sr-users July 2013

sr-users@lists.kamailio.org

92 participants
149 discussions

Problem with forward on busy
by LAA 23 Jul '13

23 Jul '13

Hi all, I'm running Kamailio 3.0.0, with SEMS integration as Media Server for Voice mail. I'm trying to get a configuration to forward calls on busy to voice mail. I have followed without success some examples. I'm using revert_uri(), rewritehostport() and append_branch(), within failure_route. It seems to be modifying R-URI properly, and generating the new branch, but Kamailio is sending the new invite packet to the IP address of the original destination UAC, and not to the IP address of the voicemail, that was indicated in the R-URI. Here you can see the packet flow: |Time | 192.168.3.20 | 192.168.0.167 | | | | 192.168.0.197 | |5,069 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | |5,071 | 407 Proxy Authentication Required | |SIP Status | |(5060) <------------------ (5060) | | |5,074 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |5,076 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | |5,084 | 100 trying -- your call is important to us | |SIP Status | |(5060) <------------------ (5060) | | |5,085 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |5,088 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |5,088 | | 486 Busy Here |SIP Status | | |(5060) <------------------ (5060) | |5,091 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |5,101 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |5,102 | | 404 Not Found |SIP Status | | |(5060) <------------------ (5060) | |5,102 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |5,103 | 404 Not Found | |SIP Status | |(5060) <------------------ (5060) | | |5,106 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | And the RAW capture of the INVITE message in timestamp 5,101. No. Time Source Destination Protocol Info 1235 5.100698 192.168.0.197 192.168.0.167 SIP/SDP Request: INVITE sip:voicemail4440@192.168.0.197:5080, with session description Frame 1235 (1151 bytes on wire, 1151 bytes captured) Ethernet II, Src: CadmusCo_96:31:84 (08:00:27:96:31:84), Dst: Micro-St_6d:77:54 (00:21:85:6d:77:54) Internet Protocol, Src: 192.168.0.197 (192.168.0.197), Dst: 192.168.0.167 (192.168.0.167) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:voicemail4440@192.168.0.197:5080 SIP/2.0 Method: INVITE Request-URI: sip:voicemail4440@192.168.0.197:5080 [Resent Packet: True] [Suspected resend of frame: 1233] Message Header Record-Route: <sip:192.168.0.197;lr=on;nat= yes> Via: SIP/2.0/UDP 192.168.0.197;branch=z9hG4bKafce.403718a6.1 Via: SIP/2.0/UDP 192.168.57.20;received=192.168.3.20;rport=5060;branch=z9hG4bK0a00030f0000003151ed60b85ec2c3de000000c8 Content-Length: 386 Contact: <sip:4095@192.168.3.20:5060> Call-ID: 8EAF9EC2-1DD2-11B2-B110-C84E476664B0(a)10.0.3.15 Content-Type: application/sdp CSeq: 2 INVITE From: "4095"<sip:4095@192.168.0.197>;tag=121754238352072516 Max-Forwards: 69 To: <sip:4440@192.168.0.197> User-Agent: SJphone/1.60.299a/L (SJ Labs) P-App-Name: voicemail P-App-Param: mod=box;usr= voicemail4440;dom=sipproxy.a.com ;uid=voicemail4440;did=sipproxy.a.com; Message Body Here you can see the failure_route in my kamailio.cfg file: # Sample failure route failure_route[FAIL_ONE] { #ifdef WITH_NAT if (is_method("INVITE") && (isbflagset("6") || isflagset(5))) { unforce_rtp_proxy(); } #endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]") ) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination if (t_check_status("486|408")) { revert_uri(); prefix("voicemail"); remove_hf("P-App-Name"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com ;uid=$rU;did=sipproxy.a.com;\r\n"); $ru = "sip:" + $rU + "@" + "192.168.0.197:5080"; #rewritehostport("192.168.0.197:5080"); #append_branch("sip:4888@192.168.0.102"); append_branch(); # do not set the missed call flag again t_relay(); } } Has anybody experienced this problem? Any help would be wellcome Best Regards LAA

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Kamailio 4.0.2 as a new Port for FreeBSD
by Oliver Mahmoudi 23 Jul '13

23 Jul '13

Hello, I am currently working on porting Kamailio 4.0.2 to the FreeBSD ports collection: /usr/ports, and I just have a few general questions regarding compilation with regard to the extra modules. There are 172 modules that come with Kamailio, and most, however not all of them are compiled by default: # gmake all The base server and the base modules compile fine on my 9.1 release. To really compile all of the 172 modules, Kamailio_4_0_2/INSTALL says that the following external libraries (LIB_DEPENDS I assume) are needed: /usr/ports/databases/mysql-client-* /usr/ports/databases/postgresql-libpqxx /usr/ports/textproc/expat2 /usr/ports/textproc/libxml2 /usr/ports/net/radiusclient This is put down as information for FreeBSD 5.4. Does that still hold current for 9.1? Is there an exhaustive list of all the modules that the all target does/does not compile and - apart from mysql, postgresql - which of the extra modules require which libraries? I would set compile time options accordingly. This would greatly facilitate my work! Oliver Mahmoudi

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Kamailio crashing when accessing db tables
by Fabian Borot 23 Jul '13

23 Jul '13

Hi, I installed the latest version 4.0 on CentOS release 5.9 , and when running any fifo command that accesses a module's table (lcr, address etc.) or when a call comes in an there is a need to access a table the app crashes. Example: Jul 23 09:07:55 proxy01 kernel: kamailio[16608]: segfault at 0000000000000000 rip 00002b3bcd39a36c rsp 00007fff28ef40d0 error 4 Jul 23 09:07:55 proxy01 ./kamailio[16565]: ALERT: <core> [main.c:788]: handle_sigs(): child process 16608 exited by a signal 11 Jul 23 09:07:55 proxy01 ./kamailio[16639]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 41 Jul 23 09:07:55 proxy01 ./kamailio[16565]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated has any one seen this? Is there something new on this version that I am missing? thank you a lot fabian

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get CDR fields in DB
by Jayesh Nambiar 23 Jul '13

23 Jul '13

Hello All, Is there no straight forward way of getting the ACC module CDR values into the DB. I mean logging the start_time, end_time and duration in DB directly using some pseudo variables from the config script !! I think, if a dialog is maintained, all the values related to that dialog is anyways available and a manual insert on dlg_end event can be called. But I am only missing the milli-second precision that the CDR module provides !! Thanks for any pointers. --- Jayesh

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Re: [SR-Users] trouble counting dialogs
by Gertjan Wolzak 23 Jul '13

23 Jul '13

Well, can't stand needing to ask for help... So, I keep trying . found it, well it works now. This is the concurrent route that works: route[CONCURRENT] { xlog("SCRIPT: Conccurrent call check"); if(!get_profile_size("caller","$fu","$avp(nrcalls)")) { sl_send_reply("403", "Call not matching profile"); exit; } xlog("SCRIPT: caller value for $fu is $avp(nrcalls)"); if($avp(nrcalls)>= 1) { sl_send_reply("403", "Active calls limit exceeded"); exit; } dlg_manage(); if(!set_dlg_profile("caller","$fu")) { sl_send_reply("500", "No new channels in this profile"); exit; } xlog("SCRIPT: caller value for $fu is now $avp(nrcalls)"); } Maybe someone else can be helped by it. Thanks. From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Gertjan Wolzak Sent: dinsdag 23 juli 2013 7:58 To: sr-users(a)lists.sip-router.org Subject: [SR-Users] trouble counting dialogs Hello, I am trying to limit the number of concurrent calls to 1 per user.. But I can't get the dialogs to be counted. Can anybody tell me what I am doing wrong..? Kamailio version 3.2.3 # ------ dialog params ------- modparam("dialog", "enable_stats", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "dlg_flag", 24) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_url", "mysql://xxxxx:xxxxxx@127.0.0.1/dbname") modparam("dialog", "db_mode", 1) request_route { . # Check if caller is in call if(is_method("INVITE") && !has_totag()) { route(CONCURRENT); } . } route[CONCURRENT] { xlog("SCRIPT: Conccurrent call check"); get_profile_size("caller", "$fu", "$var(SIZE)"); xlog("SCRIPT: profile size is $var(SIZE)"); if( $var(SIZE) >= 1 ) { sl_send_reply("503", "Simultaneous calls limit reached"); exit; } set_dlg_profile("caller","$fu"); get_profile_size("caller", "$fu", "$var(SIZE)"); xlog("SCRIPT: profile size is now $var(SIZE)"); xlog("SCRIPT: No concurrent call"); } I feel I should do a dlg_setflag(), but I am not sure and even if I have to, not sure where to place it. Hope someone can point me in the right direction. Thanks. Gertjan Wolzak

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trouble counting dialogs
by Gertjan Wolzak 23 Jul '13

23 Jul '13

Hello, I am trying to limit the number of concurrent calls to 1 per user.. But I can't get the dialogs to be counted. Can anybody tell me what I am doing wrong..? Kamailio version 3.2.3 # ------ dialog params ------- modparam("dialog", "enable_stats", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "dlg_flag", 24) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_url", "mysql://xxxxx:xxxxxx@127.0.0.1/dbname") modparam("dialog", "db_mode", 1) request_route { . # Check if caller is in call if(is_method("INVITE") && !has_totag()) { route(CONCURRENT); } . } route[CONCURRENT] { xlog("SCRIPT: Conccurrent call check"); get_profile_size("caller", "$fu", "$var(SIZE)"); xlog("SCRIPT: profile size is $var(SIZE)"); if( $var(SIZE) >= 1 ) { sl_send_reply("503", "Simultaneous calls limit reached"); exit; } set_dlg_profile("caller","$fu"); get_profile_size("caller", "$fu", "$var(SIZE)"); xlog("SCRIPT: profile size is now $var(SIZE)"); xlog("SCRIPT: No concurrent call"); } I feel I should do a dlg_setflag(), but I am not sure and even if I have to, not sure where to place it. Hope someone can point me in the right direction. Thanks. Gertjan Wolzak

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Trying to send kamailio to a uri derived from a database query
by Will Ferrer 23 Jul '13

23 Jul '13

Hi I am trying to make a kamailio script which gets a uri from the database and then directs the call to that uri. Setting the uri with a string (not derived from the database) works just fine: append_branch(); seturi("sip:user@domain"); But I can not seem to pass anything to seturi other than a string. Here are some examples: $var(branch_uri) = $dbr(ra=>[0,0]); append_branch(); seturi($var(branch_uri)); # kamailio won't even start when I try to pass the uri from a variable $var(branch_uri) = $dbr(ra=>[0,0]); append_branch(); seturi("$var(branch_uri)"); # treats the "$var(branch_uri)" as a literal string -- does not evaluate to the the value of the variable $var(branch_uri) = $dbr(ra=>[0,0]); append_branch(); seturi((str)$var(branch_uri)); # kamailio won't start with this syntax inplace The errors that kamailio gives on start up when I have a variable in passed into seturi are as follows 0(16145) : <core> [cfg.y:3567]: yyerror_at(): parse error in config file /usr/local/kamailio-4.0/etc/kamailio/kamailio.cfg, line 647, column 12-27: syntax error 0(16145) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file /usr/local/kamailio-4.0/etc/kamailio/kamailio.cfg, line 647, column 28: bad argument, string expected 0(16145) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <sl_reply_error> in module sl [/usr/local/kamailio-4.0/lib64/kamailio/modules/sl.so] 0(16145) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <sl_reply_error> in module sl [/usr/local/kamailio-4.0/lib64/kamailio/modules/sl.so] 0(16145) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <sl_reply_error> in module sl [/usr/local/kamailio-4.0/lib64/kamailio/modules/sl.so] ERROR: bad config file (2 errors) I also tried to accomplish this append_branch($var(branch_uri), "0.0"); t_load_contacts(); t_next_contacts(); # No errors but didn't work I found a way around the problem by making a perl script to allow me set the uri: Perl script: sub rewrite_uri { my $m = shift; my $uri = shift; $m->rewrite_ruri($uri); return -1; } Then I call it like this: $var(branch_uri) = $dbr(ra=>[0,0]); append_branch(); perl_exec("rewrite_uri", $var(branch_uri)); # this works So I have a viable work around but it seems like there is probably something simple I am missing which should let me accomplish this with out having to do a call into perl. Any advice any one can give me would be greatly appreciated. Best regards Will Ferrer

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Re: [SR-Users] Replication problems Registration Using t_replicate ().
by Raphael Borges 22 Jul '13

22 Jul '13

Thanks Daniel, your help was very useful. With this I created a header by append_hf () before calling the function t_replicate and standby server retrieved the value of this header and assigns this value to the parameter avp Example $avp(s:rcv) = $hdr(IP-source); Best regards Raphael Hello, > > the received is taken from an avp specified by parameter received_avp of > registrar module: > > http://kamailio.org/docs/modules/devel/modules/registrar.html#idp4433296 > > You can carry the original details via a header on the replicated REGISTER > (using append_hf()), then take the header on the second server and store > its value in the avp. > > Cheers, > Daniel > > > On 7/18/13 9:05 PM, Raphael Borges wrote: > > *Hello,* > > > I have two servers in kamailio duality, I am using the function > t_replicate () for replication log as below: > Server 1 active > t_replicate ("xxx.xxx.xxx.2", "5060"); > Server 2 reserves > t_replicate ("xxx.xxx.xxx.1", "5060"); > > But when the record is replicated to the standby server the parameter > received is being filled with the active server IP, where the IP would > Correro UAC, registry problems generating the backup server. > > Example > Server 1 active xxx.xxx.xxx.1 > > $ Kamctl ul show > Domain :: location table = 512 records = 1 = 1 max_slot > AOR :: 4301 > Contact :: sip: 4301(a)yyy.yyy.yyy.100; transport = udp Q = > Expires :: 4566 > Callid :: ccd0431c-6a4c-1231-0a94-7356c4f466e6 > CSeq :: 46740150 > User-agent :: stepo_LITE > Received :: sip: yyy.yyy.yyy.100: 5060 > State :: CS_SYNC > Flags :: 0 > CFLAG :: 0 > Socket :: udp: xxx.xxx.xxx.1: 5060 > Methods :: 8159 > > Server 2 Reserve xxx.xxx.xxx.2 > > $ Kamctl ul show > Domain :: location table = 512 records = 1 = 1 max_slot > AOR :: 4301 > Contact :: sip: 4301(a)yyy.yyy.yyy.100; transport = udp Q = > Expires :: 6665 > Callid :: ec7cfd42-6a7c-1231-0f94-7356c4f466e6 > CSeq :: 46750472 > User-agent :: stepo_LITE > Received :: sip: xxx.xxx.xxx.1: 5060 > State :: CS_SYNC > Flags :: 0 > CFLAG :: 0 > Socket :: udp: xxx.xxx.xxx.2: 5060 > Methods :: 8159 > > > > Tkanks > > Raphael > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > >

1 0

floating point operations in kamailio
by Jayesh Nambiar 22 Jul '13

22 Jul '13

Is there a plan to support floating point arithmetic operations in kamailio in the near future. I am looking at something similar to mathops module in opensips. Thanks, --- Jayesh

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New developer: Lucian Balaceanu
by Henning Westerholt 22 Jul '13

22 Jul '13

Hello, I am pleased to announce a new developer for the Kamailio project: Lucian Balaceanu. Lucian works in our 1&1 office in Bucharest and is for some years with us in the company. He worked on several internal projects with our Kamailio stack and will contribute some fixes to the sipcapture and carrierroute module. He will also probably commit some other smaller fixes from time to time. Welcome Lucian and all the best, Henning

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sr-users July 2013