software known as openser. We have >>no ideas of in what extent and timeline derivate projects follow >>(or not) SER. >> >>------------------------------------------------------------------------------- >> >>To Give it a Try >>----------------- >>warning: this is just a 'jump-start' for black-belt users. It is not a >>replacement for documentation or a detailed step-by-step installation >>guide. It assumes knowledge of operating system, SIP, and SER. >> >> >>PREPARATION >>- make sure mysql is running. In the default configuration, SER assumes it >> to be running on localhost, if this does not hold, you need to change >> ser.cfg. >>- make sure python is installed (needed for new command-line utilities) >>SER INSTALLATION FROM CVS >>- Download from CVS/HEAD using instructions at >>http://www.iptel.org/downloads >>- Compile it: make include_modules="mysql xmlrpc" all >>- Install it the default way of your administrative privileges allow to: >> make include_modules="mysql xmlrpc" install >>- Download command-line tool (see more documentation under >> >>http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/serctl/README?rev=…) >> - cvs co serctl >>- Initiate database: >> ./scripts/mysql/ser_mysql.sh create >>START >>- Start it. Call: ser >>- Begin provisioning users with the new serctl (the capital word are just >>place holders) >> # ser_ctl domain add DOMAINNAME >> # ser_ctl user add USERNAME@DOMAINNAME -p PASSWORD >> # If you want to have Asserted Identity header for your user >> # - ser_attr add uid=UID asserted_id="PID" >> # If you want to have gateway support >> # - ser_db add attr_types name=gw_ip rich_type=string raw_type=2 \ >> description="The gateway IP for the default ser.cfg" >>default_flags=33 >> # - ser_attr add global gw_ip=GATEWAY-IP >>- more things you may wish to do: >> - familiarize yourself with current ser.cfg >> - send us feedback. >> >> >> >> >> >>-- >>Jiri Kuthan http://iptel.org/~jiri/ >> >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Experience Live Search from your PC or mobile device today. http://www.live.com/?mkt=en-ca [View Less]

2 1

[Serusers] New developer
by Gustavo Passos Tourinho 10 Nov '06

10 Nov '06

Hello There! I'm trying to develop some new functions to be used at ser.cfg. I was looking the serdev.pdf but I didn't figure out how can I integrate my new function with ser.cfg. Is there some sort of "how to develop new modules for ser"? Thanks in advanced, Tourinho

2 1

[Serusers] Problem in using ser_ctl
by Ajay Kolambekar 10 Nov '06

10 Nov '06

Hi, I have installed SER on Redhat Linux 9 machine. I want to use ser_ctl to add/view users or domains. When I give command set_ctl user show I am getting an error like ser_ctl: /root/Python-2.5/lib/python2.5/site-packages/_mysql.so: undefined symbol: mysql_set_character_set: ImportError'> Does anybody know how to solve this problem? Thanks in advance, Ajay DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems … [View More]

2 2

[Serusers] new major SER pre-release, Ottendorf, is out for testing
by Jiri Kuthan 10 Nov '06

10 Nov '06

Ladies and gentlemen, girls and boys, the new testing pre-release of the best SIP proxy server ever, SER, is out! This major pre-release, codenamed Ottendorf, includes fundamental improvements to the software. The improvements allow SER users to achieve significantly higher managability and reliability by better data mode, interop features, security, etc. They recollect voluminous feedback we have received from iptel customers and open source users. In particular, the new features … [View More]include but are not limited to: - improved data model which eliminates any possible ambiguities. - key under-the-hood changes with impact on performance: timer subsystem overhaul (now fast, with fine resolution, high accuracy and variability), TCP processing overhaul (higher scalability) - key reliability changes: DNS cache and IP blacklisting - key interop changes: support for STUN-bis, presence server aligned to latest IETF development, sanity module for early detection of invalid messages - more management: management features are now available via XML-RPC remotely, as well as via an alternative binary protocol - more security: ratelimit module, more check types in permissions module - scripting features: select operator in combination with revised AVPs, named flags and routes, more types of routes (onsend, timer, branch) - applications: presence has been greatly updated to keep pace with development in the IETF/SIMPLE (includes asynchronous subscription authorization, extended PIDF); web interface includes many new managibility features (multidomain, templates and multilanguage to name some). New scripting flexibility allows safely for features like individual ringing timers and direct access to SIP message elements. - new tools: Nils's script execution tree tool http://lists.iptel.org/pipermail/serdev/2006-October/008124.html and refurbished ser_ctl see the following links for some additional details: - http://www.iptel.org/ser/doc/010whatsnew - http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/NEWS?rev=HEAD Accompanying software, SEMS and SERWEB have been greatly overhauled as well. SEMS now features improved performance (single-threaded model now), service composition via python-IVR-scripting and B2BUA, adaptive playout buffer for better voice quality, and more. SERWEB features new multilevel administrative access system, multidomain support, per-domain customization, remote xml-rpc access, multilanguage support, and more. The new command-line utility, SER_CTL, provides also many great improvements over the previous tool. Interop ------- Shortly, we have been keeping the bar high through our participation in the IETF and SIPITs. SER implements latest interop changes from the IETF (STUN-bis, presence + XCAP, to name the most important ones) and both new features and legacy features were extensively and successfully tested middle of October in SIPit. Next Steps ----------- Keep in mind, this is a pre-release for testing purposes. It is now call on users to test the software hard and provide feedback. In the meantime, the codebase remains frozen and we only commit patches fixing various bugs. Once we get good confidence that Ottendorf has stabilized and is being adequately documented, we will declare the final release. No date has been determined yet. Also, we consider changing CVS to SVN. Again, and this is very important: the final release will be as good as the feedback we obtain for this pre-release. Please do spend your time sending solid feedback back to [serusers]. Roadmap ------- We keep rolling. We are still debating with our generous contributors what is the next possible good things to work on, the topics include but are not limited to as of now: - documentation. - databases are important. LDAP and BDB, DB-API revision. - interop: identity, TLS revision - advanced scripting (scripting engine, vars returns, AVPs) - uncountable improvements (futex, dev/rtc, tm/refcnts, tcp fd cache, .....) Trivia -------- The release name Ottendorf is coming from a place in Saxonian Switzerland, where many iptel fellows and their friends spent a weekend during this Indian summer. http://www.iptel.org/jan/gallery/saxonian_switzerland_2006/ http://www.iptel.org/~andi/2006-09-22-Ottendorf/ http://www.iptel.org/~vlada/saxon_switzerland/index.html http://www.ohlmeier.de/albums/ottendorf_2006/index.html Acknowledgments --------------- This grand achievement is result of extremely many folks who did lot of hard work. Out of folks affiliated with iptel, Andrei and Jan spent immense effort on moving SER forward, with great support from Hallik (command-line tool), Michal, Miklos, Nils, Tomas, Vaclav (presence), Vlada (new STUN). In this context, I would like to thank to our company for the great support. Recently, Bogdan Pintea greatly helped to make the upcoming release better and more stable under sponsorship of his company, iptego. transnexus provided osp support. SEMS has been greatly improved thanks to Stefan and Raphael. Credits for serweb go to Karel. Same importantly, SER has been immensely benefitting of contributors, to name in alphabetic order at least the 'top-of-the-iceberg' fellows: Bogdan Pintea, Cesc Santa, Greg Fausak, Greger Teigre, Hendrik Scholz, <http://tutpro.com/>Juha Heinänen, Klaus Darilion, Maxim Sobolev, Peter Nixon, Uli Abend. Great support on the mailing list has been provided by Greger V. Teigre, Java Rockx, Klaus Darilion, Zeus Ng, and more. Simply, this list is too short and incomplete to capture all the generous contributions that came in -- my appologies for everyone else I have missed. Warnings and Disclaimers ------------------------ This is a TESTING version. We are encouraging people to give it intense testing and provide us with feedback, so that we can properly assess what it takes to declare it final. The volume and number of changes is big, caution is advised. All GPL warnings and disclaimers apply. This is SER, which is not the derivate software known as openser. We have no ideas of in what extent and timeline derivate projects follow (or not) SER. ------------------------------------------------------------------------------- To Give it a Try ----------------- warning: this is just a 'jump-start' for black-belt users. It is not a replacement for documentation or a detailed step-by-step installation guide. It assumes knowledge of operating system, SIP, and SER. PREPARATION - make sure mysql is running. In the default configuration, SER assumes it to be running on localhost, if this does not hold, you need to change ser.cfg. - make sure python is installed (needed for new command-line utilities) SER INSTALLATION FROM CVS - Download from CVS/HEAD using instructions at http://www.iptel.org/downloads - Compile it: make include_modules="mysql xmlrpc" all - Install it the default way of your administrative privileges allow to: make include_modules="mysql xmlrpc" install - Download command-line tool (see more documentation under http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/serctl/README?rev=…) - cvs co serctl - Initiate database: ./scripts/mysql/ser_mysql.sh create START - Start it. Call: ser - Begin provisioning users with the new serctl (the capital word are just place holders) # ser_ctl domain add DOMAINNAME # ser_ctl user add USERNAME@DOMAINNAME -p PASSWORD # If you want to have Asserted Identity header for your user # - ser_attr add uid=UID asserted_id="PID" # If you want to have gateway support # - ser_db add attr_types name=gw_ip rich_type=string raw_type=2 \ description="The gateway IP for the default ser.cfg" default_flags=33 # - ser_attr add global gw_ip=GATEWAY-IP - more things you may wish to do: - familiarize yourself with current ser.cfg - send us feedback. -- Jiri Kuthan http://iptel.org/~jiri/ [View Less]

6 9

[Users] auth_radius - Initializing problem
by exploiz 10 Nov '06

10 Nov '06

Hello all, I am install openSER follow (http://www.openser.org/docs/openser-radius-1.0.x.html). I have problem when run openser, below is a debug. And when remove *_radius.so the program can work. what's going wrong Thank you for all advice. BR, exploiz -------------------------------- LOG --------------------------------------------------- textops - initializing 0(0) DEBUG: init_mod: xlog 0(0) XLOG: initializing ... 0(0) DEBUG: init_mod: acc 0(0) ACC - initializing 0(0) find_export: … [View More]found <load_tm> in module tm [/usr/local/lib/openser/modules/tm.so] 0(0) DEBUG: init_mod: auth 0(0) AUTH module - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/openser/modules/sl.so] 0(0) DEBUG: init_mod: auth_radius 0(0) auth_radius - Initializing 0(0) auth_radius: No `Cisco' vendor in Radius dictionary 0(0) find_export: found <bind_auth> in module auth [/usr/local/lib/openser/modules/auth.so] 0(0) ERROR: auth_radius: can't get code for the Digest-Response attribute 0(0) init_mod(): Error while initializing module auth_radius ERROR: error while initializing modules 0(0) XLOG: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(81840) shm_mem_destroy 0(81840) destroying the shared memory lock --------------------------------------------------------------------------- -------openser.cfg-------------------------------------------- .... loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" .... ---------------------------------------------------------------------------------------------- -- บุญรักษาทุกทั่น exploiz [View Less]

3 3

Re[2]: [Users] pstn-to-openser, 403 Forbiden
by Ion Minzu 09 Nov '06

09 Nov '06

Hello Norman, thanks, Norman after I have closed the domain authorisation (proxy_authorize), everything works. It looks for me too sophisticated to use permission and domain module. I will try another method. now I want to make restrictions for voip users, I mean who has the right to make a call to pstn and who has not. someone has some ideeas? Wednesday, November 8, 2006, 2:35:34 PM, you wrote: > Hello Ion, > Perhaps the "pstn" device is not responding to the authorization … [View More]> request. A SIP trace will show you if this is happening. > "ngrep -W byline port 5060" should be all you need to trace SIP messages > (if you're listening on eth1, then you might want to add "-d eth1" when > invoking ngrep). Older versions of ngrep did not have the "-W byline" > option. If this is your case, you probably should upgrade ngrep. > A possible solution to you problem might be to use the "permissions" > module and the "trusted" table. Then, instead of > (www_authorize/www_challenge or in addition to, depending on your > particular situation) you might want to use the "allow_trusted()" > command to authenticate incoming INVITEs. > Below is a little example > <snip> > if (!is_from_local() && !allow_trusted()) { > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > exit; > } else if (!check_from()) { > sl_send_reply("403", "Use From=ID"); > exit; > }; > }; > </snip> > Regards, > Norm > Ion Minzu wrote: >> Hello , >> >> I have connected openser with pstn through cisco. when I make a >> call from voip network to pstn it's ok. >> but from pstn to voip I have a problem:openser answers 403 forbiden. >> in openser I do the authorisation on mysql, I have disabled authorisation on sip >> gateway: >> >> if (src_ip!=X.X.X.X) { >> if (!www_authorize("DOMAIN.COM","subscriber")) { >> www_challenge("DOMAIN.COM","0"); >> exit; >> } >> }; >> >> What is the problem? >> >> X.X.X.X is cisco >> >> U X.X.X.X:54177 -> 172.17.6.2:5060 >> INVITE sip:820022@172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP >> X.X.X.X:5060..From: >> <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.1 >> 7.6.2>..Date: Wed, 08 Nov 2006 11:03:14 GMT..Call-ID: >> 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..Supported: >> timer,100rel..Min-SE: 1800..Cisco-Guid >> : 2422905184-1850675675-2419190351-3917593119..User-Agent: >> Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, >> ACK, PRACK, COMET, REFER, SUBS >> CRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: >> 6..Remote-Party-ID: >> <sip:022250699@X.X.X.X>;party=calling;screen=yes;privacy=off..Timestamp: >> 116 >> 2983794..Contact: <sip:022250699@X.X.X.X:5060>..Expires: >> 180..Allow-Events: telephone-event..Content-Type: >> application/sdp..Content-Length: 235....v= >> 0..o=CiscoSystemsSIP-GW-UserAgent 1226 5023 IN IP4 X.X.X.X..s=SIP >> Call..c=IN IP4 X.X.X.X..t=0 0..m=audio 16642 RTP/AVP 18 19..c=IN IP4 >> X.X.X.X..a=rtpmap:18 G729/8000..a=fmtp:18 >> annexb=no..a=rtpmap:19 CN/8000..a=ptime:20.. >> # >> U 172.17.6.2:5060 -> X.X.X.X:5060 >> SIP/2.0 403 Use From=ID..Via: SIP/2.0/UDP X.X.X.X:5060..From: >> <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: >> >> <sip:820022@172.17.6.2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Call-ID: >> 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..CSeq: 101 >> INVITE..Server: OpenSer (1.1.0-tls (x86_64/linux))..C >> ontent-Length: 0..Warning: 392 172.17.6.2:5060 "Noisy >> feedback tells: pid=32240 req_src_ip=X.X.X.X req_src_port=54177 >> in_uri=sip:820022@172.17.6.2:5 >> 060 out_uri=sip:820022@172.17.6.2:5060 via_cnt==1".... >> # >> U X.X.X.X:54177 -> 172.17.6.2:5060 >> ACK sip:820022@172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP >> X.X.X.X:5060..From: >> <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.17.6 >> .2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Date: Wed, 08 Nov >> 2006 11:03:14 GMT..Call-ID: >> 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..Max-Forward >> s: 6..Content-Length: 0..CSeq: 101 ACK.... >> >> >> Best regards, >> Ion Minzu, >> Specialist Tehnologii Informationale, >> Administrator de sistem al Centrului de certificare, >> Administrator VoIP, >> I.S."Centrul de Telecomunicatii Speciale", >> tel:250-517 (office), 069501208 (mob), 382869185 (ICQ) >> mailto:ion.minzu@cts.md >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> -- Best regards, Ion mailto:ion.minzu@cts.md [View Less]

3 2

[Users] OpenSER Summit - BoF slides
by Daniel-Constantin Mierla 09 Nov '06

09 Nov '06

Hello, the slides presented Tuesday, during BoF, are now available online: http://www.openser.org/events/2006-OpenSER-Summit/slides/ Those from the Wednesday, need to be collected in the final version and will show up there as they arrive. Also, we are collecting photos/videos and any other story about the summit, please sent them to us (team(a)openser.org) to make them available. For the pictures/videos will be good if you can make them available for download somewhere, rather than … [View More]

1 0

[Serusers] Problem in using ser_ctl
by Ajay Kolambekar 09 Nov '06

09 Nov '06

Hi, I have installed SER on Redhat Linux 9 machine. I want to use ser_ctl to add/view users or domains. When I give command set_ctl user show I am getting an error like ser_ctl: /root/Python-2.5/lib/python2.5/site-packages/_mysql.so: undefined symbol: mysql_set_character_set: ImportError'> Does anybody know how to solve this problem? Thanks in advance, Ajay DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems … [View More]

1 0

[Users] openser + presence module (presence.so) is not compiling
by prateek kapoor 09 Nov '06

09 Nov '06

Hi All, I have installed openser version from 1.1.0 debian package installer and it is working fine. But when I am trying to include the presence module (presence.so), it is not there. When I am trying to recompile it from source tarball I am getting a lot of dependency errors. So can anyone tell me from where I can get the 'presence.so' ? I can see a 'pa' (presence agent) module. I would also like to know what is the difference between 'pa' and 'presence' modules ? Any help will be really … [View More]

2 1

[serusers]: Presence Module
by Kamal.Mann＠t-systems.com 09 Nov '06

09 Nov '06

Hi All I need to use SER as a stand alone presence server. How to install/configure ser as presence server? Any kind of related links, documents etc. Do I need some external libraries too for compiling SER code to run it as presence server or any binary package for all? Is their any proper document is already present? Thanks in anticipation Kamal Mann

2 2

[Users] usrloc-cl
by Mark Price 09 Nov '06

09 Nov '06

Hi, It seems that it has been about a year since usrloc-cl was mentioned on this list: http://www.openser.org/pipermail/users/2005-October/001424.html What is the status of usrloc-cl on openser? Has it been integrated into openser's usrloc as the referenced email suggests is a good idea? Is the usrloc-cl functionality available in the openser codebase? Thanks, Mark Price

2 1

[Users] openser summit
by Klaus Darilion 08 Nov '06

08 Nov '06

Hi guys! It was a pleasure for me to attend the openser summit and meat you all - it's funny to map faces to names only known from the mailing list. Although my presentation exceeded the suggested timeframe I could not explain all things in detail. Thus, If there are questions left, please feel free to ask me - the mailinglist is preferred. regards klaus PS: The VON t-mobile login works also at airport Tegel :-)

1 0

[Serusers] Problems with CANCEL and INVITE
by Benito Caracuel 08 Nov '06

08 Nov '06

Hello all, I have a problem when I try to cancel an invite, because I have modified the invite method. In my ser.cfg: if ((method=="INVITE")&&(!lookup("location"))) { exec_msg("printenv SRCIP; /bin/bash /usr/local/ser_snapshot/sbin/sms"); sl_send_reply("100", "Trying"); exec_msg("sleep 10"); if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("… [View More]

1 0

[Serusers] Accounting mismatch
by inge 08 Nov '06

08 Nov '06

Hi all, When my users make a call, the domain field in "acc" table from SER database have a constant FQDN something like aa.mydomain.ext. But there is some case where the domain field contain IP of SER, not FQDN. The problem is that when I consult accounting page I only see the call where the domain field contains aa.mydomain.ext. Is it possible to see both call : FQDN/IP ? Thank for your help. Best regards.

1 0

[Serusers] Cancelling an INVITE
by Benito Caracuel 08 Nov '06

08 Nov '06

Hello all, I have a problem when I try to cancel an invite, because I have modified the invite method. I'm going to explain the idea behind the problem, which is related to what we call "OffLine Invitation". An "Offline Invitation" consists of the SER set up as to DON'T send a "404 Not Found" when the Invitated party isn't registered. What is done is to keep the SER sending "100 Trying" for a while, and to send meanwhile a SMS to the Invitated -but not registered- party. That SMS is … [View More]understood by the Application that's running at the Invited party as a "Register Request", so the Invitated party goes for a Register. That works OK, but the problem comes up when you try to CANCEL the Invitation request. In that case, as the Invitated party isn't still registered, it cannot take care of the CANCEL's lifecicle, so SER has to deal with it, "emulating" that it is the Invitated party. As to do this, we assume that no race condition is produced, and neither a Final response has been sent by the Invitated party. So what is needed is, according to SIP protocol, to answer to the CANCEL with "200 OK" and to the original INVITE with a "487 Request Terminated". The problem is obviously doing the INVITE - CANCEL match at the time the CANCEL gets to SER. We know there are scripts/procedures and tables in the SER architecture that can be used to do other tasks (as it is the "saving" of REGISTER packet onto a DB table), but, to be honest, we are a bit confused about if it can be done, or if it should need SER ad-hoc programming, or if there is a way to use an specific SER DB Table to do it... I post along my SER.CFG file, and an idea about how we were thinking that it could be solved (that is, a big picture; what we need indeed is to know if it is a good idea (having in mind SER architecture and scripting capabilities) or it cannot be done. Please, if it can be done, we'd like to hear a little more about what tables can be used, or if it's needed to create another new one, or at least, a clue to solve it. In my ser.cfg: if ((method=="INVITE")&&(!lookup("location"))) { exec_msg("printenv SRCIP; /bin/bash /usr/local/ser_snapshot/sbin/sms"); sl_send_reply("100", "Trying"); exec_msg("sleep 10"); if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("404", "Not on-line"); break; } else { t_relay(); break; } }; If I send an INVITE to user2, who is off-line (not registered), SER sends a SMS to this user2 and meanwhile, sends TRYINGS to me, until user2 becomes REGISTERED (the sent SMS makes the Invited party Register, so it can receive the Invite, as It was told before). I have a problem, because If I send a CANCEL, SER still sends TRYINGS to me and SER doesn`t send "487 Request Terminated" and SER doesn´t associate CANCEL and INVITE. How can I solve this? any idea? I think that maybe, If I use a mysql table to save CSEQ of CANCEL, and before SER sends TRYINGS (I mean the next TRYINGs following the first), SER checks if CSEQ is into table, but I don´t know if SER can save CANCEL information in a mysql table. I know that there is a command "save" to save REGISTER packet into location table. Is there any similar command to save CSEQ of CANCEL (and/or other, as Call-Id, Branch, Request-Uri and To) in one table? Could someone help me, please? We are in a hurry with our customer, so fast key to a solution will be really appreciated. Thank you very much. _________________________________________________________________ Llama a tus amigos de PC a PC: ¡Es GRATIS! http://get.live.com/messenger/overview [View Less]

1 0

[Users] Not Hung Up when i Keep the Server in Router_firewall( NETGEAR)
by raviprakash sunkara 08 Nov '06

08 Nov '06

Hello User, A small Question , When I My Sip Server without any firewall_ Router .. Hung up is working Fine,,,,,, But I Keep Sip Server with in a Firewall_router, it not Hung upping , When i saw logs from X-lite in Bye method it Changing the uri into Private IP, But should it public IP( THis is not happening ) Please help me............. -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)… [View More]

2 1

[Serusers] mediaproxy and ser ottendorf incompatible right now
by samuel 08 Nov '06

08 Nov '06

I was trying to use mediaproxy module with SER ottendorf and got next error: error: mediaproxy/mod_init(): can't find is_from_local and/or is_uri_host_local functions. Check if domain.so is loaded domain.so is effectively loaded BUT these functions have been replaced by is_local,lookup_domain, and get_did in new SER version Is AG Projects taking care of this issue?!? Can someone shed dome light on this issue!??!?!? Thanks a lot, Samuel.

1 0

Re: [Users] TM : retransmission timers
by Rao Ramaratnamma 08 Nov '06

08 Nov '06

Greg, The email from you and others is giving me the type of most direct fair insights I am looking for . You know, just some practical "what is all this about" type of things for my selforientation. Just that openser link you are attaching unfortunately looks too much like openser marketing material. It is saying that ser was discontinued but actually the information I'm getting from others telling me that it has some features which are very important and openser do not have. With all … [View More]respect, it is understandable that openser people write nicely about themselves and less nicely about others but it remains so confusing that I do not have confidence to move with this particular material. The openser info just appears too little credible for me. I think I am beginning to get the bottom line and I hope I will create my mind soon and if people have other information for me pls send it. Especially, objective technical information including this timer discussion is helping me. You know, better understanding what is the difference in "what they think is important". That would surely help me beyound the filosophical "cats or dogs, who is a better friend of man" information aqusition phase people surely like to be engaged in :-) Thank you all guys for keeping helping me a lot, rr ----- Original Message ---- From: Greg Fausak <lgfausak(a)gmail.com> To: Rao Ramaratnamma <raramarat(a)yahoo.com> Cc: Christian Schlatter <cs(a)unc.edu>; users(a)openser.org Sent: Wednesday, November 8, 2006 5:21:41 AM Subject: Re: [Users] TM : retransmission timers You can go to openser.org's website and read about howhow openser and ser are related. http://www.openser.org/index.php?option=com_content&task=view&id=40&Itemid=… They are now two separate bodies of code, maintained by two different groups. Each one has gone off and implemented and support what they think is important. Both projects are open source projects, anyone is certainly welcome to contribute to either. -g On Nov 7, 2006, at 4:15 PM, Rao Ramaratnamma wrote: the ser ottendorf announcement does mention improved timers. Cannot openser include this feature too and cannot I merge ser with openser for good timers? I am still trying to understand the difference between ser and openser but standart compliance seems to be very important matter! Cannot people provide me with some hints? I am sure that I am not the only who is asking the difference between ser and openser. ser documentation does not appear uptodate, but the software as sannounced appears impressive. I have already asked this question but did not receive any answer. thank you in advance! rr ----- Original Message ---- From: Christian Schlatter <cs(a)unc.edu> To: users(a)openser.org Sent: Tuesday, November 7, 2006 10:52:56 PM Subject: Re: [Users] TM : retransmission timers Greg Fausak wrote: > Hello, > > I believe this is a well known bug. > Granularity of timers is 1 second. So, if you sign up for a timer to > be fired in 1 second it will happen anywhere between 0 seconds and 1 > second. > 2 seconds will happen between 1 and 2 seconds. I usually set up my > timers to be 2, 2, 4, 8. There are VOIP providers that are pretty > sticky about > the first 500ms. If you are using one of them you're out of luck. Yes, there is a timer process that wakes up every second to perform retransmissions. I was actually quite surprised that OpenSER, which is known to be very standards compliant, does not follow the RFC 3261 retransmission timeouts. On the other hand, the RFC 3261 timeout values are just suggestions and standards compliant SIP UA must accept shorter timeouts. Still it would be nice if OpenSER would support sub second timers, this would allow for shorter fail-over times. Christian > > I believe SER has made timer changes to support more exact timer > intervals. They are a completely different camp, with a different feature > set (although they share the same roots). > > -g > > > On 11/7/06, Jean-François SMIGIELSKI <jf-smig(a)ibelgique.com> wrote: >> Hello, >> >> I made strange observations about the intervals between >> retransmissions with the TM module. >> In my experiments, I used the default parameters for the TM module >> timers, and I sent an INVITE that cannot receive answers (it has a >> well known R-URI pattern that is forwarded to a place and port that >> nobody listen). >> >> When reading RFC3261, I expected to see intervals between >> retransmissions of |500ms|1s|2s|4s|8s|16s|. 7 transmissions, during 32s. >> >> But with OpenSER, (I have tested with the debian package 1.1.0-5 on a >> debian etch, and the cvs sources for 1.1.0 or 1.0.1compiled by >> myself), I can see intervals like <500ms, 2s, 4s, 4s,4s, ... until 26s >> are spent (9 sendings). The first interval is sometomes very short >> (40ms). >> >> Altough I like the sequence of 4s separated transmissions, I do not >> know why the first interval is so short, and why there is no sending >> after 1s. >> >> Did anybody observed such behaviours? Are they normal? >> >> Thanks in advance! >> >> JF Smigielski. >> >> >> ________________________________________________________________________ >> iBELGIQUE, exprimez-vous ! >> http://web.ibelgique.com/ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users [View Less]

1 0

[Users] TM : retransmission timers
by Jean-François SMIGIELSKI 08 Nov '06

08 Nov '06

Hello, I made strange observations about the intervals between retransmissions with the TM module. In my experiments, I used the default parameters for the TM module timers, and I sent an INVITE that cannot receive answers (it has a well known R-URI pattern that is forwarded to a place and port that nobody listen). When reading RFC3261, I expected to see intervals between retransmissions of |500ms|1s|2s|4s|8s|16s|. 7 transmissions, during 32s. But with OpenSER, (I have tested with the … [View More]

5 6

[Serusers] grandstream configuration -- help
by Ravi ***** 08 Nov '06

08 Nov '06

Hi users , I am using ser-0.96 , and every thing seems to work fine . but when I make calls from grandstream ATA .. its is working properly but the problem is when the other party hung the phone SER cannot send BYE packets to grandstream NAT ip . It is sending to grandstream private-ip which inturns delayed for acct.stop packet unless grandstream hung the phone x-lite, is O.K and its sending to SER as tel.no@Nat-ip-address but grandstream is sending to SER as tel.no@… [View More]

2 1

[Serusers] Problem with CANCEL messages and Cisco 7905
by Kristian Larsson 08 Nov '06

08 Nov '06

Hey! This is my first time posting to the list, although I'm a long time reader :) I have searched the archives as well as the internet and found no solution or answer to this so far. Perhaps my Google Kung Fu isn't what it should be. Anyhow, I have a cisco 7905 phone working together with a SER machine. When dialing to the 7905 phone from another phone and hanging up on the caller phone before answering the 7905, ie CANCEL before the call has actually been setup, I get a "SIP/2.0 481 Call … [View More]

2 2

[Users] pstn-to-openser, 403 Forbiden
by Ion Minzu 08 Nov '06

08 Nov '06

Hello , I have connected openser with pstn through cisco. when I make a call from voip network to pstn it's ok. but from pstn to voip I have a problem:openser answers 403 forbiden. in openser I do the authorisation on mysql, I have disabled authorisation on sip gateway: if (src_ip!=X.X.X.X) { if (!www_authorize("DOMAIN.COM","subscriber")) { www_challenge("DOMAIN.COM","0"); exit; } }; What is the problem? X.X.X.X is cisco U X.X.X.X:54177 -> 172.… [View More]17.6.2:5060 INVITE sip:820022@172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP X.X.X.X:5060..From: <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.1 7.6.2>..Date: Wed, 08 Nov 2006 11:03:14 GMT..Call-ID: 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..Supported: timer,100rel..Min-SE: 1800..Cisco-Guid : 2422905184-1850675675-2419190351-3917593119..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBS CRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 6..Remote-Party-ID: <sip:022250699@X.X.X.X>;party=calling;screen=yes;privacy=off..Timestamp: 116 2983794..Contact: <sip:022250699@X.X.X.X:5060>..Expires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..Content-Length: 235....v= 0..o=CiscoSystemsSIP-GW-UserAgent 1226 5023 IN IP4 X.X.X.X..s=SIP Call..c=IN IP4 X.X.X.X..t=0 0..m=audio 16642 RTP/AVP 18 19..c=IN IP4 X.X.X.X..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:19 CN/8000..a=ptime:20.. # U 172.17.6.2:5060 -> X.X.X.X:5060 SIP/2.0 403 Use From=ID..Via: SIP/2.0/UDP X.X.X.X:5060..From: <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.17.6.2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Call-ID: 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..CSeq: 101 INVITE..Server: OpenSer (1.1.0-tls (x86_64/linux))..C ontent-Length: 0..Warning: 392 172.17.6.2:5060 "Noisy feedback tells: pid=32240 req_src_ip=X.X.X.X req_src_port=54177 in_uri=sip:820022@172.17.6.2:5 060 out_uri=sip:820022@172.17.6.2:5060 via_cnt==1".... # U X.X.X.X:54177 -> 172.17.6.2:5060 ACK sip:820022@172.17.6.2:5060 SIP/2.0..Via: SIP/2.0/UDP X.X.X.X:5060..From: <sip:022250699@X.X.X.X>;tag=1A0FBC30-1472..To: <sip:820022@172.17.6 .2>;tag=329cfeaa6ded039da25ff8cbb8668bd2.13ec..Date: Wed, 08 Nov 2006 11:03:14 GMT..Call-ID: 906DA628-6E4F11DB-9034EA4F-E981BA1F@X.X.X.X..Max-Forward s: 6..Content-Length: 0..CSeq: 101 ACK.... Best regards, Ion Minzu, Specialist Tehnologii Informationale, Administrator de sistem al Centrului de certificare, Administrator VoIP, I.S."Centrul de Telecomunicatii Speciale", tel:250-517 (office), 069501208 (mob), 382869185 (ICQ) mailto:ion.minzu@cts.md [View Less]

3 2

[Users] OpenSER + SEMS + Nat?
by Brandon Armstead 08 Nov '06

08 Nov '06

Hello, I currently have an openser server running on port 5060, and a second that handles voicemail / stateful transaction stuff on port 5070. So, I am having proxy A forward to proxy B to do the handling which are both on the same server, although there seems to be a problem with NAT'ing, when I call 'voicemail' or something I can not hear anything. It seems to be because it is sending the RTP data to the wrong location? It should be sending it back to public-ip-addr-1 when it is … [View More]sending to an internal ip address 192.168.1.* Here is the log from sems: 101 44811:1466724767 <sip:public-ip-addr-1:5060;nat=yes;ftag=dd9bbf561d35ee06o0;lr=on> sip:public-ip-addr-1:5060;nat=yes;ftag=dd9bbf561d35ee06o0;lr=on P-MsgFlags: 0 UA: Linksys/PAP2-3.1.9(LSa) P-Email-Address: test(a)test.com . v=0 o=- 9200357 9200357 IN IP4 192.168.1.241 s=- c=IN IP4 192.168.1.241 t=0 0 m=audio 16420 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv > (7856) DEBUG: execute (AmInterfaceHandler.cpp:99): req.method = <INVITE> (7856) DEBUG: execute (AmInterfaceHandler.cpp:100): req.user = <115> (7856) DEBUG: execute (AmInterfaceHandler.cpp:101): req.domain = <public-ip-addr-1> (7856) DEBUG: execute (AmInterfaceHandler.cpp:102): req.dstip = <public-ip-addr-1> (7856) DEBUG: execute (AmInterfaceHandler.cpp:103): req.port = <5070> (7856) DEBUG: execute (AmInterfaceHandler.cpp:104): req.r_uri = < sip:115@public-ip-addr-1:5070> (7856) DEBUG: execute (AmInterfaceHandler.cpp:105): req.from_uri = < sip:1000@192.168.1.241:5060> (7856) DEBUG: execute (AmInterfaceHandler.cpp:106): req.from = <Brandon1 < sip:1000@test.mydomain.com>> (7856) DEBUG: execute (AmInterfaceHandler.cpp:107): req.to = << sip:115@test.mydomain.com>> (7856) DEBUG: execute (AmInterfaceHandler.cpp:108): req.callid = < 81e2d596-978fccc6(a)192.168.1.241> (7856) DEBUG: execute (AmInterfaceHandler.cpp:109): req.from_tag = <dd9bbf561d35ee06o0> (7856) DEBUG: execute (AmInterfaceHandler.cpp:110): req.to_tag = <> (7856) DEBUG: execute (AmInterfaceHandler.cpp:113): cseq_str = <101> (7856) DEBUG: execute (AmInterfaceHandler.cpp:119): cseq = <101>(101) (7856) DEBUG: execute (AmInterfaceHandler.cpp:121): req.key = <44811:1466724767> (7856) DEBUG: execute (AmInterfaceHandler.cpp:122): req.route = <<sip:public-ip-addr-1:5060;nat=yes;ftag=dd9bbf561d35ee06o0;lr=on>> (7856) DEBUG: execute (AmInterfaceHandler.cpp:123): req.next_hop = <sip:public-ip-addr-1:5060;nat=yes;ftag=dd9bbf561d35ee06o0;lr=on> (7856) DEBUG: execute (AmInterfaceHandler.cpp:128): hdrs = <P-MsgFlags: 0 UA: Linksys/PAP2-3.1.9(LSa) P-Email-Address: test(a)test.com > (7856) DEBUG: execute (AmInterfaceHandler.cpp:133): body = <v=0 o=- 9200357 9200357 IN IP4 192.168.1.241 s=- c=IN IP4 192.168.1.241 t=0 0 m=audio 16420 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv > (7856) DEBUG: execute (AmInterfaceHandler.cpp:144): Request OK: dispatch it! (7856) DEBUG: postEvent (AmEventQueue.cpp:48): AmEventQueue: trying to post event (7856) DEBUG: postEvent (AmEventQueue.cpp:56): AmEventQueue: event posted (7856) DEBUG: processEvents (AmEventQueue.cpp:69): before processing event (7856) DEBUG: process (AmSession.cpp:421): AmSession::process (7856) DEBUG: process (AmSession.cpp:425): Session received SIP Event (7856) DEBUG: onSipRequest (AmSession.cpp:472): onSipRequest: method = INVITE (7856) DEBUG: parse_sdp_media (AmSdp.cpp:632): next_line=<a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv > (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:0 PCMU/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=0; enc=PCMU; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:2 G726-32/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=2; enc=G726-32; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:4 G723/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=4; enc=G723; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:8 PCMA/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=8; enc=PCMA; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:18 G729a/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=18; enc=G729a; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:96 G726-40/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=96; enc=G726-40; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:97 G726-24/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=97; enc=G726-24; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:98 G726-16/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=98; enc=G726-16; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:100 NSE/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=100; enc=NSE; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:101 telephone-event/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=101; enc=telephone-event; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=fmtp:101 0-15 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=ptime:30 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:740): unknown attribute definition 'ptime:30' (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=sendrecv (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:745): flag definition is not yet supported (sendrecv) (7856) DEBUG: parse_sdp_media (AmSdp.cpp:651): ret=0; next_line=<> (7856) DEBUG: getCompatiblePayload (AmSdp.cpp:341): payload found: 0/PCMU/8000 (7856) DEBUG: getCompatiblePayload (AmSdp.cpp:344): using global address: 192.168.1.241 (7856) DEBUG: negotiate (AmSession.cpp:247): remote party supports telephone events (pt=101) (7856) DEBUG: setRAddr (AmRtpStream.cpp:346): RTP remote address set to 192.168.1.241:16420 (7856) DEBUG: init (AmRtpAudio.cpp:146): AmRtpAudio::init(...) (7856) DEBUG: acceptAudio (AmSession.cpp:607): Sending Rtp data to 192.168.1.241/16420 (7856) DEBUG: updateStatusReply (AmSipDialog.cpp:86): reply: transaction found! (7856) DEBUG: updateStatusReply (AmSipDialog.cpp:125): req.method = INVITE; t.method = INVITE (7856) DEBUG: init (AmCtrlInterface.cpp:231): AmUnixCtrlInterface::init @ /tmp/5D08F0A1-4552226600080041-08108600 (7856) DEBUG: write_to_socket (AmUtils.cpp:588): sending: <:t_reply:/tmp/5D08F0A1-4552226600080041-08108600 200 OK 44811:1466724766 193812C5-45522256000F0C97-08103000 Contact: <sip:115@public-ip-addr-1:5070> Content-Type: application/sdp . v=0 o=username 0 0 IN IP4 public-ip-addr-1 s=session c=IN IP4 public-ip-addr-1 t=0 0 m=audio 10022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 . > (7856) DEBUG: write_to_socket (AmUtils.cpp:614): write to unix socket: completed (7856) DEBUG: cacheMsg (AmCtrlInterface.cpp:186): recv-ed: <200 Succeeded> (7856) DEBUG: processEvents (AmEventQueue.cpp:71): event processed (7856) DEBUG: processEvents (AmEventQueue.cpp:69): before processing event (7856) DEBUG: process (AmSession.cpp:421): AmSession::process (7856) DEBUG: process (AmSession.cpp:425): Session received SIP Event (7856) DEBUG: onSipRequest (AmSession.cpp:472): onSipRequest: method = INVITE (7856) DEBUG: parse_sdp_media (AmSdp.cpp:632): next_line=<a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv > (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:0 PCMU/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=0; enc=PCMU; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:2 G726-32/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=2; enc=G726-32; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:4 G723/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=4; enc=G723; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:8 PCMA/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=8; enc=PCMA; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:18 G729a/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=18; enc=G729a; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:96 G726-40/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=96; enc=G726-40; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:97 G726-24/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=97; enc=G726-24; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:98 G726-16/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=98; enc=G726-16; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:100 NSE/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=100; enc=NSE; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=rtpmap:101 telephone-event/8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:693): sdp attribute: pt=101; enc=telephone-event; cr=8000 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=fmtp:101 0-15 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=ptime:30 (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:740): unknown attribute definition 'ptime:30' (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:658): parse_sdp_attribute: s=sendrecv (7856) DEBUG: parse_sdp_attribute (AmSdp.cpp:745): flag definition is not yet supported (sendrecv) (7856) DEBUG: parse_sdp_media (AmSdp.cpp:651): ret=0; next_line=<> (7856) DEBUG: getCompatiblePayload (AmSdp.cpp:341): payload found: 0/PCMU/8000 (7856) DEBUG: getCompatiblePayload (AmSdp.cpp:344): using global address: 192.168.1.241 (7856) DEBUG: negotiate (AmSession.cpp:247): remote party supports telephone events (pt=101) (7856) DEBUG: setRAddr (AmRtpStream.cpp:346): RTP remote address set to 192.168.1.241:16420 (7856) DEBUG: init (AmRtpAudio.cpp:146): AmRtpAudio::init(...) (7856) DEBUG: acceptAudio (AmSession.cpp:607): Sending Rtp data to 192.168.1.241/16420 (7856) DEBUG: updateStatusReply (AmSipDialog.cpp:86): reply: transaction found! (7856) DEBUG: updateStatusReply (AmSipDialog.cpp:125): req.method = INVITE; t.method = INVITE (7856) DEBUG: init (AmCtrlInterface.cpp:231): AmUnixCtrlInterface::init @ /tmp/75EE09BE-4552226600080514-08108600 (7856) DEBUG: write_to_socket (AmUtils.cpp:588): sending: <:t_reply:/tmp/75EE09BE-4552226600080514-08108600 200 OK 44811:1466724767 193812C5-45522256000F0C97-08103000 Contact: <sip:115@public-ip-addr-1:5070> Content-Type: application/sdp . v=0 o=username 0 0 IN IP4 public-ip-addr-1 s=session c=IN IP4 public-ip-addr-1 t=0 0 m=audio 10022 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 . > (7856) DEBUG: write_to_socket (AmUtils.cpp:614): write to unix socket: completed (7856) DEBUG: cacheMsg (AmCtrlInterface.cpp:186): recv-ed: <200 Succeeded> (7856) DEBUG: processEvents (AmEventQueue.cpp:71): event processed (7856) DEBUG: run (AmSession.cpp:292): 81e2d596-978fccc6(a)192.168.1.241 dlg.getUACTransPending() = 0 Here is my config for "proxy B" the transaction handling with SEMS that "proxy A" sends to: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # ser for sems configuration # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5070 children=4 unix_sock="/tmp/openser_sems_sock" # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "avpops.so" # ----------------- setting module-specific parameters --------------- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("avpops", "avp_aliases", "email=i:67") modparam("tm", "pass_provisional_replies", 1) # appends for INVITE to voicemail modparam("tm", "tw_append", "voicemail_headers:UA=$hdr(User-Agent);P-Email-Address=$avp(s:email)") # appends for dtmf per INFO modparam("tm", "tw_append", "info_append:Content-Length=$hdr(Content-Length);Content-Type=$hdr(Content-Type);$rb") # 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"); return; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); return; }; if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); return; } # 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 record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing # make transaction if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if (!t_relay()) { sl_reply_error(); } return; }; if (!t_newtran()){ log("could not create transaction\n"); sl_send_reply("500","could not create transaction"); return; } # actively absorb ACKs if (method == "ACK") { t_relay(); return; } # pass INFO to SEMS if (method=="INFO") { if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ log("could not contact sems\n"); t_reply("500","could not contact media server"); } } if (uri =~ "sip:101.*@") { if (!t_write_unix("/tmp/sems_sock","myapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:102.*@") { if (!t_write_unix("/tmp/sems_sock","myconfigurableapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:103.*@") { if (!t_write_unix("/tmp/sems_sock","myannounceapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:104.*@") { if (!t_write_unix("/tmp/sems_sock","myjukebox")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:105.*@") { if (!t_write_unix("/tmp/sems_sock","mycc")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:106.*@") { if (!t_write_unix("/tmp/sems_sock","ivr_announce")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); return; } return; } ############ default sems apps if (uri =~ "sip:110.*@") { if (!t_write_unix("/tmp/sems_sock","echo")){ log("could not contact echo\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:111.*@") { if (!t_write_unix("/tmp/sems_sock","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:112.*@") { if (!t_write_unix("/tmp/sems_sock","conference")){ log("could not contact conference\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:113.*@") { if (!t_write_unix("/tmp/sems_sock","mailbox")){ log("could not contact mailbox\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:114.*@") { if (!t_write_unix("/tmp/sems_sock","early_announce")){ log("could not contact early_announce\n"); t_reply("500","could not contact media server"); return; } return; } if (uri =~ "sip:115.*@") { # load email avp with some email address avp_write("test(a)test.com","$avp(s:email)"); # use voicemail_headers append to pass it to sems if (!t_write_unix("/tmp/sems_sock","voicemail/voicemail_headers")){ log("could not contact voicemail\n"); t_reply("500","could not contact media server"); return; } return; } t_reply("404","Not found"); } Thank you, any and all help is appreciated, thanks again! -- Brandon Armstead [View Less]

1 0

[Serusers] Using scripts with exec module
by Prettl, Kathrin 08 Nov '06

08 Nov '06

I have written a shell script which has a return value of either 0 or 1. Depending on the outcome of this script I want SER to react. I know I have to use the exec module for this but I don't know how. Can anyone help? Thanks, Kathrin

3 3

Fw: [Users] TM : retransmission timers
by Rao Ramaratnamma 08 Nov '06

08 Nov '06

sorry for reposting -- I think this question belongs to both mailing list. I am really stuck with this. rr ----- Forwarded Message ---- From: Rao Ramaratnamma <raramarat(a)yahoo.com> To: Christian Schlatter <cs(a)unc.edu>; users(a)openser.org Sent: Tuesday, November 7, 2006 11:15:27 PM Subject: Re: [Users] TM : retransmission timers the ser ottendorf announcement does mention improved timers. Cannot openser include this feature too and cannot I merge ser with openser for good … [View More]timers? I am still trying to understand the difference between ser and openser but standart compliance seems to be very important matter! Cannot people provide me with some hints? I am sure that I am not the only who is asking the difference between ser and openser. ser documentation does not appear uptodate, but the software as sannounced appears impressive. I have already asked this question but did not receive any answer. thank you in advance! rr ----- Original Message ---- From: Christian Schlatter <cs(a)unc.edu> To: users(a)openser.org Sent: Tuesday, November 7, 2006 10:52:56 PM Subject: Re: [Users] TM : retransmission timers Greg Fausak wrote: > Hello, > > I believe this is a well known bug. > Granularity of timers is 1 second. So, if you sign up for a timer to > be fired in 1 second it will happen anywhere between 0 seconds and 1 > second. > 2 seconds will happen between 1 and 2 seconds. I usually set up my > timers to be 2, 2, 4, 8. There are VOIP providers that are pretty > sticky about > the first 500ms. If you are using one of them you're out of luck. Yes, there is a timer process that wakes up every second to perform retransmissions. I was actually quite surprised that OpenSER, which is known to be very standards compliant, does not follow the RFC 3261 retransmission timeouts. On the other hand, the RFC 3261 timeout values are just suggestions and standards compliant SIP UA must accept shorter timeouts. Still it would be nice if OpenSER would support sub second timers, this would allow for shorter fail-over times. Christian > > I believe SER has made timer changes to support more exact timer > intervals. They are a completely different camp, with a different feature > set (although they share the same roots). > > -g > > > On 11/7/06, Jean-François SMIGIELSKI <jf-smig(a)ibelgique.com> wrote: >> Hello, >> >> I made strange observations about the intervals between >> retransmissions with the TM module. >> In my experiments, I used the default parameters for the TM module >> timers, and I sent an INVITE that cannot receive answers (it has a >> well known R-URI pattern that is forwarded to a place and port that >> nobody listen). >> >> When reading RFC3261, I expected to see intervals between >> retransmissions of |500ms|1s|2s|4s|8s|16s|. 7 transmissions, during 32s. >> >> But with OpenSER, (I have tested with the debian package 1.1.0-5 on a >> debian etch, and the cvs sources for 1.1.0 or 1.0.1compiled by >> myself), I can see intervals like <500ms, 2s, 4s, 4s,4s, ... until 26s >> are spent (9 sendings). The first interval is sometomes very short >> (40ms). >> >> Altough I like the sequence of 4s separated transmissions, I do not >> know why the first interval is so short, and why there is no sending >> after 1s. >> >> Did anybody observed such behaviours? Are they normal? >> >> Thanks in advance! >> >> JF Smigielski. >> >> >> ________________________________________________________________________ >> iBELGIQUE, exprimez-vous ! >> http://web.ibelgique.com/ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users [View Less]

7 12

[Serusers] RE: XCAP server
by Jayantheesh Srimushnam - TLS, Chennai 08 Nov '06

08 Nov '06

Hi, Is it possible to run XCAP server as a standalone? I didn't see any info about this in the document. --Jayantheesh _____ From: Jayantheesh Srimushnam - TLS, Chennai Sent: Tuesday, November 07, 2006 8:44 PM To: 'serusers(a)iptel.org' Cc: Jayantheesh Srimushnam - TLS, Chennai Subject: XCAP server Hi All, I have downloaded the SER and trying to run XCAP server. I have gone through the README file for the XCAP server and understood that it is server simulation. … [View More]

1 0

[Users] newbie wants to forward REGISTERs through DMZ to internal SIP server
by Simon Barber 08 Nov '06

08 Nov '06

Hi All, I have an internal SIP server that handles registrations, and is used for incoming and outgoing phone calls. It does not do any authentication. I'd like to set up SER in my DMZ, and have it: 1. authenticate all requests. (this should be easy - I think I can handle it, although I would like to get digest -> RADIUS auth against an Active directory going). 2. forward REGISTER requests to the internal SIP server, both for phones on the internal network and for phones on the … [View More]

2 1

[Users] Problems with DNS.
by nick 08 Nov '06

08 Nov '06

I'm trying to get my machine ready to work externally, with public Ip addresses. I believe I've got all the pieces ready for the domain. I have the PTR records published for my server's public IP address. I have the .sip_.udp.mobilia.it. 0 0 5060 sipserver.mobilia.it. record (I tried it with and without the .mobilia.it added to the udp) published, I tell the server to listen on the IP addresses which maps it's name in DNS, I have a line after listen : alias="mobilia.it" (I've tried both … [View More]

5 13

Re: [Users] Errors whie starting openser with radius server integration
by Sanjeev Manoli 08 Nov '06

08 Nov '06

Hi Ravi, Thanks for your help and suggestion. I went through avp module documentation and figured out that the avp_check syntax in config.cfg file was incorrect. I changed it as follows and now it's working, if(!avp_check("$avp(i:2)", "eq/$avp($src_ip)/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; But still I get parse error for following 2 statements in openser.cfg (I commented them to make forward progress).… [View More] Please advice modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("avpops", "avp_aliases", "day=i:101;time=i:102") Another thing is I am currently using the file based radius authentication for the users. (I have created a file called /usr/local/etc/raddb/users with 2 users). Do you know how to enable/use radius with user authentication done from a database (i am using mysql). I couldn't find any documentation for it. Thanks for the help - Sanjeev On 11/6/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > Usrs usrloc is comment, > > > > > > On 11/7/06, Sanjeev Manoli <meghsan(a)gmail.com> wrote: > > Hi, > > I am getting following errors as seen in /var/log/message while > > running openser with radius integration, > > > ***************************************************************************************** > > Nov 6 20:33:45 lx-dev monit[13565]: 'openser' start: /etc/init.d/openser > > Nov 6 20:33:45 lx-dev monit[13565]: 'openser' failed to start > > Nov 6 20:33:45 lx-dev openser: init_tcp: using epoll_lt as the io > > watch method (auto detected) > > Nov 6 20:33:45 lx-dev openser: INFO: statistics manager successfully > > initialized > > Nov 6 20:33:45 lx-dev openser: StateLess module - initializing > > Nov 6 20:33:45 lx-dev openser: TM - initializing... > > Nov 6 20:33:46 lx-dev openser: Maxfwd module- initializing > > Nov 6 20:33:46 lx-dev openser: AVPops - initializing > > Nov 6 20:33:46 lx-dev openser: TextOPS - initializing > > Nov 6 20:33:46 lx-dev openser: ACC - initializing > > Nov 6 20:33:46 lx-dev openser: AUTH module - initializing > > Nov 6 20:33:46 lx-dev openser: xl_parse_item: error - bad parameters > > Nov 6 20:33:46 lx-dev openser: ERROR:avpops:fixup_check_avp: unable > > to get pseudo-variable in P1 > > Nov 6 20:33:46 lx-dev openser: ERROR: fix_actions: fixing failed > > (code=-2) at cfg line 146 > > Nov 6 20:33:46 lx-dev openser: ERROR: fix_expr : fix_actions error > > > ***************************************************************************************** > > I am using openser (Version: openser-1.1.0-tls) and radius server > > (freeradius-1.1.3) along with radiusclient-ng (radiusclient-ng-0.5.2 ). > > I exactly followed the following radius integration documentation > > from openser web site, > > http://openser.org/docs/openser-radius-1.0.x.html > > > > If I remove the radius integration related part from openser.cfg then > > my openser server starts fine, I have tested it with kphone SIP UA and > > it works fine. > > > > One other question is I get parse error for following 2 statements in > > openser.cfg (I commented them to make forward progress). Please advice > > the right syntax to use following modparam statements. > > modparam("acc", "radius_extra", > "Sip-Src-IP=$si;Sip-Src-Port=$sp") > > modparam("avpops", "avp_aliases", "day=i:101;time=i:102") > > > > This is kind of urgent for me and I am clueless at this point so > > really appreciate all your help. > > > > Thanks, > > - San > > P.S. For your reference here is the openser.cfg file that I am using, > > # > > # $Id$ > > # > > # radius config script > > # > > > > # ----------- global configuration parameters ------------------------ > > > > debug=9 # debug level (cmd line: -dddddddddd) > > fork=no > > log_stderror=no # (cmd line: -E) > > > > check_via=no # (cmd. line: -v) > > dns=no # (cmd. line: -r) > > rev_dns=no # (cmd. line: -R) > > port=5060 > > children=4 > > listen=udp: 192.168.0.5 > > alias="192.168.0.5" > > > > #fifo="/tmp/openser_fifo" > > > > # ------------------ module loading > ---------------------------------- > > mpath="/usr/local/lib/openser/modules" > > > > loadmodule "mysql.so" > > loadmodule "sl.so" > > loadmodule "tm.so" > > loadmodule "rr.so" > > loadmodule "maxfwd.so" > > loadmodule "avpops.so" > > loadmodule " usrloc.so" > > loadmodule "registrar.so" > > loadmodule "textops.so" > > loadmodule "xlog.so" > > loadmodule "uri.so" > > loadmodule "acc.so" > > loadmodule "auth.so" > > loadmodule "auth_radius.so" > > loadmodule "group_radius.so" > > loadmodule "avp_radius.so" > > > > # ----------------- setting module-specific parameters --------------- > > > > # -- usrloc params -- > > > #modparam("usrloc","db_url","mysql://openser:openserrw@localhost/openser") > > modparam("usrloc", "db_mode", 2) > > > > # -- acc params -- > > modparam("acc", "radius_flag", 1) > > modparam("acc", "radius_missed_flag", 2) > > modparam("acc", "log_flag", 1) > > modparam("acc", "log_missed_flag", 1) > > modparam("acc", "service_type", 15) > > #modparam("acc", "radius_extra", > "Sip-Src-IP=$si;Sip-Src-Port=$sp") > > modparam("acc|auth_radius|group_radius|avp_radius", > "radius_config", > > "/etc/radiusclient-ng/radiusclient.conf") > > > > # -- group_radius params -- > > modparam("group_radius", "use_domain", 1) > > > > # -- avpops params -- > > #modparam("avpops", "avp_aliases", "day=i:101;time=i:102") > > > > # -- rr params -- > > # add value to ;lr param to make some broken UAs happy > > modparam("rr", "enable_full_lr", 1) > > > > # ------------------------- 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; > > }; > > > > # check if user is suspended > > > if(is_method("REGISTER|INVITE|MESSAGE|OPTIONS|SUBSCRIBE")) > > { > > if (radius_is_user_in("From", "suspended")) { > > sl_send_reply("403", "Forbidden - > suspended"); > > exit; > > }; > > }; > > > > # 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); > > }; > > > > if(is_method("INVITE") && !has_totag()) > > { # set the acc flags > > setflag(1); > > setflag(2); > > }; > > > > 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) { > > # authenticate registers > > if (method=="REGISTER") { > > if (!radius_www_authorize("192.168.0.5")) { > > www_challenge("192.168.0.5", "0"); > > exit; > > }; > > > > # check the src ip address > > > Call the method is Wrong , Check the Module doc od avp_radius and Avp, > > > > > if(!avp_check("i:2", "eq/$src_ip/ig")) > > { > > sl_send_reply("403", "Forbidden IP"); > > exit; > > }; > > > > save("location"); > > exit; > > }; > > > > # calls to pstn > > if(uri=~"sip:00[1-9][0-9]+@") { > > if(is_method("INVITE") && !has_totag()) { > > if (!radius_is_user_in("From", "pstn")) { > > sl_send_reply("403", "Forbidden > PSTN"); > > exit; > > }; > > }; > > # set gateway address > > rewritehostport(" 10.10.10.10:5090"); > > route(1); > > }; > > > > # load callee's avps > > if(avp_load_radius("callee")) > > { > > # check if user has time filter enabled > > if(avp_check("i:3", "eq/i:1")) > > { > > # print time in an avp > > avp_printf("i:100", "$Tf"); > > # extract day > > avp_subst("i:100/i:101", "/(.{3}) > .+/*\1*/"); > > if(!avp_check("i:6", "fm/$day")) { > > sl_send_reply("403", "Forbidden - > day"); > > exit; > > }; > > # extract 'hours:minutes' > > avp_subst("i:100/i:102", "/(.{10}) > (.{5}):.+/\2/"); > > if((is_avp_set("i:4") && avp_check("i:4", > "gt/$time")) > > || (is_avp_set("i:5") && avp_check("i:5", "lt/$time"))) { > > sl_send_reply("403", "Forbidden - > time"); > > exit; > > }; > > }; > > }; > > > > # native SIP destinations are handled using our USRLOC DB > > if (!lookup("location")) { > > # log to acc as missed call > > acc_rad_request("404 Not Found"); > > acc_log_request("404 Not Found"); > > sl_send_reply("404", "Not Found"); > > exit; > > }; > > append_hf("P-hint: usrloc applied\r\n"); > > }; > > > > route(1); > > } > > > > # generic forward > > route[1] { > > # send it out now; use stateful forwarding as it works reliably > > # even for UDP2TCP > > if (!t_relay()) { > > sl_reply_error(); > > }; > > exit; > > } > > # > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com > www.hyperion-tech.com [View Less]

1 0

Re: [Users] TM : retransmission timers
by Rao Ramaratnamma 08 Nov '06

08 Nov '06

the ser ottendorf announcement does mention improved timers. Cannot openser include this feature too and cannot I merge ser with openser for good timers? I am still trying to understand the difference between ser and openser but standart compliance seems to be very important matter! Cannot people provide me with some hints? I am sure that I am not the only who is asking the difference between ser and openser. ser documentation does not appear uptodate, but the software as sannounced appears … [View More]impressive. I have already asked this question but did not receive any answer. thank you in advance! rr ----- Original Message ---- From: Christian Schlatter <cs(a)unc.edu> To: users(a)openser.org Sent: Tuesday, November 7, 2006 10:52:56 PM Subject: Re: [Users] TM : retransmission timers Greg Fausak wrote: > Hello, > > I believe this is a well known bug. > Granularity of timers is 1 second. So, if you sign up for a timer to > be fired in 1 second it will happen anywhere between 0 seconds and 1 > second. > 2 seconds will happen between 1 and 2 seconds. I usually set up my > timers to be 2, 2, 4, 8. There are VOIP providers that are pretty > sticky about > the first 500ms. If you are using one of them you're out of luck. Yes, there is a timer process that wakes up every second to perform retransmissions. I was actually quite surprised that OpenSER, which is known to be very standards compliant, does not follow the RFC 3261 retransmission timeouts. On the other hand, the RFC 3261 timeout values are just suggestions and standards compliant SIP UA must accept shorter timeouts. Still it would be nice if OpenSER would support sub second timers, this would allow for shorter fail-over times. Christian > > I believe SER has made timer changes to support more exact timer > intervals. They are a completely different camp, with a different feature > set (although they share the same roots). > > -g > > > On 11/7/06, Jean-François SMIGIELSKI <jf-smig(a)ibelgique.com> wrote: >> Hello, >> >> I made strange observations about the intervals between >> retransmissions with the TM module. >> In my experiments, I used the default parameters for the TM module >> timers, and I sent an INVITE that cannot receive answers (it has a >> well known R-URI pattern that is forwarded to a place and port that >> nobody listen). >> >> When reading RFC3261, I expected to see intervals between >> retransmissions of |500ms|1s|2s|4s|8s|16s|. 7 transmissions, during 32s. >> >> But with OpenSER, (I have tested with the debian package 1.1.0-5 on a >> debian etch, and the cvs sources for 1.1.0 or 1.0.1compiled by >> myself), I can see intervals like <500ms, 2s, 4s, 4s,4s, ... until 26s >> are spent (9 sendings). The first interval is sometomes very short >> (40ms). >> >> Altough I like the sequence of 4s separated transmissions, I do not >> know why the first interval is so short, and why there is no sending >> after 1s. >> >> Did anybody observed such behaviours? Are they normal? >> >> Thanks in advance! >> >> JF Smigielski. >> >> >> ________________________________________________________________________ >> iBELGIQUE, exprimez-vous ! >> http://web.ibelgique.com/ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users [View Less]

2 1

[Users] one way audio
by rilawich ango 08 Nov '06

08 Nov '06

Hi all, Anyone experiences an one way audio problem when a call is connected to the called party? I am using openser with mediaproxy to provide services to our customer. However, some of our customer complain about the one way audio problem (calling party can hear called party but not vice versa). The system works to most of the customer without that problem. I wonder why there are some customers experience that problem. I assume there is the port problem during RTP packet transmission. … [View More]

1 0

[Users] Forwarding requests
by Harry Harcrow 07 Nov '06

07 Nov '06

_____ From: Harry Harcrow [mailto:harry.harcrow@verizonbusiness.com] Sent: Tuesday, November 07, 2006 3:36 PM To: 'users(a)openser.org' Subject: Forwarding requests I have my SER server running on a Linux machine and I am attempting to forward some requests to a Cisco SIP server. I have been running X-lite clients on windows and the same Linux box that the SER server is running on. I have no problem completing calls between the X-lite clients OR with either X-lite client calling the … [View More]

1 0

[Serusers] ser accounting Issue
by ram 07 Nov '06

07 Nov '06

Hi i have SER running with mysql i have a question related to accounting when the calls are live, when i reboot the server i wont see the Bye messages, this kind of situation, how can i terminate the call i mean send bye message to all pending calls and account them when i server come back any help on this issue iam not sure if this issue already addressed Ram

1 0

[Users] photos from OpenSER BoF
by Daniel-Constantin Mierla 07 Nov '06

07 Nov '06

Hello, just to get a feeling about how the event is going, we uploaded few pictures, since the wireless is very crowded, bandwidth is quite limited -- more will come in the next days, along with the slides. We are very please to have more that 75% attendees from the first days, the room for BoF session proved very small, hopefully tomorrow the places in summit room will be enough -- we got a lot of new people interested in the summit from VoN event. http://www.openser.org/pictures/openser-bof/ Cheers, Daniel

1 0

[Serusers] error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
by vijay tiwari 07 Nov '06

07 Nov '06

hi all we i call to one UA to another UA i found this error. error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy thanks vijay --------------------------------- Sponsored Link Free Uniden 5.8GHz Phone System with Packet8 Internet Phone Service

1 0

[Users] OpenSER Summit @ VoN Berlin - ready to go
by Daniel-Constantin Mierla 07 Nov '06

07 Nov '06

Hello everybody, OpenSER summit is ready to start in one day. Everything is in place, although some fine tunings will be there all the time. Next days the activity on mailing lists may decrease a bit, as many of OpenSER users are joining the event. To check the event's schedule and details: http://www.voneurope.com/expo.html http://www.openser.org The event will not be broadcasted on the web, the presentations will be collected and published after the event. Photos and briefing will be … [View More]

3 2

Re: [Serusers] new major SER pre-release, Ottendorf, is out for testing
by sip 07 Nov '06

07 Nov '06

Well it's about time!!! ;) Seriously though, thanks for all the hard work you've all put in on this release. I'm looking forward to getting the chance to install and play with it, and (most importantly for me) documenting as much as I can about how to migrate from a production 0.9.X solution to the new code -- something I imagine is going to be tricky as much of our actual web/management code will need to be rewritten. I can't express really how much SER and the Iptel team have changed my … [View More]life for the better. Thanks again for everything. N. On Mon, 06 Nov 2006 12:02:24 +0100, Jiri Kuthan wrote > Ladies and gentlemen, girls and boys, > > the new testing pre-release of the best SIP proxy server ever, SER, > is out! > > This major pre-release, codenamed Ottendorf, includes fundamental > improvements to the software. The improvements allow SER users to > achieve significantly higher managability and reliability by better > data mode, interop features, security, etc. They recollect > voluminous feedback we have received from iptel customers and open > source users. In particular, the new features include but are not > limited to: - improved data model which eliminates any possible ambiguities. > - key under-the-hood changes with impact on performance: timer subsystem > overhaul (now fast, with fine resolution, high accuracy and > variability), TCP processing overhaul (higher scalability) - key > reliability changes: DNS cache and IP blacklisting - key interop > changes: support for STUN-bis, presence server aligned to latest > IETF development, sanity module for early detection of invalid messages > - more management: management features are now available via XML-RPC > remotely, as well as via an alternative binary protocol - more > security: ratelimit module, more check types in permissions module - > scripting features: select operator in combination with revised AVPs, > named flags and routes, more types of routes (onsend, timer, > branch) - applications: presence has been greatly updated to keep > pace with development in the IETF/SIMPLE (includes asynchronous subscription > authorization, extended PIDF); web interface includes many new managibility > features (multidomain, templates and multilanguage to name some). New > scripting flexibility allows safely for features like individual ringing > timers and direct access to SIP message elements. > - new tools: Nils's script execution tree tool > http://lists.iptel.org/pipermail/serdev/2006-October/008124.html > and refurbished ser_ctl > see the following links for some additional details: > - http://www.iptel.org/ser/doc/010whatsnew > - http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/NEWS?rev=HEAD > > Accompanying software, SEMS and SERWEB have been greatly overhauled > as well. SEMS now features improved performance (single-threaded > model now), service composition via python-IVR-scripting and B2BUA, > adaptive playout buffer for better voice quality, and more. SERWEB > features new multilevel administrative access system, multidomain > support, per-domain customization, remote xml-rpc access, > multilanguage support, and more. The new command-line utility, > SER_CTL, provides also many great improvements over the previous tool. > > Interop > ------- > Shortly, we have been keeping the bar high through our participation > in the IETF and SIPITs. SER implements latest interop changes from > the IETF (STUN-bis, presence + XCAP, to name the most important ones) > and both new features and legacy features were extensively and > successfully tested middle of October in SIPit. > > Next Steps > ----------- > Keep in mind, this is a pre-release for testing purposes. It is now > call on users to test the software hard and provide feedback. In the > meantime, the codebase remains frozen and we only commit patches > fixing various bugs. Once we get good confidence that Ottendorf has > stabilized and is being adequately documented, we will declare the > final release. No date has been determined yet. Also, we consider > changing CVS to SVN. > > Again, and this is very important: the final release will be as good > as the feedback we obtain for this pre-release. Please do spend your > time sending solid feedback back to [serusers]. > > Roadmap > ------- > We keep rolling. We are still debating with our generous > contributors what is the next possible good things to work on, the > topics include but are not limited to as of now: - documentation. - > databases are important. LDAP and BDB, DB-API revision. - interop: > identity, TLS revision - advanced scripting (scripting engine, vars > returns, AVPs) - uncountable improvements (futex, dev/rtc, > tm/refcnts, tcp fd cache, .....) > > Trivia > -------- > The release name Ottendorf is coming from a place in Saxonian > Switzerland, where many iptel fellows and their friends spent a > weekend during this Indian summer. http://www.iptel.org/jan/gallery/saxonian_switzerland_2006/ > http://www.iptel.org/~andi/2006-09-22-Ottendorf/ > http://www.iptel.org/~vlada/saxon_switzerland/index.html > http://www.ohlmeier.de/albums/ottendorf_2006/index.html > > Acknowledgments > --------------- > This grand achievement is result of extremely many folks who did lot > of hard work. Out of folks affiliated with iptel, Andrei and Jan > spent immense effort on moving SER forward, with great support from Hallik > (command-line tool), Michal, Miklos, Nils, Tomas, Vaclav (presence), > Vlada (new STUN). In this context, I would like to thank to our > company for the great support. Recently, Bogdan Pintea greatly > helped to make the upcoming release better and more stable under > sponsorship of his company, iptego. transnexus provided osp support. > > SEMS has been greatly improved thanks to Stefan and Raphael. > Credits for serweb go to Karel. > > Same importantly, SER has been immensely benefitting of contributors, > to name in alphabetic order at least the 'top-of-the-iceberg' > fellows: Bogdan Pintea, Cesc Santa, Greg Fausak, Greger Teigre, > Hendrik Scholz, <http://tutpro.com/>Juha Heinänen, Klaus Darilion, > Maxim Sobolev, Peter Nixon, Uli Abend. > > Great support on the mailing list has been provided by Greger V. > Teigre, Java Rockx, Klaus Darilion, Zeus Ng, and more. > > Simply, this list is too short and incomplete to capture all the > generous contributions that came in -- my appologies for everyone > else I have missed. > > Warnings and Disclaimers > ------------------------ > This is a TESTING version. We are encouraging people to give it intense > testing and provide us with feedback, so that we can properly assess > what it takes to declare it final. The volume and number of changes > is big, caution is advised. > > All GPL warnings and disclaimers apply. > > This is SER, which is not the derivate software known as openser. We > have no ideas of in what extent and timeline derivate projects follow > (or not) SER. > > ------------------------------------------------------------------------------- > > To Give it a Try > ----------------- > warning: this is just a 'jump-start' for black-belt users. It is not > a replacement for documentation or a detailed step-by-step installation > guide. It assumes knowledge of operating system, SIP, and SER. > > PREPARATION > - make sure mysql is running. In the default configuration, SER > assumes it to be running on localhost, if this does not hold, you > need to change ser.cfg. - make sure python is installed (needed for > new command-line utilities) SER INSTALLATION FROM CVS - Download > from CVS/HEAD using instructions at http://www.iptel.org/downloads - > Compile it: make include_modules="mysql xmlrpc" all - Install it > the default way of your administrative privileges allow to: make > include_modules="mysql xmlrpc" install - Download command-line tool > (see more documentation under http://cvs.berlios.de/cgi- > bin/viewcvs.cgi/*checkout*/ser/serctl/README?rev=HEAD) - cvs co serctl > - Initiate database: > ./scripts/mysql/ser_mysql.sh create > START > - Start it. Call: ser > - Begin provisioning users with the new serctl (the capital word are > just place holders) # ser_ctl domain add DOMAINNAME # ser_ctl > user add USERNAME@DOMAINNAME -p PASSWORD # If you want to have > Asserted Identity header for your user # - ser_attr add uid=UID asserted_id="PID" > # If you want to have gateway support > # - ser_db add attr_types name=gw_ip rich_type=string raw_type=2 \ > description="The gateway IP for the default ser.cfg" default_flags=33 > # - ser_attr add global gw_ip=GATEWAY-IP > - more things you may wish to do: > - familiarize yourself with current ser.cfg > - send us feedback. > > -- > Jiri Kuthan http://iptel.org/~jiri/ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers [View Less]

2 2

[Users] Re: [Semsdev] SEMS users meeting at VON Europe 2006 (Berlin)
by Stefan Sayer 07 Nov '06

07 Nov '06

Hello, we would like to meet tomorrow, Wednesday, at 13:00 in the exhibition area. I hope to see those of you who are at VON Europe and interested in SEMS to discuss its development. Regards Stefan Raphael Coeffic wrote: > > Hello everybody, > > Stefan Sayer and I are proud to invite you to a SEMS users meeting at the > VON > Europe 2006 in Berlin on wednesday 8th november. It could a great chance > to > discuss all your favorite features and gives us the possibility … [View More]

1 0

[Serusers] Using different domains
by Nick Hoffman 07 Nov '06

07 Nov '06

Hi guys. I just noticed that when user agents register, the domain in their SIP URI does not have to equal their domain in the subscribers database table. For instance, I have 2 domains in the domain table: sip.altcall.com sip.voxpak.com The username "foobar" in my subscribers table has the domain "sip.altcall.com", but can register and make calls as foobar(a)sip.voxpak.com . Why is this? Cheers, -- Nick E: nick.hoffman(a)voxpak.com P: +61 7 5591 3588 F: +61 7 5591 6588 If you … [View More]

3 2

[Users] Connect two OpenSER...
by Steffen Witt 07 Nov '06

07 Nov '06

Hello, I have two OpenSER servers that I want to "connect" with each other: Phone "a" <---> Server "A" <------------> Server "B" <---> Phone "b" Phone "a" should call phone "b" without adding the domain during call initiation and vice versa. Phone numbers from "A" starts with "5xxxx" and from "B" with "6xxxx". I have tried to edit openser.cfg e.g. at server "A": route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!… [View More]

2 1

[Users] Messege is too big Error Code 513
by ShivaSankar Padhy 07 Nov '06

07 Nov '06

Dear all! I am still new to the openser. Let me ask another silly question:) I am getting error messege as 513, messege is too big, May I know in what condition is is comming??What is the rule to avoid this. Expectig a sooner response as always, Thanks in Advance With cheers, ShivaSankar

3 2

[Users] FW: XMPP module
by Herchi Silviu 07 Nov '06

07 Nov '06

> Hi list, > > I'm new to OpenSER, and I'm trying to compile the XMPP module. > > So far, I've checkd out the CVS version and compiled it successfully. > However, the XMPP module wouldn't compile, even though I have taken it > out of the exclude list in the Makefile. The only way to compile it > was to run "make" in the modules/xmpp directory (I know, I'm not > supposed to, but it worked...). > > I now have the following in my configuration file: > > [.… [View More]

1 0

[Serusers] Variables in Ser.cfg
by Benito Caracuel 07 Nov '06

07 Nov '06

Hello all, I have a file txt and a variable. I need to do this in ser.cfg: $variable = cat /tmp/file.txt Replace_all("bobobo", "%$variable"); But It`s not run. How can I do this? Thank you very much. _________________________________________________________________ Llama a tus amigos de PC a PC: ¡Es GRATIS! http://get.live.com/messenger/overview

1 0

[Serusers] Re: some simple newbie help with dialplan needed...
by Evert 07 Nov '06

07 Nov '06

Hi! I have SER running on 5060, and Asterisk on 5061, so if Asterisk passes it to @localhost there should be no problem there. The remaining problem seems to be the passing on of the call. The mentioned dialplan does send the entered extension to SER, and the phone on that SER-account rings. However, when the called phone anwers the call, the calling phone does not notice this, and keeps on ringing... No connection is established. I am not sure whether this is caused by Asterisk or SER, so I … [View More]

1 0

RE: [serusers]: trusted table(permissions module)
by Kamal.Mann＠t-systems.com 07 Nov '06

07 Nov '06

Hi All In current scenario when a sip invite message comes to SER from my SIP_AS; SER take this massage trusted (as I defined in trusted table) but after this in my INVITE route(3) it sends it to route(1) default message handler. I am enclosing my ser.cfg; please take a glance on it. Thanks in anticipation Kamal Mann -----Original Message----- From: John Shellhouse [mailto:jwshel@hotmail.com] Sent: Friday, November 03, 2006 9:17 PM To: Mann, Kamal Subject: RE: [serusers]: trusted table(… [View More]permissions module) Hi, you've already tried something more filled in like this... if (!proxy_authorize("somedomain.com","subscriber")) { proxy_challenge("somedomain.com","0"); break; } else if (method == "INVITE" & !check_from()) { log("from!=digest\n"); sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); T >From: <Kamal.Mann(a)t-systems.com> >To: <andrey.kouprianov(a)gmail.com>, <serusers(a)iptel.org> >Subject: RE: [serusers]: trusted table(permissions module) >Date: Fri, 3 Nov 2006 15:40:26 +0100 > >Hi All >Now SER is trusting packates from trusted table IP and in the following >part of route(3) of Invite message handler allow_trusted results 'true' and >it directly bypasses this code. >if (!allow_trusted()) >{ > if (!proxy_authorize("","subscriber")) > { > proxy_challenge("","0"); > break; > } > else if (!check_from()) > { > sl_send_reply("403", "Use From=ID"); > break; > }; > consume_credentials(); >}; >But after in route(1) its sending 407-proxy auth required to trusted ip of >sip_AS. Please help me out of this dilemma. > >Thanks in anticipation >Kamal Mann > >-----Original Message----- >From: serusers-bounces(a)lists.iptel.org >[mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kuprianov >Sent: Thursday, November 02, 2006 1:13 PM >To: serusers(a)iptel.org >Subject: Re: [serusers]: trusted table(permissions module) > >The one you attached, didnt have a semicolon. Anyway, try using a >fresh copy of a ready made ser.cfg. > > Bests, > > Andrey. > >On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > Yup I always do the same, Might be copy paste error. Its having >semicolon & working fine here. > > > > Regards > > Kamal Mann > > > > -----Original Message----- > > From: Andrey Kuprianov [mailto:andrey.kouprianov@gmail.com] > > Sent: Thursday, November 02, 2006 10:34 AM > > To: serusers(a)iptel.org > > Subject: Re: [serusers]: trusted table(permissions module) > > > > Hi Kamal, > > > > Whenever you modified your ser.cfg did you restart SER after? Did you > > check your ser.cfg for errors using "ser -c" command? I found some > > errors in your ser.cfg code. For instance, inside the main route > > block, your "if", which processes INVITEs, is missing a semicolon at > > the end of the block. > > > > if (method=="INVITE") > > { > > # sl_send_reply("404", "INVITE "); > > route(3); > > break; > > } <----- missing semicolon here > > > > > > If you are comfortable with your current ser.cfg, that's ok, but I'd > > suggest you start with a fresh one, make your trusted table work, and > > then add a "REFER" processing, like in your currect ser.cfg. Just > > download one fresh ser.cfg from iptel.org site and add your > > allow_trusted() "if" clause there. > > > > Regards, > > > > Andrey. > > > > > > On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > > Hi > > > Please find ser.cfg enclosed. > > > > > > Trusted table entry: > > > +---------------+-------+--------------+ > > > | src_ip | proto | from_pattern | > > > +---------------+-------+--------------+ > > > | 10.25.119.156 | any | ^sip:.*$ | > > > +---------------+-------+--------------+ > > > 1 row in set (0.00 sec) > > > > > > Thanks in anticipation > > > Kamal Mann > > > -----Original Message----- > > > From: serusers-bounces(a)lists.iptel.org >[mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kuprianov > > > Sent: Wednesday, November 01, 2006 7:03 PM > > > To: serusers(a)iptel.org > > > Subject: Re: [serusers]: trusted table(permissions module) > > > > > > Hi Kamal, > > > > > > I understand your setup. Nevertheless, it will be difficult to figure > > > out your problem, w/o your ser.cfg file at hand. Also, send a trusted > > > table query result (i.e. that row which contains entry for your > > > SIP-AS). > > > > > > Regards, > > > > > > Andrey. > > > > > > On 11/1/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > > > Hi All > > > > In my scenario I need ser to communicate with my SIP-AS. This SIP-AS >is having an application xyz running on it. This XYZ sends an invite to URI >"A(a)xcv.de" which is a subscriber in SER and this is in TO header of Invite >msg & FROM header contains another URI test(a)xcv.de but this URI >(test(a)xcv.de) isn't registered / subscribed to SER. I need SER trust all >packets from SIP-AS ip. BUT SER is sending 407 reply of INVITE to XYZ >application! IP in trusted table is of SIP-AS on top of which XYZ >application is running. > > > > > > > > Thanks in anticipation > > > > Kamal Mann > > > > > > > > -----Original Message----- > > > > From: Maciej ¯wirski [mailto:mzwirek@poczta.fm] > > > > Sent: Wednesday, November 01, 2006 3:20 PM > > > > To: serusers(a)lists.iptel.org > > > > Subject: Re: [serusers]: trusted table(permissions module) > > > > > > > > Kamal.Mann(a)t-systems.com wrote: > > > > > > > > > Hi All > > > > > > > > > > I made an entry /"10.25.119.156, any, ^sip:.*$"/ into trusted >table so > > > > > that any packet sent from 10.25.119.156 would be trusted by SER >and > > > > > don't ask for its credentials. But SER is replying *407- proxy* >auth > > > > > needed!! After this I tried with permissions.allow file: > > > > > > > > > Hi, > > > > I had the same issue while interconnecting SER and Asterisk. It >turned > > > > out to be an issue on Asterisk side (I had the same user added for >SER > > > > and Asterisk, so the poor thing couldn't authorize either :)) So you > > > > could check if the 407 you're getting is from SER or from the remote >host. > > > > > > > > Regards, > > > > Maciej Zwirski > > > > > > > > >---------------------------------------------------------------------- > > > > Jestes kierowca? To poczytaj! >>> http://link.interia.pl/f199e > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > Serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > _______________________________________________ > > > > Serusers mailing list > > > > Serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Get FREE company branded e-mail accounts and business Web site from Microsoft Office Live http://clk.atdmt.com/MRT/go/mcrssaub0050001411mrt/direct/01/ [View Less]

4 17

[Users] Errors whie starting openser with radius server integration
by Sanjeev Manoli 07 Nov '06

07 Nov '06

Hi, I am getting following errors as seen in /var/log/message while running openser with radius integration, ***************************************************************************************** Nov 6 20:33:45 lx-dev monit[13565]: 'openser' start: /etc/init.d/openser Nov 6 20:33:45 lx-dev monit[13565]: 'openser' failed to start Nov 6 20:33:45 lx-dev openser: init_tcp: using epoll_lt as the io watch method (auto detected) Nov 6 20:33:45 lx-dev openser: INFO: statistics manager … [View More]successfully initialized Nov 6 20:33:45 lx-dev openser: StateLess module - initializing Nov 6 20:33:45 lx-dev openser: TM - initializing... Nov 6 20:33:46 lx-dev openser: Maxfwd module- initializing Nov 6 20:33:46 lx-dev openser: AVPops - initializing Nov 6 20:33:46 lx-dev openser: TextOPS - initializing Nov 6 20:33:46 lx-dev openser: ACC - initializing Nov 6 20:33:46 lx-dev openser: AUTH module - initializing Nov 6 20:33:46 lx-dev openser: xl_parse_item: error - bad parameters Nov 6 20:33:46 lx-dev openser: ERROR:avpops:fixup_check_avp: unable to get pseudo-variable in P1 Nov 6 20:33:46 lx-dev openser: ERROR: fix_actions: fixing failed (code=-2) at cfg line 146 Nov 6 20:33:46 lx-dev openser: ERROR: fix_expr : fix_actions error ***************************************************************************************** I am using openser (Version: openser-1.1.0-tls) and radius server (freeradius-1.1.3) along with radiusclient-ng (radiusclient-ng-0.5.2). I exactly followed the following radius integration documentation from openser web site, http://openser.org/docs/openser-radius-1.0.x.html If I remove the radius integration related part from openser.cfg then my openser server starts fine, I have tested it with kphone SIP UA and it works fine. One other question is I get parse error for following 2 statements in openser.cfg (I commented them to make forward progress). Please advice the right syntax to use following modparam statements. modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("avpops", "avp_aliases", "day=i:101;time=i:102") This is kind of urgent for me and I am clueless at this point so really appreciate all your help. Thanks, - San P.S. For your reference here is the openser.cfg file that I am using, # # $Id$ # # radius config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=no log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=udp:192.168.0.5 alias="192.168.0.5" #fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "avpops.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "uri.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc","db_url","mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- #modparam("avpops", "avp_aliases", "day=i:101;time=i:102") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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; }; # check if user is suspended if(is_method("REGISTER|INVITE|MESSAGE|OPTIONS|SUBSCRIBE")) { if (radius_is_user_in("From", "suspended")) { sl_send_reply("403", "Forbidden - suspended"); exit; }; }; # 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); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; 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) { # authenticate registers if (method=="REGISTER") { if (!radius_www_authorize("192.168.0.5")) { www_challenge("192.168.0.5", "0"); exit; }; # check the src ip address if(!avp_check("i:2", "eq/$src_ip/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; # calls to pstn if(uri=~"sip:00[1-9][0-9]+@") { if(is_method("INVITE") && !has_totag()) { if (!radius_is_user_in("From", "pstn")) { sl_send_reply("403", "Forbidden PSTN"); exit; }; }; # set gateway address rewritehostport("10.10.10.10:5090"); route(1); }; # load callee's avps if(avp_load_radius("callee")) { # check if user has time filter enabled if(avp_check("i:3", "eq/i:1")) { # print time in an avp avp_printf("i:100", "$Tf"); # extract day avp_subst("i:100/i:101", "/(.{3}) .+/*\1*/"); if(!avp_check("i:6", "fm/$day")) { sl_send_reply("403", "Forbidden - day"); exit; }; # extract 'hours:minutes' avp_subst("i:100/i:102", "/(.{10}) (.{5}):.+/\2/"); if((is_avp_set("i:4") && avp_check("i:4", "gt/$time")) || (is_avp_set("i:5") && avp_check("i:5", "lt/$time"))) { sl_send_reply("403", "Forbidden - time"); exit; }; }; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } # generic forward route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } # [View Less]

1 0

[Serusers] flatstore data fields
by Matthew McGuire 06 Nov '06

06 Nov '06

I am using flatstore module for capturing stateful call records. This gives much faster performance than mysql inserts into acc. However, I'd like to do batch inserts so that I can still use the mysql acc table structure for accounting and billing. Unfortunately the data fields do not match up exactly. Is there a document that explains what data fields are being generated by flatstore and how to correlate them to acc? thanks, matt

1 0

[Users] OpenSER + Sems?
by Brandon＠voipo.com 06 Nov '06

06 Nov '06

Hello, Can anyone provide me with a basic implementation of confrencing and voicemail with sems? For the life of me I can't seem to figure out a few loose ends and get it working correctly, well thankyou for any advice or help you may have. Regards, Brandon Armstead Sent via BlackBerry from T-Mobile

1 0

[Serusers] 2 clients behind the same NAT
by Ira Kadin 06 Nov '06

06 Nov '06

Dear all, we are using the NAT - mediaproxy solution. Is there a way to avoid using mediaproxy in case both sides (client and server) are located behind the same NAT (something like comparing private and receiving from (NAT) IP's - and in case they are equal - don't call use_mediaproxy. Thanks Irena

2 1

Re: [Serusers] Problem with mysql module [mysql.so] file
by samuel 06 Nov '06

06 Nov '06

You can just install the dev packages and afterwards compile SER again. Samuel 2006/11/6, tezet(a)touk.pl <tezet(a)touk.pl>: > Thank u!! > > So i have to install mysql from the beginning or it is possible just to > install this missing packages? I mean would it be enough to download the > source instead of binary distribution or this development version is > separate distribution or extra package?? > > Tomasz. > > When this kind of error appear in the … [View More]compilation (missing .h file) it > > usually means you have to install the developement version of the > > packet. In your case it means you have to install mysql dev pacakges. > > > > Samuel. > > 2006/11/6, tzieleniewski <tzieleniewski(a)o2.pl>: > >> Hi all > >> > >> I've installed ser as it is written in the Getting started manuall with > >> ser source 0.9.7-pre1. I've also installed the mysql5.0 binary > >> distribution. > >> unfortunatelly i've encountered a strange problem. At think that this > >> may be caused by the fact that i used o binary distribution od mysql > >> instead of source. > >> Any way i'm just starting my adventure with ser:) so i ask u ser guru's > >> to help me;) > >> > >> When i tried to compile the source again including mysql package the > >> mysql.co file did not appeared? > >> and the log contained only such lines regarding mysql: > >> make[1]: Wejďż˝ďż˝cie do katalogu > >> `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' > >> make[1]: Opuszczenie katalogu > >> `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' > >> make[1]: Wejďż˝ďż˝cie do katalogu > >> `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' > >> gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall > >> -minline-all-stringops -malign-double -falign-loops -ftree-vectorize > >> -mtune=athlon64 -DNAME='"ser"' -DVERSION='"0.9.7-pre1"' > >> -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 > >> -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM > >> -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP > >> -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT > >> -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN > >> -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL > >> -DHAVE_ALLOCA_H -I/usr/local/include -I/usr/local/include/mysql > >> -I/usr/local/mysql/include -I/usr/include/mysql -c dbase.c -o dbase.o > >> make[1]: Opuszczenie katalogu > >> `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' > >> > >> When i tried to compile the package manually using Makefile in the > >> "source"/modules/myqsl directory i got plenty of errors:): > >> dbase.c:34:25: error: mysql/mysql.h: Nie ma takiego pliku ani katalogu > >> dbase.c:35:26: error: mysql/errmsg.h: Nie ma takiego pliku ani katalogu > >> In file included from dbase.c:39: > >> val.h:44: error: expected ďż˝ďż˝ďż˝)ďż˝ďż˝ďż˝ before ďż˝ďż˝ďż˝*ďż˝ďż˝ďż˝ > >> token > >> In file included from dbase.c:40: > >> my_con.h:39: error: expected specifier-qualifier-list before > >> ďż˝ďż˝ďż˝MYSQL_RESďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝submit_queryďż˝ďż˝ďż˝: > >> dbase.c:67: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝timestampďż˝ďż˝ďż˝ > >> dbase.c:68: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_pingďż˝ďż˝ďż˝ > >> dbase.c:68: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:72: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝timestampďż˝ďż˝ďż˝ > >> dbase.c:90: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_queryďż˝ďż˝ďż˝ > >> dbase.c:90: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:93: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_errnoďż˝ďż˝ďż˝ > >> dbase.c:93: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:94: error: ďż˝ďż˝ďż˝CR_SERVER_GONE_ERRORďż˝ďż˝ďż˝ undeclared > >> (first use in this function) > >> dbase.c:94: error: (Each undeclared identifier is reported only once > >> dbase.c:94: error: for each function it appears in.) > >> dbase.c:94: error: ďż˝ďż˝ďż˝CR_SERVER_LOSTďż˝ďż˝ďż˝ undeclared (first > >> use in this function) > >> dbase.c:98: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_errorďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:98: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:98: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c: At top level: > >> dbase.c:138: error: expected ďż˝ďż˝ďż˝)ďż˝ďż˝ďż˝ before > >> ďż˝ďż˝ďż˝*ďż˝ďż˝ďż˝ token > >> dbase.c:166: error: expected ďż˝ďż˝ďż˝)ďż˝ďż˝ďż˝ before > >> ďż˝ďż˝ďż˝*ďż˝ďż˝ďż˝ token > >> dbase.c:207: error: expected ďż˝ďż˝ďż˝)ďż˝ďż˝ďż˝ before > >> ďż˝ďż˝ďż˝*ďż˝ďż˝ďż˝ token > >> dbase.c: In function ďż˝ďż˝ďż˝store_resultďż˝ďż˝ďż˝: > >> dbase.c:304: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝resďż˝ďż˝ďż˝ > >> dbase.c:304: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_store_resultďż˝ďż˝ďż˝ > >> dbase.c:304: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:305: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝resďż˝ďż˝ďż˝ > >> dbase.c:306: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_field_countďż˝ďż˝ďż˝ > >> dbase.c:306: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c:311: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:311: warning: format ďż˝ďż˝ďż˝%sďż˝ďż˝ďż˝ expects type > >> ďż˝ďż˝ďż˝char *ďż˝ďż˝ďż˝, but argument 3 has type ďż˝ďż˝ďż˝intďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝db_free_resultďż˝ďż˝ďż˝: > >> dbase.c:348: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝mysql_free_resultďż˝ďż˝ďż˝ > >> dbase.c:348: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝resďż˝ďż˝ďż˝ > >> dbase.c:349: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝resďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝db_queryďż˝ďż˝ďż˝: > >> dbase.c:398: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝print_whereďż˝ďż˝ďż˝ > >> dbase.c:398: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝db_insertďż˝ďż˝ďż˝: > >> dbase.c:471: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝print_valuesďż˝ďż˝ďż˝ > >> dbase.c:471: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝db_deleteďż˝ďż˝ďż˝: > >> dbase.c:516: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c: In function ďż˝ďż˝ďż˝db_updateďż˝ďż˝ďż˝: > >> dbase.c:559: warning: implicit declaration of function > >> ďż˝ďż˝ďż˝print_setďż˝ďż˝ďż˝ > >> dbase.c:559: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> dbase.c:568: error: ďż˝ďż˝ďż˝struct my_conďż˝ďż˝ďż˝ has no member named > >> ďż˝ďż˝ďż˝conďż˝ďż˝ďż˝ > >> make: *** [dbase.o] Bďż˝ďż˝ďż˝ďż˝d 1 > >> > >> > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >> > > > > > [View Less]

1 0

[Users] Re: [Serusers] new major SER pre-release, Ottendorf, is out for testing
by Rao Ramaratnamma 06 Nov '06

06 Nov '06

Thank you very much. I am now in learning phase, can someone possibly bring me to the speed and explain how SER compares to OPENSER. I only found this: http://openser.org/index.php?option=com_content&task=view&id=44 and it seems to be outdated now. rr P.S.: excuse my crossposting but I think this is relevant with this question. ----- Original Message ---- From: Jiri Kuthan <jiri(a)iptel.org> To: serusers(a)iptel.org Sent: Monday, November 6, 2006 12:02:24 PM Subject: [… [View More]Serusers] new major SER pre-release, Ottendorf, is out for testing Ladies and gentlemen, girls and boys, the new testing pre-release of the best SIP proxy server ever, SER, is out! This major pre-release, codenamed Ottendorf, includes fundamental improvements to the software. The improvements allow SER users to achieve significantly higher managability and reliability by better data mode, interop features, security, etc. They recollect voluminous feedback we have received from iptel customers and open source users. In particular, the new features include but are not limited to: - improved data model which eliminates any possible ambiguities. - key under-the-hood changes with impact on performance: timer subsystem overhaul (now fast, with fine resolution, high accuracy and variability), TCP processing overhaul (higher scalability) - key reliability changes: DNS cache and IP blacklisting - key interop changes: support for STUN-bis, presence server aligned to latest IETF development, sanity module for early detection of invalid messages - more management: management features are now available via XML-RPC remotely, as well as via an alternative binary protocol - more security: ratelimit module, more check types in permissions module - scripting features: select operator in combination with revised AVPs, named flags and routes, more types of routes (onsend, timer, branch) - applications: presence has been greatly updated to keep pace with development in the IETF/SIMPLE (includes asynchronous subscription authorization, extended PIDF); web interface includes many new managibility features (multidomain, templates and multilanguage to name some). New scripting flexibility allows safely for features like individual ringing timers and direct access to SIP message elements. - new tools: Nils's script execution tree tool http://lists.iptel.org/pipermail/serdev/2006-October/008124.html and refurbished ser_ctl see the following links for some additional details: - http://www.iptel.org/ser/doc/010whatsnew - http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/NEWS?rev=HEAD Accompanying software, SEMS and SERWEB have been greatly overhauled as well. SEMS now features improved performance (single-threaded model now), service composition via python-IVR-scripting and B2BUA, adaptive playout buffer for better voice quality, and more. SERWEB features new multilevel administrative access system, multidomain support, per-domain customization, remote xml-rpc access, multilanguage support, and more. The new command-line utility, SER_CTL, provides also many great improvements over the previous tool. Interop ------- Shortly, we have been keeping the bar high through our participation in the IETF and SIPITs. SER implements latest interop changes from the IETF (STUN-bis, presence + XCAP, to name the most important ones) and both new features and legacy features were extensively and successfully tested middle of October in SIPit. Next Steps ----------- Keep in mind, this is a pre-release for testing purposes. It is now call on users to test the software hard and provide feedback. In the meantime, the codebase remains frozen and we only commit patches fixing various bugs. Once we get good confidence that Ottendorf has stabilized and is being adequately documented, we will declare the final release. No date has been determined yet. Also, we consider changing CVS to SVN. Again, and this is very important: the final release will be as good as the feedback we obtain for this pre-release. Please do spend your time sending solid feedback back to [serusers]. Roadmap ------- We keep rolling. We are still debating with our generous contributors what is the next possible good things to work on, the topics include but are not limited to as of now: - documentation. - databases are important. LDAP and BDB, DB-API revision. - interop: identity, TLS revision - advanced scripting (scripting engine, vars returns, AVPs) - uncountable improvements (futex, dev/rtc, tm/refcnts, tcp fd cache, .....) Trivia -------- The release name Ottendorf is coming from a place in Saxonian Switzerland, where many iptel fellows and their friends spent a weekend during this Indian summer. http://www.iptel.org/jan/gallery/saxonian_switzerland_2006/ http://www.iptel.org/~andi/2006-09-22-Ottendorf/ http://www.iptel.org/~vlada/saxon_switzerland/index.html http://www.ohlmeier.de/albums/ottendorf_2006/index.html Acknowledgments --------------- This grand achievement is result of extremely many folks who did lot of hard work. Out of folks affiliated with iptel, Andrei and Jan spent immense effort on moving SER forward, with great support from Hallik (command-line tool), Michal, Miklos, Nils, Tomas, Vaclav (presence), Vlada (new STUN). In this context, I would like to thank to our company for the great support. Recently, Bogdan Pintea greatly helped to make the upcoming release better and more stable under sponsorship of his company, iptego. transnexus provided osp support. SEMS has been greatly improved thanks to Stefan and Raphael. Credits for serweb go to Karel. Same importantly, SER has been immensely benefitting of contributors, to name in alphabetic order at least the 'top-of-the-iceberg' fellows: Bogdan Pintea, Cesc Santa, Greg Fausak, Greger Teigre, Hendrik Scholz, <http://tutpro.com/>Juha Heinänen, Klaus Darilion, Maxim Sobolev, Peter Nixon, Uli Abend. Great support on the mailing list has been provided by Greger V. Teigre, Java Rockx, Klaus Darilion, Zeus Ng, and more. Simply, this list is too short and incomplete to capture all the generous contributions that came in -- my appologies for everyone else I have missed. Warnings and Disclaimers ------------------------ This is a TESTING version. We are encouraging people to give it intense testing and provide us with feedback, so that we can properly assess what it takes to declare it final. The volume and number of changes is big, caution is advised. All GPL warnings and disclaimers apply. This is SER, which is not the derivate software known as openser. We have no ideas of in what extent and timeline derivate projects follow (or not) SER. ------------------------------------------------------------------------------- To Give it a Try ----------------- warning: this is just a 'jump-start' for black-belt users. It is not a replacement for documentation or a detailed step-by-step installation guide. It assumes knowledge of operating system, SIP, and SER. PREPARATION - make sure mysql is running. In the default configuration, SER assumes it to be running on localhost, if this does not hold, you need to change ser.cfg. - make sure python is installed (needed for new command-line utilities) SER INSTALLATION FROM CVS - Download from CVS/HEAD using instructions at http://www.iptel.org/downloads - Compile it: make include_modules="mysql xmlrpc" all - Install it the default way of your administrative privileges allow to: make include_modules="mysql xmlrpc" install - Download command-line tool (see more documentation under http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/serctl/README?rev=…) - cvs co serctl - Initiate database: ./scripts/mysql/ser_mysql.sh create START - Start it. Call: ser - Begin provisioning users with the new serctl (the capital word are just place holders) # ser_ctl domain add DOMAINNAME # ser_ctl user add USERNAME@DOMAINNAME -p PASSWORD # If you want to have Asserted Identity header for your user # - ser_attr add uid=UID asserted_id="PID" # If you want to have gateway support # - ser_db add attr_types name=gw_ip rich_type=string raw_type=2 \ description="The gateway IP for the default ser.cfg" default_flags=33 # - ser_attr add global gw_ip=GATEWAY-IP - more things you may wish to do: - familiarize yourself with current ser.cfg - send us feedback. -- Jiri Kuthan http://iptel.org/~jiri/ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers [View Less]

1 0

[Serusers] Problem with mysql module [mysql.so] file
by tzieleniewski 06 Nov '06

06 Nov '06

Hi all I've installed ser as it is written in the Getting started manuall with ser source 0.9.7-pre1. I've also installed the mysql5.0 binary distribution. unfortunatelly i've encountered a strange problem. At think that this may be caused by the fact that i used o binary distribution od mysql instead of source. Any way i'm just starting my adventure with ser:) so i ask u ser guru's to help me;) When i tried to compile the source again including mysql package the mysql.co file did not … [View More]appeared? and the log contained only such lines regarding mysql: make[1]: Wej��cie do katalogu `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' make[1]: Opuszczenie katalogu `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' make[1]: Wej��cie do katalogu `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -ftree-vectorize -mtune=athlon64 -DNAME='"ser"' -DVERSION='"0.9.7-pre1"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -I/usr/local/include/mysql -I/usr/local/mysql/include -I/usr/include/mysql -c dbase.c -o dbase.o make[1]: Opuszczenie katalogu `/home/tezet/apps/ser/ser-0.9.7-pre1/modules/mysql' When i tried to compile the package manually using Makefile in the "source"/modules/myqsl directory i got plenty of errors:): dbase.c:34:25: error: mysql/mysql.h: Nie ma takiego pliku ani katalogu dbase.c:35:26: error: mysql/errmsg.h: Nie ma takiego pliku ani katalogu In file included from dbase.c:39: val.h:44: error: expected ��)�� before ��*�� token In file included from dbase.c:40: my_con.h:39: error: expected specifier-qualifier-list before ��MYSQL_RES�� dbase.c: In function ��submit_query��: dbase.c:67: error: ��struct my_con�� has no member named ��timestamp�� dbase.c:68: warning: implicit declaration of function ��mysql_ping�� dbase.c:68: error: ��struct my_con�� has no member named ��con�� dbase.c:72: error: ��struct my_con�� has no member named ��timestamp�� dbase.c:90: warning: implicit declaration of function ��mysql_query�� dbase.c:90: error: ��struct my_con�� has no member named ��con�� dbase.c:93: warning: implicit declaration of function ��mysql_errno�� dbase.c:93: error: ��struct my_con�� has no member named ��con�� dbase.c:94: error: ��CR_SERVER_GONE_ERROR�� undeclared (first use in this function) dbase.c:94: error: (Each undeclared identifier is reported only once dbase.c:94: error: for each function it appears in.) dbase.c:94: error: ��CR_SERVER_LOST�� undeclared (first use in this function) dbase.c:98: warning: implicit declaration of function ��mysql_error�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:98: error: ��struct my_con�� has no member named ��con�� dbase.c:98: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c: At top level: dbase.c:138: error: expected ��)�� before ��*�� token dbase.c:166: error: expected ��)�� before ��*�� token dbase.c:207: error: expected ��)�� before ��*�� token dbase.c: In function ��store_result��: dbase.c:304: error: ��struct my_con�� has no member named ��res�� dbase.c:304: warning: implicit declaration of function ��mysql_store_result�� dbase.c:304: error: ��struct my_con�� has no member named ��con�� dbase.c:305: error: ��struct my_con�� has no member named ��res�� dbase.c:306: warning: implicit declaration of function ��mysql_field_count�� dbase.c:306: error: ��struct my_con�� has no member named ��con�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c:311: error: ��struct my_con�� has no member named ��con�� dbase.c:311: warning: format ��%s�� expects type ��char *��, but argument 3 has type ��int�� dbase.c: In function ��db_free_result��: dbase.c:348: warning: implicit declaration of function ��mysql_free_result�� dbase.c:348: error: ��struct my_con�� has no member named ��res�� dbase.c:349: error: ��struct my_con�� has no member named ��res�� dbase.c: In function ��db_query��: dbase.c:398: warning: implicit declaration of function ��print_where�� dbase.c:398: error: ��struct my_con�� has no member named ��con�� dbase.c: In function ��db_insert��: dbase.c:471: warning: implicit declaration of function ��print_values�� dbase.c:471: error: ��struct my_con�� has no member named ��con�� dbase.c: In function ��db_delete��: dbase.c:516: error: ��struct my_con�� has no member named ��con�� dbase.c: In function ��db_update��: dbase.c:559: warning: implicit declaration of function ��print_set�� dbase.c:559: error: ��struct my_con�� has no member named ��con�� dbase.c:568: error: ��struct my_con�� has no member named ��con�� make: *** [dbase.o] B��d 1 [View Less]

2 1

Re: [Users] Messege is too big Error Code 513
by Rao Ramaratnamma 06 Nov '06

06 Nov '06

I just began to study the differences between SER and OPENSER. So I am not sure it is the same and it is not some OPENSER related error but the SER doc says this: http://old.iptel.org/ser/doc/seruser/seruser.html#AEN1039 -rr P.S.: viva google ----- Original Message ---- From: ShivaSankar Padhy <shivapd1983(a)gmail.com> To: users(a)openser.org Sent: Monday, November 6, 2006 10:35:42 AM Subject: [Users] Messege is too big Error Code 513 Dear all! I am still new to the openser. Let me … [View More]

2 1

[Users] Not Hung up Properly
by raviprakash sunkara 06 Nov '06

06 Nov '06

Hello User, A small Question , When I My Sip Server without any firewall_ Router .. Hung up is working Fine,,,,,, But I Keep Sip Server with in a Firewall_router, it not Hung upping , When i saw logs from X-lite in Bye method it Changing the uri into Private IP, But should it public IP( THis is not happening ) Please help me............. -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)… [View More]

1 0

[Users] OT: sipit19 conclusions
by Klaus Darilion 06 Nov '06

06 Nov '06

Hi! I found this link on the sip mailing list. http://www.sipit.net/report19.txt regards klaus -- Klaus Darilion nic.at

3 3

[Serusers] t_replicate for REGISTERs
by SER LIST 06 Nov '06

06 Nov '06

I am replicating a REGISTER from the SER PROXY (0.9.6) to a presence server (0.10.99) using t_replicate. On the presence server I am doing a save_noreply("location"). Everything works fine but the only "issue" I have is that SER PROXY sends the REGISTER to the presence server THREE TIMES. It seems that since SER PROXY is not receiving a reply to the REGISTER sent using t_replicate, it is re-transmitting the REGISTER. I just do not think that this is "clean". I could have the presence … [View More]

1 0

RE: [Serusers] Ser with remote mysql
by John Shellhouse 05 Nov '06

05 Nov '06

Hi, You might want to make sure that db_url parameter is set correctly, because i believe by default the param is set to point to localhost. j >From: ram <talk2ram(a)gmail.com> >To: serusers(a)iptel.org >Subject: [Serusers] Ser with remote mysql >Date: Sun, 5 Nov 2006 22:33:06 +0530 > >Hi all > >I have Ser running one server ( centos) >and another server Database ( FC5) , Mysql mysql-5.0.X > > >iam able to connect from ser server to mysql server … [View More]

1 0

[Serusers] Ser with remote mysql
by ram 05 Nov '06

05 Nov '06

Hi all I have Ser running one server ( centos) and another server Database ( FC5) , Mysql mysql-5.0.X iam able to connect from ser server to mysql server from prompt iam able to connect. wheni configure in ser.cfg and run ,iam getting this following error. why this still conencting to local server ? Nov 5 22:26:11 ser2 ser[22959]: Maxfwd module- initializing Nov 5 22:26:11 ser2 ser[22959]: new_connection(): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (… [View More]

3 2

Re: [Serusers] Ser with remote mysql
by Atle Samuelsen 05 Nov '06

05 Nov '06

ALWAYS REPLY TO THE LIST. The problem Im guessing your runnin ginto is that the serro user is still trying to use localhost as DB host. Check in config.h. -A * ram <talk2ram(a)gmail.com> [061105 19:44]: > hi > > what problems ? > > Ram > > > On 11/5/06, Atle Samuelsen <clona(a)cyberhouse.no> wrote: > > > >Hi, > > > >Without knowing, I'm betting your getting into tourble related to serro > >user. > > > >- Atle … [View More]

1 0

[Users] Fw: [Serusers] rewriting From
by Rao Ramaratnamma 05 Nov '06

05 Nov '06

Can possibly the openser folks give me some answer? ----- Forwarded Message ---- From: Rao Ramaratnamma <raramarat(a)yahoo.com> To: serusers(a)iptel.org Sent: Saturday, November 4, 2006 4:34:55 PM Subject: [Serusers] rewriting From Hello, I am new to the community so please forgive me if I am missing some obvious answer but till now I have found only some suggestions what one could or could not do, but is there any known-to-work proven configuration file which rewrites From? rr … [View More]

2 1

[Serusers] Service Provider Solution
by John Shellhouse 05 Nov '06

05 Nov '06

Hello Everyone, I have finally up ser up and its running well, tested it to be true -thus far - with asterisk backend w/PSTN. Now, i want to start offering services, but only have one pstn channel. I was hoping that someone could advice me on a service provider that is willing to allow me to route calls to their sip router for a monthly fee, and if this is possible so I can offer services to users without being limited to a single channel and see how how sip really works in action … [View More]

1 1

[Users] Sip Provider
by John Shellhouse 05 Nov '06

05 Nov '06

Hello Everyone, I have finally got my sip router up and its running well, tested it to be true -thus far - with asterisk backend w/PSTN. Now, i want to start offering services, but only have one pstn channel. I was hoping that someone could advice me on a service provider that is willing to allow me to route calls to their sip router for a (monthly) fee, and if this is possible so I can offer services to users without being limited to a single channel and then I can see how sip … [View More]

2 2

[Serusers] rewriting From
by Rao Ramaratnamma 05 Nov '06

05 Nov '06

Hello, I am new to the community so please forgive me if I am missing some obvious answer but till now I have found only some suggestions what one could or could not do, but is there any known-to-work proven configuration file which rewrites From? rr

2 1

[Users] Any trick to log ACK with siptrace?
by Rodrigo P. Telles 05 Nov '06

05 Nov '06

Hi Folks, Is there any trick to log ACK method using siptrace module? I'm trying with: ... modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) route { if (is_method("INVITE|ACK|BYE")) { setflag(22); }; ... but only INVITE and BYE method has been logged! I'm using OpenSER 1.1.0-notls. Thanks for any help Rodrigo Telles

2 1

[Users] Difference between method and is_method
by Rodrigo P. Telles 05 Nov '06

05 Nov '06

Hi Folks, Today I realised something strange in openser-1.1.0-notls. I'm trying to mach any ACKs to do something (e.g. logging) but its doesn't work at all. I'm using "method" like this: If (method=="ACK") { xlog("L_ALERT","ACK received\n"); }; and I it doesn't work, but... if (is_method("ACK")) { xlog("L_ALERT","ACK received\n"); }; works perfectly! I could not find anything about it in archive mail list neither in doc. Rodrigo P. Telles

2 1

[Users] Cisco-like Radius accounting
by Dimo 05 Nov '06

05 Nov '06

Hi, I have a radius accounting configuration I would like to share, if someone needs. Please can I get your feedback guys if this can be done in a better way. Just share your thoughts if you have done something similar. I need to account to my RADIUS the source IP (of the calling UA), the destination IP (called UA/gw) and the type of the leg - originating or terminating depending on whether the call is from the UA or to the UA. To do this I decided to use the cisco vendor-specific radius AVPs … [View More]which have similar fields but for h323 calls. In my dictionary file of radiusclient-ng i added this according to the cisco specification: VENDOR Cisco 9 ATTRIBUTE Cisco-AVPair 1 string Cisco ATTRIBUTE H323-remote-address 23 string Cisco ATTRIBUTE H323-call-origin 26 string Cisco and in my openser.cfg i have: modparam("acc", "radius_extra", "Cisco-AVPair=$avp(s:c_src_ip); H323-remote-address=$avp(s:c_dst_ip); H323-call-origin=$avp(s:c_origin); Cisco-AVPair=$avp(s:c_call_id)") Now in my config i fill in the values of the c_src_ip, c_dst_ip, c_origin, and c_call_id avps according to the direction of the call. Here is how i do it for the originating leg, terminating is similar: route[14] { #ACCOUNT ORIGINATING if (is_direction("downstream")) { avp_printf("$avp(s:c_src_ip)", "call-origin-endpt=$si"); avp_printf("$avp(s:c_dst_ip)", "$ru"); avp_subst("$avp(s:c_dst_ip)", "/sip:.*@(.*)/h323-remote-address=\1/g"); avp_subst("$avp(s:c_dst_ip)", "/(.*):.*/\1/g"); avp_printf("$avp(s:c_origin)", "h323-call-origin=originate"); avp_printf("$avp(s:c_call_id)", "call-id=$ci"); } if (is_direction("upstream")) { avp_printf("$avp(s:c_src_ip)", "$ru"); avp_subst("$avp(s:c_src_ip)", "/sip:.*@(.*)/call-origin-endpt=\1/g"); avp_subst("$avp(s:c_src_ip)", "/(.*):.*/\1/g"); avp_printf("$avp(s:c_dst_ip)", "h323-remote-address=$si"); avp_printf("$avp(s:c_origin)", "h323-call-origin=originate"); avp_printf("$avp(s:c_call_id)", "call-id=$ci"); } setflag(2); } Now as you can see i take one of the IP addresses from the request-uri (and strip it with avp_subst), which I am not sure if always contains IP. Is there a better way - i can call this route after rewritehostport or lookup so the destination IP will hopefully be in the r-uri of the message, but maybe there is a better way of getting it? Any input/ideas will be greatly appreciated. Best, Dimo [View Less]

3 3

[Users] VoN Berlin & OpenSER Summit
by Daniel-Constantin Mierla 05 Nov '06

05 Nov '06

Hello everybody, I will see many of you at Berlin, next week. The summit has a BoF session on Tuesday, 15:45-17:00 and continues next days with morning and afternoon sessions: see http://www.openser.org for more. Anyone who wants to get in contact before or to have a private meeting, the can find us at booth 321. All participants registered before November 1 got full VON conference pass (valid for the three days, November 6-8, for exhibition and conference, including special events, … [View More]

2 2

[Serusers] LCS+SER+ASTERISK
by Felipe Louback 04 Nov '06

04 Nov '06

Howdy all! I am trying to setup the LCS communicating with SER and ASTERISK in the company I work for. I read the following tutorial: http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Int… But there the author points the necessity of 2 LCSs, but I just couldnt grasp why the need of a second LCS. Could anyone who has already setup that give me a hint of why it is necessary a second LCS? I thought I could route LCS to communicated directly with SER. I tried to use only … [View More]

1 0

RE: [Serusers] how to search in usrloc?
by SER LIST 04 Nov '06

04 Nov '06

You do not need to check usrloc t to implement what you want. Define a timer and when you forward the INVITE to the PBX, you will automatically jump to the failure_route[] if you do not receive a response from the PBX (PBX down!) and you can put your logic there. >From: "sandrinho1976(a)libero.it" <sandrinho1976(a)libero.it> >To: "serusers" <serusers(a)lists.iptel.org> >Subject: [Serusers] how to search in usrloc? >Date: Fri, 3 Nov 2006 01:13:24 +0100 > >Hi … [View More]

2 1

Re: [serusers]: trusted table(permissions module)
by Andrey Kuprianov 04 Nov '06

04 Nov '06

The one you attached, didnt have a semicolon. Anyway, try using a fresh copy of a ready made ser.cfg. Bests, Andrey. On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > Yup I always do the same, Might be copy paste error. Its having semicolon & working fine here. > > Regards > Kamal Mann > > -----Original Message----- > From: Andrey Kuprianov [mailto:andrey.kouprianov@gmail.com] > Sent: Thursday, November 02, 2006 10:34 AM > … [View More]To: serusers(a)iptel.org > Subject: Re: [serusers]: trusted table(permissions module) > > Hi Kamal, > > Whenever you modified your ser.cfg did you restart SER after? Did you > check your ser.cfg for errors using "ser -c" command? I found some > errors in your ser.cfg code. For instance, inside the main route > block, your "if", which processes INVITEs, is missing a semicolon at > the end of the block. > > if (method=="INVITE") > { > # sl_send_reply("404", "INVITE "); > route(3); > break; > } <----- missing semicolon here > > > If you are comfortable with your current ser.cfg, that's ok, but I'd > suggest you start with a fresh one, make your trusted table work, and > then add a "REFER" processing, like in your currect ser.cfg. Just > download one fresh ser.cfg from iptel.org site and add your > allow_trusted() "if" clause there. > > Regards, > > Andrey. > > > On 11/2/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > Hi > > Please find ser.cfg enclosed. > > > > Trusted table entry: > > +---------------+-------+--------------+ > > | src_ip | proto | from_pattern | > > +---------------+-------+--------------+ > > | 10.25.119.156 | any | ^sip:.*$ | > > +---------------+-------+--------------+ > > 1 row in set (0.00 sec) > > > > Thanks in anticipation > > Kamal Mann > > -----Original Message----- > > From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kuprianov > > Sent: Wednesday, November 01, 2006 7:03 PM > > To: serusers(a)iptel.org > > Subject: Re: [serusers]: trusted table(permissions module) > > > > Hi Kamal, > > > > I understand your setup. Nevertheless, it will be difficult to figure > > out your problem, w/o your ser.cfg file at hand. Also, send a trusted > > table query result (i.e. that row which contains entry for your > > SIP-AS). > > > > Regards, > > > > Andrey. > > > > On 11/1/06, Kamal.Mann(a)t-systems.com <Kamal.Mann(a)t-systems.com> wrote: > > > Hi All > > > In my scenario I need ser to communicate with my SIP-AS. This SIP-AS is having an application xyz running on it. This XYZ sends an invite to URI "A(a)xcv.de" which is a subscriber in SER and this is in TO header of Invite msg & FROM header contains another URI test(a)xcv.de but this URI (test(a)xcv.de) isn't registered / subscribed to SER. I need SER trust all packets from SIP-AS ip. BUT SER is sending 407 reply of INVITE to XYZ application! IP in trusted table is of SIP-AS on top of which XYZ application is running. > > > > > > Thanks in anticipation > > > Kamal Mann > > > > > > -----Original Message----- > > > From: Maciej Żwirski [mailto:mzwirek@poczta.fm] > > > Sent: Wednesday, November 01, 2006 3:20 PM > > > To: serusers(a)lists.iptel.org > > > Subject: Re: [serusers]: trusted table(permissions module) > > > > > > Kamal.Mann(a)t-systems.com wrote: > > > > > > > Hi All > > > > > > > > I made an entry /"10.25.119.156, any, ^sip:.*$"/ into trusted table so > > > > that any packet sent from 10.25.119.156 would be trusted by SER and > > > > don't ask for its credentials. But SER is replying *407- proxy* auth > > > > needed!! After this I tried with permissions.allow file: > > > > > > > Hi, > > > I had the same issue while interconnecting SER and Asterisk. It turned > > > out to be an issue on Asterisk side (I had the same user added for SER > > > and Asterisk, so the poor thing couldn't authorize either :)) So you > > > could check if the 407 you're getting is from SER or from the remote host. > > > > > > Regards, > > > Maciej Zwirski > > > > > > ---------------------------------------------------------------------- > > > Jestes kierowca? To poczytaj! >>> http://link.interia.pl/f199e > > > > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > [View Less]

2 2

[Users] State of T.38 and MediaProxy/RTPProxy
by Bastian Schern 04 Nov '06

04 Nov '06

Hello, does anybody knows the state of T.38 with MediaProxy or RTPProxy/NATHelper? Is it supported in MediaProxy v1.7.2 or the current RTPProxy(NATHelper release? Regards Bastian

4 6

[Users] Syntax Error
by Brandon Armstead 03 Nov '06

03 Nov '06

Hello guys, I have been trying to get openser implemented with SEMS the past day or so although I keep running into problems, any advice on setting up openser with SEMS? (Sip Express Media Server). Although here is a syntax error straight from the documentation: ... modparam("tm","tw_append","append1:Email=avp[i:12];UA=hdr[User-Agent]") modparam("tm","tw_append","append2:body=msg[body]") ... t_write_fifo("voicemail/append1","/tmp/appx_fifo"); ... t_write_unix("logger/append2","/var/run/… [View More]

1 0

[Users] Parallel calls with multiple contacts for 1 user
by Jean-François SMIGIELSKI 03 Nov '06

03 Nov '06

Hello, list! I currently use the debian package of OpenSER (v1.1.0-5), and I see a behaviour I do not explain with RFC's and OpenSER Documentation. I have registered several User-Agent under the same username/domain couple in an OpenSER instance with the registrar module. When I try to call "username@domain", I expected OpenSER to call the contact with the highest q-value, and keep the other contacts into transaction branches for later use (as said in the usrloc module's documentation, in the … [View More]

2 1

[Users] Syntax Error
by Brandon Armstead 03 Nov '06

03 Nov '06

Hello, I get syntax error on the following two commands, does anyone have any idea, thanks: avp_db_load("$ru", "i:34/$email_scheme") modparam("tm", "tw_append", "vm_email: P-Email-Address=avp[i:34]") -- Brandon Armstead

2 1

[Serusers] Multiple user registrations using same port number
by Alan 03 Nov '06

03 Nov '06

I am currently running load testing on SER and have a question regarding multiple contacts registered with the same IP and port number but different contact name. Will this work? Does each contact ID have to be defined with a different port number to establish a session? For example: Contact : sip:loadtest0100@10.180.0.71:10140 Contact : sip:loadtest0101@10.180.0.71:10140 Contact : sip:loadtest0102@10.180.0.71:10140 Contact : sip:loadtest0103@10.180.0.71:10140

2 1

[Users] About forking and unregister
by Bill Zhang 03 Nov '06

03 Nov '06

Hi, Anyone know how openser works with following: 1. Forking. Often times, I may have one account register with multiple SIP client, does it support that? I don't seem to make it working. 2. Unregister and register. Nowadays, there are a lot of SIP client will send an Unregister message to the SIP server and after received response, it will send the new registration again, this is because when a client behind the same router reboot multiple times, it might have registration in the … [View More]

1 0

[Users] mediaproxy pulse / jitter and dropout
by Mark Price 03 Nov '06

03 Nov '06

I installed the openser.cfg at http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy (modified to authenticate against my domain, and to use exit instead of break). Calls between two phones on the domain work, but there is a lot of jitter. Well, not exacty jitter, but kind of a pulsing sound in the background of the call. Also, will occasionally drop out, and then come back after a little bit.

1 0

[Serusers] how to search in usrloc?
by sandrinho1976＠libero.it 03 Nov '06

03 Nov '06

Hi everyone. I'm facing this problem when configuring a Ser as a proxy and a registrar. I have the Ser and a IP pbx: the pbx registers itself on the proxy as pbx@<pbx_ip> (Contact header). I had to manage INVITEs messages for incoming call that are going towards the Ser, sent by UACs in the network. INVITEs are coming in the form <telephone_number>@<ser_ip> (Req-URI): Ser had to decide if <telephone_number> is managed by the IP pbx (it has this information) and … [View More]

1 0

[Users] two invites, different session description => broken call
by Mark Price 02 Nov '06

02 Nov '06

I have openser configured with asterisk as following: If a sip REGISTER packet arrives, fix_nated_register() save("location") exit; If a sip packet arrives at openser with a source ip != asterisk ip, t_relay(to asterisk) If a sip packet arrives from asterisk, lookup("location"); force_rport() fix_nated_contact() t_relay(); The asterisk dial plan says the following: if inbound traffic arrives from openser's ip address, Dial( SIP/dstuser@dstdomain). … [View More]

1 0

Re: [Users] TLS: what is encrypted in TLS?
by Ferianto siregar 02 Nov '06

02 Nov '06

Dear Witt, Thank you very much for your help before. Witt, may I ask you again,please? I have downloaded ssldump.tar.gz packet and installed it. As the INSTALL note in ssldump, I have installed libcap.tar.gz (from www.tcpdump.org) and openssl. I have installed it by using commands: # tar -zxvf ssldump-0.9b3.tar.gz # cd ssldump-0.9b3 # ./configure --with-openssl #make #make install All installation process can run sucessfully (nothing error message shown). But, … [View More]why when I try to run ssldump, I got nothing. There is any mesage shown when I tried to run #ssldump -i eth0 What happens? Did I do wrong installation process? or I have missed something? I do hope you can help me.Please help me.. Thank you very much for your help Regards, Ferianto Note: Actually, when I run ./configure --with-pcap, I got error message. The error message said that there is no pcap file. Why? Is lipcap packet that I got from www.tcpdump.org (as I read from ssldump INSTALL), same with pcap file ? or I have to download pcap ? Steffen Witt <witt.steffen(a)googlemail.com> wrote: Hello Ferianto, TLS is used for encrypting your SIP signaling, not the media data. You should install "ssldump" on your server to see each step of the TLS handshake. Maybe you have a closer look to this short SSL tutorial: http://www.eventhelix.com/RealtimeMantra/Networking/SSL.pdf Best regards Steffen 2006/10/30, Ferianto siregar : > Dear all, > > Thank you for this chance. > All, I have tried to learn about how can TLS can secure communication in > VoIP. > I must to understand it in order to responsible my openser server (working) > with TLS to my lecturer. > But, I have still doubt. > Actually, what is encrypted in TLS system so the communication become > secure? Does it encrypt the data (voise packet)? or The traffic (so the > sniffer can not get the data? OR anything else? > > I do hope anybody can help me :) > Thanks, > > > Ferianto > > > Note: I cryptography tutorials, I read that the cryptography system secure > the message (text) by encrypting it using public key and private key. So, > How about TLS in VoIP? > > > ________________________________ > Everyone is raving about the all-new Yahoo! Mail. > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > --------------------------------- Get your email and see which of your friends are online - Right on the new Yahoo.com [View Less]

1 0

[Users] sarge didnt install avp_radius
by Zoran Milic 02 Nov '06

02 Nov '06

Hi, I changed my linux distro to debian (from slack) so i can install openser from packages, but aptitude failed to install avp_radius. When I install from source, avp_radius.so is created, after tweaking the Makefile. What is the problem, and how do i get avp_radius module when i install from aptitude/apt-get/dpkg ? best regards, Zoran

2 7

[Users] determining originating address
by Mark Price 02 Nov '06

02 Nov '06

What is the best way to detect whether an INVITE message came from a certain IP address? I.e. if a call comes from a certain asterisk box? I have thought of using the textops search(), but this seems a little clumsy. Perhaps there is a way to get directly at the via headers? Or is there a keyword of corresponding functionality to the uri keyword? Thanks, Mark Price

3 2

[Serusers] please help me out
by vijay tiwari 02 Nov '06

02 Nov '06

hi all i have problem in configurations of mediaproxy module i have download mediaproxy in my ser server and i have make one file name(mediaproxy.ini) in mediaproxy dir and do some change and run this command mediaproxy start nothing can heppand please help me out on this because at this time i am in very bad conditions thanks vijay --------------------------------- Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.… [View More]

1 0

[Serusers] No Media with Nated Client and rtp proxy
by Ryan Churches 02 Nov '06

02 Nov '06

I just posted about a problem with rtpproxy, but i was able to correct that. now ser is able to connect to rtpproxy, but media is still being lost. someone suggested i put my external ip in the listen= value at the top, but that breaks remote registration. I am following the rtpproxy section of the "Getting Started with Ser Docs" http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html so one would expect i would have ser.cfg set up correctly, but obviously something is wrong. here is … [View More]some ngrep output. 10.0.0.10 is SER. 10.0.0.110 is a client LOCAL to ser, and 10.0.0.100/67.84.215.54 is a remote client behind yet another nat U 10.0.0.10:6060 -> 10.0.0.110:9267 INVITE sip:1000@10.0.0.110:9267 SIP/2.0. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. To: <sip:1000@bmy.gotdns.org>. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Contact: <sip:1001@67.84.215.54:9170>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 3004w stamp 16863. Content-Length: 285. . v=0. o=- 105355536 105355589 IN IP4 10.0.0.100. s=eyeBeam. c=IN IP4 10.0.0.10. t=0 0. m=audio 20026 RTP/AVP 100 6 0 8 3 18 5 101. a=alt:1 1 : 7D3DDAD1 578F25CA 10.0.0.100 9172. a=fmtp:101 0-15. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=nortpproxy:yes. # U 10.0.0.110:9267 -> 10.0.0.10:6060 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1;received=10.0.0.10. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. . # U 10.0.0.10:6060 -> 67.84.215.54:9170 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. . # U 10.0.0.110:9267 -> 10.0.0.10:6060 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.10:6060;branch=z9hG4bK44a9.9b2912a1.1;received=10.0.0.10. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. . # U 10.0.0.10:6060 -> 67.84.215.54:9170 SIP/2.0 180 Ringing. To: <sip:1000@bmy.gotdns.org>;tag=703e0d1b. From: <sip:1001@bmy.gotdns.org>;tag=d752304f. Via: SIP/2.0/UDP 10.0.0.100:9170;received=67.84.215.54;branch=z9hG4bK-d87543-71698269-1--d87543-;rport=9170. Call-ID: cc766a6d3e104232. CSeq: 2 INVITE. Record-Route: <sip:10.0.0.10:6060;ftag=d752304f;lr=on>. Contact: <sip:1000@10.0.0.110:9267>. Content-Length: 0. and here is my ser.cfg which ive also attached. # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=10.0.0.10 alias=hostname.dyndns.org port=6060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://user:pw@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #From Getting Started modparam("auth_db|uri_db|usrloc", "db_url", "mysql://user:pw@localhost/ser") # ----------------- setting module-specific parameters --------------- # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; }else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; #======= call type processing section if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" || isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { #------------------------- #REGISTER Message Handler #------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { #---------- #--INVITE Message Handler #----------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { #NAT Traversal if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Contact-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } [View Less]

1 0

[Users] 484 UAC handeling
by Helge Waastad 02 Nov '06

02 Nov '06

Hi, I was wondering about the correct handeling of an 484 address incomplete message. I've seen different handeling from UAC to UAC Problem: My grandstreams receives a 484 and returns an ACK. This is how it should.And i hear beep-beep tones. the moment I terminate (on hook), the UAC sends a CANCEL message which generates a baaaad loop in my system.... 1. Is this cancel according to RFC? 2. Is there a way to consume/drop such an cancel? br hw -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

2 2

[Serusers] Presence ( @msg.supported=~"eventlist")
by SER LIST 02 Nov '06

02 Nov '06

Can someone please explain what the line if ((@msg.supported=~"eventlist")) is testing in the presence example code? This lines always returns false for me and seemingly the XCAP server is not being queried then. Thanks _________________________________________________________________ Ready for the world's first international mobile film festival celebrating the creative potential of today's youth? Check out Mobile Jam Fest for your a chance to WIN $10,000! www.mobilejamfest.com

2 1

[serusers]: trusted table(permissions module)
by Kamal.Mann＠t-systems.com 02 Nov '06

02 Nov '06

Hi All I made an entry "10.25.119.156, any, ^sip:.*$" into trusted table so that any packet sent from 10.25.119.156 would be trusted by SER and don't ask for its credentials. But SER is replying 407- proxy auth needed!! After this I tried with permissions.allow file: "ALL : "sip:mann@imsfw.de$"" but still no progress same response from SER. My ser.cfg entries: modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "source_col",… [View More]

3 8

[Serusers] WARNING: rtpp_test: can't get version of the RTP proxy
by Ryan Churches 02 Nov '06

02 Nov '06

Im following "Getting Started with SER" and I'm stuck on the chapter about nat traversal Ive tried both methods, currently im working on rtpptoxy. the exact errors I get when i try to make calls from behind NAT are: 0(1731) WARNING: rtpp_test: can't get version of the RTP proxy 0(1731) WARNING: rtpp_test: support for RTP proxy has been disabled 0(1731) ERROR: force_rtp_proxy2: support for RTP proxy is disabled i have compiled and installed rtpproxy, and made sure the permissions were set. … [View More]bmy bin # ps ax|grep rtp 1714 ? Ss 0:00 ./rtpproxy bmy bin # ls -la /var/run/rtpproxy.* -rw-r--r-- 1 root root 5 Nov 1 19:27 /var/run/rtpproxy.pid srwxr-xr-x 1 root root 0 Nov 1 19:27 /var/run/rtpproxy.sock i have attached my ser.cfg but below are some of the parts of it debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=10.0.0.10 alias=hostname.dyndns.org port=6060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://user:pw@localhost/ser" <skip> modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") <skip> 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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # 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(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; }else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; #======= call type processing section if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" || isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { #------------------------- #REGISTER Message Handler #------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { #---------- #--INVITE Message Handler #----------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { #NAT Traversal if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Contact-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } [View Less]

1 0

[Serusers] Voice problem
by ram 01 Nov '06

01 Nov '06

Hi all I have installed CDROUTER working fine for 30days now almost all iam able to see 120 users calling at a time no problem but past 7days iam getting lot of voice breaking problems and other side not able to hear my side voice my Server config Xeon with 1 GB Ram 86GB hdd and most of the time looks like below ~# free total used free shared buffers cached Mem: 903484 897328 6156 0 1892 830208 -/+ buffers/cache: … [View More]

1 0

[Serusers] Fwd: help in mediaproxy module
by vijay tiwari 01 Nov '06

01 Nov '06

Note: forwarded message attached. Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Check out the New Yahoo! Mail - Fire up a more powerful email and get things done faster.

1 0

[Serusers] my mail are not coming on mail list
by vijay tiwari 01 Nov '06

01 Nov '06

hi all problem my mail are not coming on mail list Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.

1 0

[Users] problem with usrloc db-only mode
by tele＠plexia.com 01 Nov '06

01 Nov '06

Hi all, A simple question about the db-only mode of usrloc (dbmode 3) I've two proxy configured with db_mode 3 the second proxy it's unable to do lookup on the db, there are no error in the debug log but it send reply "404 not found" to the first proxy. I've noticed that when i do lookup("location") on the second proxy it clean the location table and even if the table is filled with the correct contact it reply with 404. sometimes delete all contacts and sometimes contacts remain but failed … [View More]

2 2

Re: [Users] Implementing the CDS(Call Detail Record)
by Bogdan-Andrei Iancu 01 Nov '06

01 Nov '06

1) be sure you are using TM for request forwarding (r_relay) 2) be sure the *db* acc flag is set (and not the one for syslog acc) - see the README 3) look into logs for possible errors. regards, bogdan ShivaSankar Padhy wrote: > > Yah, The same thing now I am looking but also bit confused, I have > used the > > modparam("acc", "log_flag", 2) > and setflag(2) > but I am not able to populate the acc table. > > What is the exact syntax, Can I get it from any where??… [View More]

1 0

[Users] send im in php
by sinclair fils 01 Nov '06

01 Nov '06

when i send im in a script php i have this error 2(3549) ERROR: fifo_server: command must begin with :: MESSAGE 2(3549) **** done consume 2(3549) ERROR: fifo_server: command must have at least 3 chars can help me --------------------------------- Découvrez une nouvelle façon d'obtenir des réponses à toutes vos questions ! Profitez des connaissances, des opinions et des expériences des internautes sur Yahoo! Questions/Réponses.

2 1

[Users] New presence server: immediate NOTIFY missing
by Franz Edler 01 Nov '06

01 Nov '06

Hello, many thanks to all developers of the new presence server. Its a great enhancement to OpenSER. As a small contribution from my side I did some tests with Eyebeam clients. Now I found, that the immediate NOTIFY according to RFC3265 § 3.1.6.2 is missing. To document the behaviour I have attached a trace-file from the server. Frame 6 contains a SUBSCRIBE request which is not confirmed with an immediate NOTIFY request. The first NOTIFY is sent only four minutes later after the first … [View More]

4 9

[Users] TLS: what is encrypted in TLS?
by Ferianto siregar 01 Nov '06

01 Nov '06

Dear all, Thank you for this chance. All, I have tried to learn about how can TLS can secure communication in VoIP. I must to understand it in order to responsible my openser server (working) with TLS to my lecturer. But, I have still doubt. Actually, what is encrypted in TLS system so the communication become secure? Does it encrypt the data (voise packet)? or The traffic (so the sniffer can not get the data? OR anything else? I do hope anybody can help me :) Thanks, Ferianto Note: I … [View More]

2 1

[Serusers] on hold problem
by Shaun Hofer 01 Nov '06

01 Nov '06

Hi, I've been having a problem, where audio is lost either in one or both directions when conversaion is taken off 'hold'. The parties are both behind NAT and depending UA as whether one or both loose audio. From what I can tell its to do with my loose route and nathelper, and how my ser.cfg deals with the take off hold INVITE from the phones. When the call is taken off hold the rtp streams aren't setup properly again (eg not using mediaproxy correctly). What is the best way to solve this … [View More]

3 8

[Users] cpledit
by Doug Meredith 31 Oct '06

31 Oct '06

Hi, I'm trying to find cpledit. It is no longer at iptel.org. Someone on the SER mailing list told me to try here, as he thought Bogdan was behind it. Does anyone know if this still exists and where I can find it? Can anyone suggest an alternative? Thanks for any help. Doug -- Doug Meredith 506-854-7997 ext. 801

2 2

[Serusers] test message
by SER LIST 31 Oct '06

31 Oct '06

pls. ignore _________________________________________________________________ Not only does Windows Live OneCare provide all-in-one PC care to keep your computer protected and well-maintained, but it also makes creating backup files a breeze. Try it today! http://ideas.live.com/programpage.aspx?versionid=b2456790-90e6-4d28-9219-5d…

1 0

[Users] Implementing the CDS(Call Detail Record)
by ShivaSankar Padhy 31 Oct '06

31 Oct '06

Dear All, Can I implement the CDR without using the RADIUS Server.??I am using openser 1.0.1 notls. Expecting a sooner reply, Thanks in Advance, Shiva

2 1

[Serusers] help in mediaproxy module
by vijay tiwari 31 Oct '06

31 Oct '06

hi all i have problem in configurations of mediaproxy module i have download mediaproxy in my ser server and i have make one file name(mediaproxy.ini) in mediaproxy dir and do some change and run this command mediaproxy start nothing can heppand please help me out on this because at this time i am in very bad conditions thanks vijay --------------------------------- Everyone is raving about the all-new Yahoo! Mail.

1 0

[Users] CVS 31.03, TLS and dead tcp child
by Frank DInnocenzo 31 Oct '06

31 Oct '06

Hello, I'm new to openSer and wondered if there was any resolution to this post??? I'm hitting this same crash with openSer (with TLS enabled) using a Snom phone (300 with latest Beta firmware - snom300-6.5-SIP-j.bin). I've gone through the archives looking for help getting this setup working. I've also tried the Snom Softphone with the same result. Thanks Frank >>>>>>>>>>>>>>>>>>>>>>>>>>>>&… [View More]

2 3

[Users] Re: openser
by samuel 31 Oct '06

31 Oct '06

First of all: It's not considered polite to send direct mails for topics which ought to be directed to the mailing list... Try to go to www.onsip.org and download the getting started document, which has a good explanation and several examples for configuring. It is targeted to SER but your openser version should be really close to the config files explained in the getting started document...You might have to adapt some syntax so you would have to check the logs when you start openser. Samuel. … [View More]

1 0

[Serusers] No query limit ?
by inge 31 Oct '06

31 Oct '06

Hi all, I find a way to show all result for accounting page. I don't want to split it in some pages. So I find in "method.get_acc_entries" the parameter. I write false instead of true. Then I see all entries for the query, but there is always the split in some pages. So for each pages I have all result (lol). Someone have an idea ? Regards.

1 0

[Serusers] SER with a Gateway
by inge 31 Oct '06

31 Oct '06

Hi all, We use SER with a PSTN gateway. My problem is : Although I see my caller ID in result of "show isdn history", my interco provider says that he must put the principal number by default because we don't put the good signalisation for ETSI PRI with the caller ID. That's why on the calling phone, it's not the caller id wich is show on. Someone have an idea ? Is it a problem from SER or Gateway according to you ? (I try to contact Cisco support for this problem too). Regards

1 0

[Serusers] Core dump Presence snapshot 0.10.99-dev35-pa-4.1
by Dave 31 Oct '06

31 Oct '06

The version 0.10.99 of presence snapshot core dumps on a regular basis after running for a certain time. Doing a gdb shows that the core dump occurs at msg_queue_destroy: #0 0x001b87f2 in msg_queue_destroy (q=0xb6293e98) at /usr/local/src/ser-0.10.99-dev35-pa-4.1/lib/../fastlock.h:147 147 asm volatile( Doing a backtrace shows: (gdb) backtrace #0 0x001b87f2 in msg_queue_destroy (q=0xb6293e98) at /usr/local/src/ser-0.10.99-dev35-pa-4.1/lib/../fastlock.h:147 #1 0x008e0a52 … [View More]

2 1

[Serusers] serctl domain show ERROR
by lin9k＠yahoo.com 31 Oct '06

31 Oct '06

Hi, When I tried to execute "serctl domain show" but it complained "500 command domain_dump not available". What did I do wrong? Thanks in advance, Ling ____________________________________________________________________________________ Access over 1 million songs - Yahoo! Music Unlimited (http://music.yahoo.com/unlimited)

2 2

[Users] parse_via, bad via?
by Dominik Bay 30 Oct '06

30 Oct '06

Hi, i'm getting sometimes this message: ERROR: parse_via: bad char </> on state 100 ERROR: parse_via on: </UDP ^M ^M > ERROR: parse_via: via parse error ERROR: get_hdr_field: bad via ERROR: bad header field ERROR: parse_msg: message=<SIP/2.0 404 Not Found^M Via: /UDP ^M ^M > ERROR: receive_msg: parse_msg failed it seems that a UA sends ^M which is wrong and openser doesn't support it. Is there any workaround for this? -- Mit freundlichen Gruessen / Kind regards Dominik Bay Cablesurf Technik

1 1

[Serusers] Core dump Presence snapshot 0.10.99-dev35-pa-4.1
by Dave 30 Oct '06

30 Oct '06

The version 0.10.99 of presence snapshot core dumps on a regular basis after running for a certain time. Doing a gdb shows that the core dump occurs at msg_queue_destroy: #0 0x001b87f2 in msg_queue_destroy (q=0xb6293e98) at /usr/local/src/ser-0.10.99-dev35-pa-4.1/lib/../fastlock.h:147 147 asm volatile( Doing a backtrace shows: (gdb) backtrace #0 0x001b87f2 in msg_queue_destroy (q=0xb6293e98) at /usr/local/src/ser-0.10.99-dev35-pa-4.1/lib/../fastlock.h:147 #1 … [View More]

1 0

[Users] Does anyone use CPL-C module?
by Doug Meredith 30 Oct '06

30 Oct '06

Hi, I'm quite interested in CPL but I see almost no discussion of this on the list. Is anyone actually using this module? Is it worth looking at? Thanks for any info you can give me. Doug -------------------------------------------- Doug Meredith SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com

3 3

[Serusers] BYEs are not written to acc if gateway sends BYEs resulting from RTP timeout
by Ricardo Carvalho 30 Oct '06

30 Oct '06

Hi all, I've implemented RTP timeout in the gateway that I use along with Ser, to be able to overpass problems of wrong accounting resulting from power or network failure of some UA. This is apparently working fine as my gateway sends to Ser BYE messages after the established timeout, although only now I realised that Ser doesn't account those BYE messages!!! Why? Following ser.cfg code, I can see that the execution reads the setflag(1) that I use for accounting, but it seems that Ser … [View More]

1 1

Fwd: [Users] Two Invites simultaneously
by ShivaSankar Padhy 30 Oct '06

30 Oct '06

Thank you Vamsi, The problem is for the saving(location) twice, Thanks a lot for cooperating and help me in solving the problem. Thanks, Shiva ---------- Forwarded message ---------- From: Vamsi Pottangi <vamsipottangi(a)gmail.com> Date: Oct 30, 2006 5:32 PM Subject: Re: [Users] Two Invites simultaneously To: ShivaSankar Padhy <shivapd1983(a)gmail.com> Thats good. I feel you should reply to the mailing list saying that problem is solved and it is a mistake in cfg where you were … [View More]saveing the location twice. This will help people later who would face similar problem. ~Vamsi On 10/30/06, ShivaSankar Padhy <shivapd1983(a)gmail.com> wrote: > > > I am extremely Sorry, Actually, I was having Authentication problem, thats > why I have used the Same(please refere to the commented lines near by > save(location), I have commented when the problem is short out buy > bymistakely, I have not commented the Save(location). Well, Its a my foolish > fault. > > Thanks a Lot, It's worked now.Than ks a Lot. > > Bye, > Shiva > On 10/30/06, Vamsi Pottangi < vamsipottangi(a)gmail.com > wrote: > > > > Remove the first occurence of the save("location") in your script and > > try. Is there any reason why your have two occurences of save("location") ? > > > > On 10/30/06, ShivaSankar Padhy <shivapd1983(a)gmail.com> wrote: > > > > > > Yah, I forgot to Attch the file,Hear is the attachment. Hoping for a > > > quick response. > > > > > > Thanks and regards, > > > Shiva > > > > > > On 10/29/06, Vamsi Pottangi < vamsipottangi(a)gmail.com> wrote: > > > > > > > > > > > > Siva, > > > > > > > > Please send us the openser.cfg ... it is missing in your previous > > > > mail .... > > > > > > > > ~Vamsi > > > > > > > > On 10/29/06, ShivaSankar Padhy < shivapd1983(a)gmail.com> wrote: > > > > > > > > > Hi every one, > > > > > I am a new user of the openser.When I am trying to call to a > > > > > number, the invite messege is going to proxy server but from proxy, its > > > > > sending two invites simultaneously to two phones. > > > > > > > > > > I am attaching my openser.cfg for your kind information.Point to > > > > > be noted that, when the phone resisters, no value is being inserted in to > > > > > the location table of openser Database. > > > > > > > > > > Thanks in Advance, > > > > > Shiva > > > > > > > > > > _______________________________________________ > > > > > Users mailing list > > > > > Users(a)openser.org > > > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > > > > > > > > > > > > > > > > > > [View Less]

1 0

[Users] Openser ALG
by laoli 30 Oct '06

30 Oct '06

Hello, Has anyone configured Openser to as an Application Layer Gateway? Is this configuration possible? Any hints would be greatly appreciated. :-) Thanks!

3 3

[Serusers] cpled
by Doug Meredith 30 Oct '06

30 Oct '06

Hi, I am searching for cpled. I searched iptel.org, google, and the mailing list. The only thing I could find was this page http://old.iptel.org/products/cpled/. The page it links to for download doesn't seem to work anymore. Does this product still exist? Thanks for any help. Doug -- Doug Meredith 506-854-7997 ext. 801

2 1

[Users] contact header radius accounting
by Christian Schlatter 30 Oct '06

30 Oct '06

hi all, I'd like to include caller and callee SIP contact header in radius accounting messages. Our dynamic QoS provisioning framework needs this information to set up QoS per source/destination IP address pair. Caller contact header is easily done using radius_extra and setting an AVP in the INVITE handling route block. But I haven't figured out yet how I could store the callee's contact header in an AVP. The main problem I have is that I can't use any AVP related functions in … [View More]

2 1

[Users] Redirect with openSER depending on the DNS SRV response
by Oezguer Akkaya 30 Oct '06

30 Oct '06

Dear OpenSER Users or Admins, I would like to setup OpenSER as a SIP Redirect Server in my environment. The point is to redirect depending on the response from DNS or even better DNS SRV. The SIP UA will get than as quick as possible the 3xx Response and resend the request to the redirected address (IP or Domain address) SIP Device. Is there a possible way to do that via config? How many SIP Request (e.g. REGISTER, INVITE) will it be possible to redirect per second? Thanks in Advance for your responses Oetzi

2 1

[Users] Two Invites simultaneously
by ShivaSankar Padhy 30 Oct '06

30 Oct '06

Hi every one, I am a new user of the openser.When I am trying to call to a number, the invite messege is going to proxy server but from proxy, its sending two invites simultaneously to two phones. I am attaching my openser.cfg for your kind information.Point to be noted that, when the phone resisters, no value is being inserted in to the location table of openser Database. Thanks in Advance, Shiva

3 2

[Users] Why is a second RPID header created when I do two subst()?
by Alan Crosswell 30 Oct '06

30 Oct '06

Please ignore previous post which I sent before completing! I have two problems: 1. A 2nd Remote-Party-ID header gets appended when I try to do two subst() on it. 2. Xlog seems to only have a ref to the orignal unedited rpid header. I am attempting to clean up my Remote-Party-ID header so that my Polycom UAs will display the incoming number as a number rather than as a URI. This requires the domain to match that of the UA's server (e.g. siptest.columbia.edu). And, I want to rewrite … [View More]

2 1

[Users] openser.cfg
by simo med 30 Oct '06

30 Oct '06

dear sir or madame; i'm trying to start an openser, but i can't, i need to configure properly my openser.cfg! if you have an example of how can i configure my openser.cfg, to make it work, i will very thanksfull simo

2 1

[Users] How to xlog failure status in failure_route?
by Alan Crosswell 30 Oct '06

30 Oct '06

I tried xlog $rs $rr to log the reason for a failure in failure_route but it just logs NULL. What am I doing wrong? The error status is available to test with t_check_status and is logged by ACC but not xlog. # Error from media gateway failure_route[4] { xlog("L_INFO","failure_route[4] $si: method $rm r-uri <$ru>\n"); if (t_check_status("(5..)|(6..)|(403)")) { # gateway failure/refusal xlog("L_INFO","failure_route[4] gateway returned 5xx or 6xx. Status: $rs $rr\n"); … [View More]

2 1

[Users] Adapting the openserctrl file to a UNIX odbc client...
by Marvin Ottley 30 Oct '06

30 Oct '06

Hello all, I am looking for some information/pointers on how to adapt the openserctl file to use a UNIX ODBC client. Thanks, Marvin

2 1

[Users] Ring group
by Marcos Biscaysaqu 29 Oct '06

29 Oct '06

Hi there. I just getting my Opserser server up and I want to be able to create child accounts from my system to be able to ring multiple extension at the same time, is this possible ? Thanks Marcos Biscayaqu

2 1

[Users] openser and PSTN
by geovani ferronetti 29 Oct '06

29 Oct '06

Hi, I want to connect OpenSER with PSTN, as media gateway I use Cisco AS5300, vfc c542. I am interested about config part from OpenSER. Where I can get some documentation, information? Thx __________________________________________________ Do You Yahoo!? Poco spazio e tanto spam? Yahoo! Mail ti protegge dallo spam e ti da tanto spazio gratuito per i tuoi file e i messaggi http://mail.yahoo.it

2 2

[Users] yate client windows
by sinclair fils 29 Oct '06

29 Oct '06

hello, all someone can tell me how i can configurate yate client to work which openser when i make a config the account cant register please help ___________________________________________________________________________ Découvrez une nouvelle façon d'obtenir des réponses à toutes vos questions ! Profitez des connaissances, des opinions et des expériences des internautes sur Yahoo! Questions/Réponses http://fr.answers.yahoo.com

4 3

[Users] Why is a second RPID header created when I do two subst()?
by Alan Crosswell 28 Oct '06

28 Oct '06

I have two problems: 1. I am attempting to clean up my Remote-Party-ID header so that my Polycom UAs will display the incoming number as a number rather than as a URI. This requires the domain to match that of the UA's server (e.g. siptest.columbia.edu). And, I want to rewrite incoming calling numbers from my PBX to be in 5-digit format and other US 10-digit numbers into proper E.164 format since my PBX and at least one of my ITSPs doesn't provide them as E.164. So, I do a couple of subst() … [View More]

1 0

[Serusers] Accounting problems after mysql backup restore
by C. Mewes 28 Oct '06

28 Oct '06

After backup and restoring MySQL the accounting did not worked anymore. I could make calls but the amount does not show up. The problem is that o_uri has the format sip:00 + codigo de tarifa pero en la base de datos se esta guardando sin el 00 solamente sip:+codigo de tarifa ( la razon no la se, no he modificado nada de eso ). After changig in accounting $codigo2 = "sip:".$row2->codigo; if ( strncmp ( $codigo2, $row->o_uri, strlen($codigo2) ) == 0) { $tarifa=$row2->tarifa; $… [View More]

1 0

[Users] failure route doubts
by Jose Silva 27 Oct '06

27 Oct '06

Hi, I'm having some doubts on how to use/do some things with dispatcher module and failure route. I'll try to explain my problem. On my main route block I select the destination (using ds_select_dst) to where I want to send the message, and indicate that responses should pass through the failure_route, I also add a header with some things that are needed by the next proxy. On the failure route when I get 408 Time Out I mark the previous destination and select the next destination. until here … [View More]

4 7

[Serusers] some tunning to syslog messages
by Ricardo Carvalho 27 Oct '06

27 Oct '06

Hi all, Along my ser.cfg code I want to print a formatted message using LOG functions like xlog implemented with xlog module of Ser. I already can do it, but now I pretend to write in the same xlog formatted message, some formatted text that can be splitted in multiple lines. I thought that it could be done using \n sintax, like in C, but some testing made me realise that it doesn't work. I also found that this isn't a problem of xlog module of Ser, but a limitation of Syslog service … [View More]

1 1

[Serusers] Where to ask questions about CDRTool
by Lokesh Kumar 27 Oct '06

27 Oct '06

Hello List, Can anybody tell me where to ask questions about CDRTool. Thanks very much Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.14/502 - Release Date: 10/27/2006

1 0

[Serusers] New route graph utility
by Nils Ohlmeier 27 Oct '06

27 Oct '06

Hello, in case you are not following the CVS commit log messages: I just checked in a small python script which should be able to print a dependency graph in ASCII of the route blocks of a SER config file. When you try to understand more complex configuration files this tool might be helpfull. But be warned: this is still alpha code! It will not eat your config file, but the output might not meet your expectations :-) If you run it with the default configuration file the output will look … [View More]

4 6

[Users] openser domain
by simo med 27 Oct '06

27 Oct '06

dear sir or madame; i enjoy to be one of the members of your great site! i think you for all your pretty informations that you give us. If you want to send me how to create an openser domain! because when i log a user softphone to the openser, i see that it not log to the server! i think that i have problem with domain name. if i must create a domain name to make it work please send me how to do it, and if there is another way send me too, i need your help! for more information i have … [View More]

2 1

[Serusers] Jslee Arch
by Kamal.Mann＠t-systems.com 27 Oct '06

27 Oct '06

Hi All Do SER architecture based on JSlee architecture? Regards Kamal Mann

3 4

[Serusers] memory allocation error
by Lol Zimmerli 27 Oct '06

27 Oct '06

Hello, In my ser.cfg I would like to add a lot of peers in this form: ... if (uri=~"^sip:\*123[0-9]*@.*"){ xlog("L_INFO", "Forwarding *123 to 'other.sip.com': %fu (%ua - %is) -> %tu, %mf\n"); strip(4); rewritehost("other.sip.com"); } ... If I put four of them, everything is okay, ser starts and the calls are correctly redirected. But if I put 1000 of them: Oct 27 02:55:57 ser ser: ERROR:lex:addstr: memory allocation error Oct 27 02:55:57 ser last message repeated 71 times Oct … [View More]

3 3

[Users] OpenSER summit - registration will be closed Monday, Nov 30
by Daniel-Constantin Mierla 27 Oct '06

27 Oct '06

Hello everybody, the registration for OpenSER summit will be closed Monday, November 30. Afterwards, you can register via VoN, the access will be free for those having access to VoN conference or exhibition, if there is space in the summit room. There are over 80 registered people, about 20 special guests, any many other executives present at VoN were invited to visit us. The agenda is now in final shape (http://www.openser.org/index.php?option=com_content&task=view&id=60&… [View More]

1 1

[Users] Problems with Mediaproxy. Bad Port.
by Ricardo Martinez 27 Oct '06

27 Oct '06

Hello list. I'm having a problem with my mediaproxy box. I have a NAT'd client (a softphone) calling to a SIP-H323 conversor through my OpenSER-Mediaproxy box and then to a Cisco Call Manager (it's a very weird configuration). Anyway, the problem presents when the 200 - OK message arrives from my conversor, the SDP part indicate the media address (10.10.148.230) and port (16952) where the remote endpoint will listen RTP, despite of this the mediaproxy open a session for the port 17360. The … [View More]escenario is something like this. 100.100.148.248 ------- 100.100.148.246 ----- 100.100.154.36 ------ 100.100.154.119 ---- 100.100.148.230 Softphone OpenSer/Mediaproxy Proxy Sip siph323 conversor Call Manager.H323 Please check the log from the mediaproxy and the execution of the "sessions.py" utility, plus the OK message. U 100.100.154.36:5060 -> 100.100.148.246:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.148.246;branch=z9hG4bK67ea.aa12dc97.0. Via: SIP/2.0/UDP 192.168.1.101;received=100.100.148.248;rport=46096;branch=z9hG4bKc0a801650000001d4533a0ee00007c2900000011. Record-Route: <sip:100.100.154.36;ftag=2214787970;lr=on>. Record-Route: <sip:7072408196@100.100.148.246:5060;nat=yes;ftag=2214787970;lr=on>. From: "unknown"<sip:5502408196@sip.desa.mydomain.net>;tag=2214787970. To: <sip:7072408196@sip.desa.mydomain.net>;tag=2260729709. Call-ID: A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101. Cseq: 2 INVITE. Date: Mon, 16 Oct 2006 15:10:37 GMT. Server: siph323. Content-Type: application/sdp. Content-Length: 162. Contact: sip:7072408196@100.100.154.119. . v=0. o=100.100.154.119 1 505533324 IN IP4 100.100.154.119. s=SIP Library call. c=IN IP4 100.100.148.230. t=3370000237 0. m=audio 16952 RTP/AVP 0. a=rtpmap:0 PCMU/8000. Oct 16 12:12:00 sip mediaproxy[18289]: lookup A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101 100.100.148.230:16952:audio 100.100.154.36 sip.desa.mydomain.net local sip.desa.mydomain.net unknown siph323 info=from:5502408196@sip.desa.mydomain.net,to:7072408196@sip.desa.mydomain.net,fromtag:2214787970,totag:2260729709 Oct 16 12:12:00 sip mediaproxy[18289]: execution time: 0.48 ms Oct 16 12:12:00 sip mediaproxy[18289]: session A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101: caller signed in from 100.100.148.248:49154 (RTP) (will return to 100.100.148.248:49154) Oct 16 12:12:00 sip mediaproxy[18289]: session A2304B7B-AA05-42C5-A467-C16B1BD9102A(a)192.168.1.101: called signed in from 100.100.148.230:17360 (RTP) (will return to 100.100.148.230:17360) [root@sip ]# /usr/src/mediaproxy-1.7.2/mediaproxy/sessions.py Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------------------------------------------------------ 100.100.148.248:49154 - 100.100.148.246:60022 - 100.100.148.230:17360 active 0'39" G711u Audio 336.91k/669.53k/332.62k Total traffic: 79.69kbps/78.12kbps/157.81kbps (in1/in2/out) Session count: 1 Proxy version: 1.7.2 Doing more debugs i discover the RTP traffic coming from the 100.100.148.230 it has source port 17360, but the listening port is not the same (i guess is the one indicated in the SDP). Why mediaproxy assumes the 17360 port is for listening too and ommit the SDP 16952 port ? Is this maybe an asymmetric regarding the RTP media?. If so, i added this line into the rtp-asymmetric-clients file: # # Lines starting with a '#' character or empty lines are ignored. # Put a User-Agent name or regular expression per line. # Check is case insensitive. # # This file should only list SIP clients that are asymmetric regarding # the RTP media streams. For clients that are asymmetric regarding the SIP # signaling, use the equivalent SIP version of this file. # Clients that are asymmetric for both SIP signaling as well as RTP media # streams should go in both files. # siph323 But not solve the problem. Can someone help me here? Thanks in advance Regards Ricardo Martinez.- RedVoiss. [View Less]

2 1

[Users] Q:ABOUT:400 Bad Request RESPONESE
by Alejandro Sanchez 27 Oct '06

27 Oct '06

Hi evreybody My sip server sends this message to the Phone. INVITE sip:192.168.1.10:7191 SIP/2.0 Record-Route: <sip:192.168.1.20;lr=on;ftag=1604fbda> Via: SIP/2.0/UDP 192.168.1.20;branch=z9hG4bK2f9.039444a7.0 Via: SIP/2.0/UDP 192.168.1.9:7191;branch=z9hG4bK71Fso4gTlg Max-Forwards: 69 Content-Type: application/sdp To: <sip:2109075@192.168.1.20> From: "4422109087" <sip:4422109087@192.168.1.20>;tag=1604fbda Call-ID: 5f1e5301-40b07670(a)192.168.1.9 CSeq: 1596803196 INVITE Allow: … [View More]

2 1

[Serusers] variables in log messages
by Mark Price 26 Oct '06

26 Oct '06

Is there a method to print variables into log messages? For example, something like this: instead of saying: if(method=="SUBSCRIBE") { log(1,"SUBSCRIBE\n"); } do this log(1,method) which obviously doesn't work. It would be very helpful to be able to print out things like headers. Perhaps the select or more likely AVP functions will work in 0.10? Is there anything available now? Mark

2 1

[Users] uri or to_uri
by Gregoire 26 Oct '06

26 Oct '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi the list! I am just wondering the difference between the variables "uri" and "to_uri". I saw that "to_uri" refers to the value of the field "To", but I can not see at which field refers "uri". Does someone know something about that? Thanks Greg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFQHp6I8gmGeMTr0sRAgbuAKCmISnqxdAvCQeWTsJjxtNoqoCY8gCeJA8c kW4+/… [View More]

2 1

[Users] BAD REQUEST
by Jose Gil Navarrete 26 Oct '06

26 Oct '06

Hi. My problem is the next. When my sip server send the message INVITE to the phone, the phone always respond with 400 Bad Request, I think is something wrong in the message, in the UDP message the Checksum always is corrupted by the sip server, is it the problem? or is something else in the sip message? How I can correct this? this is the trace: User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 7191 (7191) Source port: 5060 (5060) Destination port: 7191 (7191) Length: 1057 Checksum: … [View More]

2 1

[Users] Mysql table version problem in cvs version
by Pletli Antal 26 Oct '06

26 Oct '06

Hi, I have a problem with openser cvs version. Before the installing the cvs version i dropped the previous openser database. I have installed the new version, and after make dbinstall the "version" table contains the following record for version silo: | silo | 5 | but the debug log shows this problem: 0(0) table_version(): Invalid type (2) or nul (0) version columns for silo 0(0) MSILO:mod_init: Wrong version v-1 for table <silo>, need v5 0(0) init_mod()… [View More]

2 1

[Users] pbxnsip and openser
by Nick De Cristofaro 26 Oct '06

26 Oct '06

Anyone here using pbxnsip as their BB user agent with OpenSER. I'm wondering what steps would be required to make it work. Is it a complicated process? Thanks Nick

2 1

[Users] Replication ACK problem
by Dioris Moreno 26 Oct '06

26 Oct '06

Hi everybody I have a working installation of OpenSER (Proxy A). In this installation, the ACK messages are delivery without any problem through loose_route() and t_relay(). Then, I installed a second OpenSER proxy (Proxy B) and configure both servers to replicate REGISTER messages with t_replicate(), and it works fine. I made a lot of captures to check the replication process was working fine and also to check the NAT pings; each proxy takes care of the NATed clients it receives sending … [View More]

2 1

[Users] Conference is Not Working.... with OpenSER And Asterisk
by sunkara 26 Oct '06

26 Oct '06

Hello Users, Good Morning, I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql modules. And I'm not Using the Zapptel Cards. 9001 ----------> dial 19001(conference Users)-------openSER ---------> Asterisk ------------------------------------------------------------------------ *In Extension.conf * [from-sip] exten => 19001,1,Playback(conf-hasentered) ;exten => 19001,2,Answer exten => 19001,2,Wait(2) exten => 19001,3,CBMysql() exten => … [View More]19002,1,Playback(conf-hasentered) ;exten => 19002,,Answer exten => 19002,2,Wait(2) exten => 19002,3,MeetMe(1234|pm) exten => 19003,1,Playback(conf-hasentered) ;exten => 19003,2,Answer exten => 19003,2,Wait(2) exten => 19003,3,MeetMe(1234|pm) ------------------------------------------------------------------------ * In cbmysql.conf* [global] hostname=localhost dbname=conference password= user=root port=3306 sock=/var/lib/mysql/mysql.sock DBOpts=yes ;OptsAdm=asdp ;OptsUsr=sdp ConfApp=MeetMe ConfAppCount=MeetMeCount ; Choose one of the following to modify early join behaviour earlyalert=300 ; Tell the participant if they are too early (seconds) ;fuzzystart= ; Allow participants to join early (seconds) ------------------------------------------------------------------------ * In sip.conf *[19001] type=friend username=9001 secret=august context=from-sip host=192.168.2.75 fromdomain=192.168.2.76 realm=192.168.2.75 ;mailbox=9001@from-sip insecure=very callerid="Ravi" <9001009> disallow=all allow=ulaw allow=gsm nat=yes ------------------------------------------------------------------------ 9001,9002,and 9003 is register from openSER, When They dial 19001 to enter the conference. Following are Showing the Errors and Warning .... *Executing Playback("SIP/9001-08f8d7e0", "conf-hasentered") in new stack -- Playing 'conf-hasentered' (language 'en') -- Executing Wait("SIP/9001-08f8d7e0", "2") in new stack -- Executing CBMySQL("SIP/9001-08f8d7e0", "") in new stack -- Playing 'conf-getconfno' (language 'en') Oct 25 18:15:47 NOTICE[12281]: app_cbmysql.c:373 cb_exec: getConf: 1 -- Playing 'agent-pass' (language 'en') Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags: Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags: Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:146 passQuery: CBMySQL: Invalid room or pass Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY: -- Playing 'auth-incorrect' (language 'en') Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags: Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags: Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:146 passQuery: CBMySQL: Invalid room or pass Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY: -- Playing 'auth-incorrect' (language 'en') Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags: Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags: Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY: Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:376 cb_exec: getPass: 1 == Parsing '/etc/asterisk/meetme.conf': Found Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:97 checkMax: Currentusers: 0 Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:390 cb_exec: checkMax: 1 Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:276 enterConf: Roomtype: 1234|| == Parsing '/etc/asterisk/meetme.conf': Found Oct 25 18:16:13 WARNING[12281]: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Oct 25 18:16:13 ERROR[12281]: chan_zap.c:7396 chandup: Unable to dup channel: No such file or directory Oct 25 18:16:13 WARNING[12281]: app_meetme.c:460 build_conf: Unable to open pseudo channel - trying device Oct 25 18:16:13 WARNING[12281]: app_meetme.c:463 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') Oct 25 18:16:17 NOTICE[12281]: app_cbmysql.c:393 cb_exec: enterConf: -1 == Spawn extension (from-sip, 19001, 3) exited non-zero on 'SIP/9001-08f8d7e0' * Please Help me..... -- Thanks & Regards, *Ravi Prakash Sunkara* *M*:+91 9985077535* O*:+91 40 23114549* F*:+91 40 40208727 /*ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com */ [View Less]

2 1

[Users] Changing call ID?
by Olle E Johansson 26 Oct '06

26 Oct '06

I got a bug report in our bug report tracker for Asterisk that claims that SER changes the call ID by adding a :port after the IP address. From reading the RFC, I don't think it's a valid change. For me, it's a new call ID. Questions: - Is changing a Call ID permitted within a dialog? - Does SER/OpenSER actually do this? http://bugs.digium.com/view.php?id=8220 See you at the Summit in Berlin! regards, /Olle

2 1

[Users] Cvs snapshot Invalid table version
by Pletli Antal 26 Oct '06

26 Oct '06

Hi, How can I fix the "version" table to work with openser cvs snapshot? I have problem with grp,silo and location tables. I have installed the DB from the cvs too. +-----------------+---------------+ | table_name | table_version | +-----------------+---------------+ | subscriber | 5 | | missed_calls | 3 | | location | 1003 | | aliases | 1003 | | grp | 2 | | re_grp | 1 | | acc … [View More]

2 1

[Users] Registration of Polycom SoundPointIP phone with OpenSER
by jeevan ravula 26 Oct '06

26 Oct '06

Hi all, I am using Polycom SoundPointIP phone as User Agent.I want to register Polycom phone with OpenSER(with TLS support) server.Can anybody help me out in this regard? I have generated my rootCA and given to polycom phone.The polycom phone does not accept certificate from openser server side.It shows bad certificate. anybody who has used polycom phone earlier can help me out in this matter.Ishall be greatful to them Regards, Jeevan.

2 6

Re: [Serusers] Re: How to change Caller ID in FROM field ?
by lawrence k.y. lin 26 Oct '06

26 Oct '06

Hi there, This is an update for the Caller ID with Telco Carrier. The reason that Caller ID did not work on Verizon's mobile phone is because I only change the Display Name in the From Field. After I changed the From-URI (the field after the Displan Name which is <sip: XXXXXXXXXX@ip>), then Caller ID is displayed on Verizons mobile phone without any problem. Thanks for all that replied to this question. Best Regards, Larry >lawrence k.y. lin wrote: >Steve, … [View More]> > Thanks for your email and your solutions. > > I tried the first method as follws: > >replace ("^From:(.*)<", "From: \"XXXXXXXXXX\" <") > > before I sent it to another SIP device and PSTN gateway. I did see the >new Caller ID XXXXXXXXXX in another SIP device but the Caller ID to Verison >Mobile phone showed "Unavailable No". I will check with Telco carrier to >see if we did send the Caller ID to them or not. > Also I will try your second method and update to the list later. > > Thanks again. > >Larry > >>From: Steve Blair <blairs(a)isc.upenn.edu> >>To: "lawrence k.y. lin" <lawlin888(a)hotmail.com> >>CC: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Re: How to change Caller ID in FROM field ? >>Date: Tue, 24 Oct 2006 18:57:42 -0400 >> >> >>You can do this several ways. I use to just completely replace the From >>header but this screwed up call accounting (for callers with caller ID >>blocking enabled) which we do in our Cisco gateways. Now I append an >>Remote-Party-ID header leaving the Fromheader intact for accounting >>purposes. Try one of the following: >> >>1) replace("^From:(.*)>" , "From: \"Anonymous\" >><sip:anonymous@invalid.nowhere>"); >> >> Where the second string contains whatever you want the caller ID info >>to be. This example is for caller ID blocking. >> >>2) >> >>if (append_rpid_hf("Anonymous >><sip:","@upenn.edu>;party=calling;id-type=subscriber;screen=yes")) { >>} else { >> log(1, "RPID not found\n"); >>}; >> >>Of course this requires other configuration items. You may also want to >>add the following: >> >>modparam("auth", "rpid_prefix", "<sip:") >>modparam("auth", "rpid_suffix", >>"@upenn.edu;user=phone>;party=calling;screen=no;id-type=subscriber;privacy=off") >> >># >> >>-Steve >> >>lawrence k.y. lin wrote: >> >>>Hi all, >>> >>> Appreciate if someone can give me a suggestion or even an answer. >>> Since the call from SIP device to PSTN phone did not have Caller ID >>>and our Telco carriers request for the Caller ID, we have to set a number >>>before the calls are routed to PSTN gateway. >>> I did it in Asterisk before with the function calls SetCallerID(CLID) >>>or Set(CALLERID(number)=CLID) (depends on the Asterisk version) and these >>>function calls worked. But because of the scalabliity issue, I changed to >>>SER and would like to do the similar thing as in Asterisk. >>> I read the SER document and awared SIP_HF_FROM has the data for the >>>entired FROM field. Is there any other variables for Caller ID only ? >>> I searched the mailing list for SER and did not find any related >>>topic (only found one with title: caller-id with raius using sip-rpid, >>>but it is irrelevant). >>> >>> Thanks in advance. >>> >>>Larry _________________________________________________________________ Try the next generation of search with Windows Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&source=… [View Less]

1 0

[Users] Re: Conference is Not Working.... with OpenSER And, Asterisk
by Paul Smith 25 Oct '06

25 Oct '06

This is an Asterisk config problem rather than an OpenSer one. The conference applications in Asterisk need a timing source. If you install a Digium or Sangoma card then Asterisk will use it as a timing source. As you are not using such a card you need to compile and modprobe a module called ztdummy to create a pseudo timing device.

1 0

[Serusers] How do I allow gateway on ser to not ask for any authentication
by Lokesh Kumar 25 Oct '06

25 Oct '06

Hello , How do I configure my ser and radius so that it won’t ask for credentials from PSTN gateway, Please help Thanks very much Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 - Release Date: 10/24/2006

2 1

[Serusers] Re: Asterisk Manager
by Maps 25 Oct '06

25 Oct '06

Dear Friends and Supporters! I try to write a php application to monitor the asterisk, but when I open the .php to access to asterisk to retrieve the information about the queues status, sip show peers, realtime mysql status etc... However, It just return to me "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" Asterisk is current running with the a file in /var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the owner of the folder /var/run, and … [View More]

2 1

[Serusers] MySQL has gone away
by Mark Price 25 Oct '06

25 Oct '06

Hi, I'm getting the MySQL has gone away error. I have set the following in ser.cfg: modparam("mysql", "auto_reconnect", 1) modparam("mysql", "ping_interval", 60) Restart fixes this, but that is of course not sufficient. I suppose I can detect every time that error happens with swatch or some such thing and restart the service. This seems to be a common error. Is there a common solution? Mark

1 0

[Serusers] MySQL confusion
by Chris Parkinson 25 Oct '06

25 Oct '06

Hello, I'm a complete newbie to SER and I'm having a problem following the MySQL section of the "getting started" guide. Whenever I try to use serctl to add a user I get access denied; LabServ1:/home/chris# export SIP_DOMAIN=mylab.com LabServ1:/home/chris# serctl add test1 test1pass test1(a)mylab.com MySql password: ERROR 1045: Access denied for user: 'serro@localhost' (Using password: YES) error: SER/FIFO not accessible: 2 LabServ1:/home/chris# I think this may be … [View More]

3 5

[Users] X-lite is not Working, But SipEyeMedia Softphones is Working Fine.
by sunkara 25 Oct '06

25 Oct '06

Hello Users, I don't Know What Happening in this Issues... Help me in this . When Testing the X-lite Softphones, is not working. When I use the Sip EyeMedia Softphone it Working Fine. Is Problem in X-lite Configuration or Router(Firewall Gateway Keepers ) is blocking it... Help me . -- Thanks & Regards, *Ravi Prakash Sunkara* *M*:+91 9985077535* O*:+91 40 23114549* F*:+91 40 40208727 /*ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com */

2 1

[Serusers] Using different domains
by Nick Hoffman 25 Oct '06

25 Oct '06

Hi guys. I just noticed that when user agents register, the domain in their SIP URI does not have to equal their domain in the subscribers database table. For instance, I have 2 domains in the domain table: sip.altcall.com sip.voxpak.com The username "foobar" in my subscribers table has the domain "sip.altcall.com", but can register and make calls as foobar(a)sip.voxpak.com . Why is this? Cheers, -- Nick E: nick.hoffman(a)voxpak.com P: +61 7 5591 3588 F: +61 7 5591 6588 If you … [View More]

1 0

[Serusers] Re: SER "presence snapshots": "id" value in <tuple>
by Vaclav Kubart 25 Oct '06

25 Oct '06

Hi, SER always generates new tuple IDs for newly published tuples. This is because it can collect information from more sources - more clients can send PUBLISH with the same tuple IDs, but according to RFC 3903 tuples can rewrite each other only if they were published with the same entity tag (SIP-ETag header field). If you don't want to use tuples created from registration status you can set PA module parameter "use_callbacks" to 0. Or do you need for clients which published something not to … [View More]

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users