- sr-users - kamailio.org

RPCFIFOPATH / DEFINE_FIFO_NAME settings problem
by Ginhoux, Patrick 06 Apr '17

06 Apr '17

In my 'kamctlrc' file : ## path to FIFO file for engine RPCFIFO RPCFIFOPATH="/var/run/kamailio/kamailio_rpc_fifo" #RPCFIFOPATH="/tmp/kamailio_rpc_fifo" In my 'kamailio.cfg' : !!ifndef DEFINE_FIFO_NAME !!define DEFINE_FIFO_NAME "/var/run/kamailio/kamailio_rpc_fifo" !!endif modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 2) modparam("jsonrpcs", "fifo_name", DEFINE_FIFO_NAME) modparam("jsonrpcs", "fifo_mode", 0755) modparam("jsonrpcs", "fifo_group", "kamailio") modparam("jsonrpcs", "fifo_user", "kamailio") kamailio doesn't start. It reports 'Permission denied' : Mar 24 17:31:21 localhost /usr/sbin/kamailio[1138]: ERROR: jsonrpcs [jsonrpcs_fifo.c:144]: jsonrpc_init_fifo_server(): Can't create FIFO: Permission denied (mode=493) Mar 24 17:31:21 localhost /usr/sbin/kamailio[1138]: CRITICAL: jsonrpcs [jsonrpcs_fifo.c:489]: jsonrpc_fifo_process(): failed to init jsonrpc fifo server Mar 24 17:31:21 localhost /usr/sbin/kamailio[1120]: ALERT: <core> [main.c:741]: handle_sigs(): child process 1138 exited normally, status=255 Mar 24 17:31:21 localhost /usr/sbin/kamailio[1130]: DEBUG: <core> [core/sr_module.c:920]: init_mod_child(): rank 4: tm Mar 24 17:31:21 localhost /usr/sbin/kamailio[1137]: DEBUG: <core> [core/sr_module.c:920]: init_mod_child(): rank -1: tm Mar 24 17:31:21 localhost /usr/sbin/kamailio[1127]: DEBUG: htable [htable.c:226]: child_init(): rank is (1) Mar 24 17:31:21 localhost /usr/sbin/kamailio[1120]: INFO: <core> [main.c:759]: handle_sigs(): terminating due to SIGCHLD Mar 24 17:31:21 localhost /usr/sbin/kamailio[1139]: DEBUG: <core> [core/sr_module.c:920]: init_mod_child(): rank -2: kex Mar 24 17:31:21 localhost /usr/sbin/kamailio[1130]: DEBUG: tm [callid.c:137]: child_init_callid(): callid: '15b1f0d63a718465-1130(a)129.227.83.108' Mar 24 17:31:21 localhost /usr/sbin/kamailio[1137]: DEBUG: tm [callid.c:137]: child_init_callid(): callid: '15b1f0d63a718465-1137(a)129.227.83.108' Mar 24 17:31:21 localhost /usr/sbin/kamailio[1127]: DEBUG: <core> [core/action.c:1656]: run_child_one_init_route(): attempting to run event_route[core:worker-one-init] Mar 24 17:31:21 localhost /usr/sbin/kamailio[1136]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1135]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1134]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1133]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1132]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1131]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1129]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1128]: INFO: <core> [main.c:814]: sig_usr(): signal 15 received Mar 24 17:31:21 localhost /usr/sbin/kamailio[1120]: ERROR: ctl [ctl.c:387]: mod_destroy(): ERROR: ctl: could not delete unix socket /var/run/kamailio//kamailio_ctl: Permission denied (13) Mar 24 17:31:21 localhost /usr/sbin/kamailio[1120]: ERROR: jsonrpcs [jsonrpcs_fifo.c:595]: jsonrpc_fifo_destroy(): FIFO stat failed: Permission denied If I replace the values in the 2 files as appropriate : In the 'kamctlrc" to RPCFIFOPATH="/tmp/kamailio_rpc_fifo" In the 'kamailio.cfg" to !!define DEFINE_FIFO_NAME "/tmp/kamailio_rpc_fifo" Then kamailo starts : [root@vm-vse02-siprouter1 ~]# ps -ef |grep kam kamailio 1235 1 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1236 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1237 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1238 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1239 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1240 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1241 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1242 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1243 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1244 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1245 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1246 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1247 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio kamailio 1248 1235 0 17:37 ? 00:00:00 /usr/sbin/kamailio -P /var/run/kamailio.pid -m 1024 -M 8 -u kamailio -g kamailio root 1251 1165 0 17:37 pts/0 00:00:00 grep --color=auto kam and I can get result from kamctl/kamcmd commands : [root@vm-vse02-siprouter1 ~]# kamctl dispatcher dump which: no gdb in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/) { "jsonrpc": "2.0", "result": { "NRSETS": 1, "RECORDS": [{ "SET": { "ID": 1, "TARGETS": [{ "DEST": { "URI": "sip:cs1-tool-misc.orange-voicemail.net:5060", "FLAGS": "AP", "PRIORITY": 0 } }] } }] }, "id": 1301 } [root@vm-vse02-siprouter1 ~]# kamcmd dispatcher.list { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:cs1-tool-misc.orange-voicemail.net:5060 FLAGS: AP PRIORITY: 0 } } } } } Now, if I change the fifo patch and name to "/var/run/kamailio/kamailio_rpc_fifo' and apply the following rights on /var/run/ to: chmod 755 kamalio/ chown + kamailio:kamailio kamailio/ then kamailio starts. Is there a reason for these results ? Thanks in advance for your answer. Cordialement Patrick GINHOUX

2 26

RTPengine + kernel module long call RTP drops
by Anthony Joseph Messina 05 Apr '17

05 Apr '17

Recently, I've had to be on a lot of long conference calls and have noticed that after about 18-22 minutes, the audio on the call drops out. There are no errors logged from Asterisk, Kamailio, or RTPengine. The signalling portion of the call is unaffected and continues as normal. My setup is as follows asterisk-14.3.0-7.gita047b25.fc25.x86_64 (14 branch) kamailio-5.0.0-5.git2608015.fc25.x86_64 (5.0 branch) kernel-4.10.8-200.fc25.x86_64 (stable Fedora kernel) rtpengine-5.1.1.1-9.git27af349.fc25.x86_64 (master branch) Yealink or Zoiper softphones Calls are originated as follows using TLS & SRTP Yealink/Zoiper -> Kamailio/RTPengine+kernel -> Asterisk/DAHDI/PSTN Related IPtables output below (generated by firewalld) ~]# iptables -S |grep INPUT -P INPUT ACCEPT -N INPUT_ZONES -N INPUT_ZONES_SOURCE -N INPUT_direct -A INPUT -m conntrack --ctstate RELATED,ESTABLISHED -j ACCEPT -A INPUT -i lo -j ACCEPT -A INPUT -j INPUT_direct -A INPUT -j INPUT_ZONES_SOURCE -A INPUT -j INPUT_ZONES -A INPUT -m conntrack --ctstate INVALID -j DROP -A INPUT -j REJECT --reject-with icmp-host-prohibited -A INPUT_direct -p udp -m udp --dport 30000:40000 -j RTPENGINE --id 0 -A INPUT_ZONES_SOURCE -m set --match-set mss4 src -g IN_mss -A INPUT_ZONES -g IN_public -A IN_public -j IN_public_allow -A IN_public_allow -p udp -m udp --dport 30000:40000 -m conntrack \ --ctstate NEW -j ACCEPT I saw https://github.com/sipwise/rtpengine/issues/102 and https://github.com/sipwise/rtpengine/issues/108, but do not seem to have any issues (that I know of) with the SRTP, though the latter ticket references "call that would last longer than 20 minutes". Can you share suggestions for looking into the cause of the audio drops, especially as nothing appears in the logs? Thanks. -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 6

Kamailio v5.0.1 Released
by Daniel-Constantin Mierla 05 Apr '17

05 Apr '17

Hello, Kamailio SIP Server v5.0.1 stable release is out! This is a maintenance release of the latest stable branch, 5.0, that includes fixes since the release of v5.0.0. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous 5.x.x versions are strongly recommended to be upgraded to v5.0.1. For more details about version 5.0.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/04/kamailio-v5-0-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

nathelper: ping options over tcp
by Abdoul Osséni 05 Apr '17

05 Apr '17

Hi, Is it possible to send ping OPTIONS over tcp or tls? If yes, could you me how? Regards Abdoul.

5 7

Replacing a message body received with a subsequent message sending out of kamailio
by narayan panigrahi 05 Apr '17

05 Apr '17

Hi All, I have a architecture of pjsip,kamailio,freeswitch and one external server. When kamailio is receiving "407"-proxy authentication required from the external server it sends to freeswitch and received "407" from freeswitch.But when replying back to kamailio the freeswitch is adding some extra headers in that "407". Now I need to send the message from kamailio to pjsip the same thing received from external server.So I need to replace the message received from external server with the message received from freeswitch at kamailio. I am using redis db to store the message.Can anybody help me how to configure the kamailio so that it will replace the message with freeswitch message and send the original message received before i.e. from external server. Any suggestion will be highly appreciated. Thanks, Narayan +91-9035438928

2 1

Kamailio 5.0.0 KEMI Lua event routes
by José Seabra 05 Apr '17

05 Apr '17

Hi all, I'm starting using kamailio 5 and was trying to convert my actual routing scripts to new KEMI interface using Lua. In my old script i'm using event routes for the evapi, dialog and dispatcher modules, so i was wondering how can i use it on KEMI. It exports some special Lua function to the KEMI framework? Can anyone give an example please. Many thanks José Seabra

2 3

Dialog Timers- Session Refresh
by Yasin CANER 05 Apr '17

05 Apr '17

2 1

Kamailio Issue
by 노형균 05 Apr '17

05 Apr '17

Hello all, I’m newbie to Kamailio and I’m having problem with installation. I tried to install Kamailio through '/usr/local/sbin/kamdbctl create’ but it showed error that database engine not specified. When I checked the kamctlrc file, the DBENGINE was already MYSQL. (DBENGINE=MYSQL) What am I missing here?? Cheers, Hyoung Kyoon Roh

3 2

Kamailio database fail
by 노형균 05 Apr '17

05 Apr '17

Hello all, I’m having problem again. After solving DBENGINE=MYSQL, I’ve been asked to type password but still get this message. I’ve tried to install mysql-server but it showed the same. What am I missing here? root@ubuntu:/home/parallels# kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Usage: grep [OPTION]... PATTERN [FILE]... Try 'grep --help' for more information. /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql: line 117: [: =: unary operator expected INFO: creating database kamailio ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR: Creating database kamailio failed! Regards, Hyoung Kyoon Roh

2 1

Re: [SR-Users] Via header with i attribute
by DanB 05 Apr '17

05 Apr '17

Hey Daniel, Indeed, call comes in over tcp and goes out over udp. They claim that it is not RFC compatible to add the i. Thanks, DanB

2 1

Planning Kamailio v5.0.1
by Daniel-Constantin Mierla 05 Apr '17

05 Apr '17

Hello, I am considering to release Kamailio v5.0.1 sometime next week, likely on Wednesday, April 5, 2017. Should anyone be aware of issues not listed yet on bug tracker, report them there as soon as possible to try to fix. Soon after, we should release a new version from branch 4.4 and the last one from branch 4.3. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

3 5

Problems with avp_db_load
by Kevin Bushong 04 Apr '17

04 Apr '17

I am using the avpops module to read from a mysql database to look for CFWD information for the call. When a call is placed, it appears the command avp_db_load attempts to read the file but all values come back as NULL. There are no messages in the log and of course the call does not process correctly. We are using a config that is exactly the same on a 4.3.4 version of kamailio using cento 6.7, with a local version of mysql Ver 14.14 Distrib 5.1.73, that works perfectly. When moving the code to kamailio 4.4.4 on Debian 8, Jessie, mysql Ver 14.14 Distrib 5.5.53 the avp_db_load returns no information. Here is the snippet of the code we are using. Since I am not used to troubleshooting kamailio, please let me know what other information you need. Thanks, Kevin Code snippet: #!ifdef WITH_CFWD xlog ("$TF Call to $rU made it into WITH_CFWD in route[LOCATION] block\n"); # Call forwarding is enabled, so lets do a lookup to pull all AVP's for possible call forwarding types/destinations # Below line should load all AVP's into memory from DB for the username portion of the TO: field from table usr_preferences xlog("script should load from DB usr_preferences next"); avp_db_load("$tU","s/usr_preferences"); # This line i simple want to print to log when the avp load is complete and what the values are (for debug purposes only and should typically be commented out) xlog("DB load should be complete"); xlog("SCRIPT: $TF Call Forwarding Unconditional set to $avp(Unconditional)\n"); xlog("SCRIPT: $TF Call Forwarding NoAnswer set to $avp(NoAnswer)\n"); xlog("SCRIPT: $TF Call Forwarding Busy set to $avp(Busy)\n"); avp_print(); # Check to see if the AVP's are set and if so put them in log for logging purposes if(is_avp_set("$avp(Unconditional)")) xlog("SCRIPT: $TF Call Forwarding Unconditional set to $avp(Unconditional)\n"); if(is_avp_set("$avp(NoAnswer)")) xlog("SCRIPT: $TF Call Forwarding NoAnswer set to $avp(NoAnswer)\n"); if(is_avp_set("$avp(Busy)")) xlog("SCRIPT: $TF Call Forwarding Busy set to $avp(Busy)\n"); t_set_fr(16000, 15000); # set fr invite timeout to 16s and no response at all to 15s to meet the specs for call fowarding no answer ringtime # If Call fowarding Unconditional is set, we need to change the dialed number and sent it off to the original users trunk from subscriber table # Check to see if unconditional is set if(is_avp_set("$avp(Unconditional)")) # Shove this call into route [PSTN_CFWD] Block route("PSTN_CFWD"); Here is the syslog for the same snippet: 09:29 2017 Call to 15703771055 made it into WITH_CFWD in route[LOCATION] block Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: ERROR: <script>: script should load from DB usr_preferences next Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: ERROR: <script>: DB load should be complete Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: ERROR: <script>: SCRIPT: Wed Mar 29 14:09:29 2017 Call Forwarding Unconditional set to <null> Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: ERROR: <script>: SCRIPT: Wed Mar 29 14:09:29 2017 Call Forwarding NoAnswer set to <null> Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: ERROR: <script>: SCRIPT: Wed Mar 29 14:09:29 2017 Call Forwarding Busy set to <null> Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: INFO: avpops [avpops_impl.c:1473]: ops_print_avp(): p=0x7f440ed86f90, flags=0x0113 Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: INFO: avpops [avpops_impl.c:1477]: ops_print_avp(): #011#011#011name=<direction> Mar 29 14:09:29 Limitless-Calling-Server /usr/local/sbin/kamailio[7831]: INFO: avpops [avpops_impl.c:1485]: ops_print_avp(): #011#011#011val_str=<INCOMING / 8>

2 8

Why these 2 items in the same entry for a htable
by Ginhoux, Patrick 04 Apr '17

04 Apr '17

Hi, While working on the project to migrate my kamailio.cfg script from kamailio 3.3.1 to 5.0.x, I discovered that in there are 2 items in the same htable. Below the figure that illustrate this case : 1) Content of the MBXRANGE table : mysql> select * from mbxrange; +-----+-------------+-------------------------------------------------+----- -------+----------+ | id | key_name | key_value | value_type | key_type | +-----+-------------+-------------------------------------------------+----- -------+----------+ | 0 | VERSION | 120104-173400 | 0 | 0 | | 101 | 1 | min=0100000000;max=0199999999;node=OPMVTS1VSE02 | 0 | 0 | | 102 | 2 | min=0200000000;max=0299999999;node=OPMVTS1VSE02 | 0 | 0 | | 103 | 3 | min=0300000000;max=0399999999;node=OPMVTS1VSE02 | 0 | 0 | | 104 | 4 | min=0400000000;max=0499999999;node=OPMVTS1VSE02 | 0 | 0 | | 105 | 5 | min=0500000000;max=0599999999;node=OPMVTS1VSE02 | 0 | 0 | | 106 | 6 | min=0600000000;max=0699999999;node=OPMVTS1VSE02 | 0 | 0 | | 107 | 7 | min=0700000000;max=0799999999;node=OPMVTS1VSE02 | 0 | 0 | | 108 | 8 | min=0800000000;max=0899999999;node=OPMVTS1VSE02 | 0 | 0 | | 109 | 9 | min=0900000000;max=0999999999;node=OPMVTS1VSE02 | 0 | 0 | | 199 | maxmbxrange | 9 | 0 | 0 | +-----+-------------+-------------------------------------------------+----- -------+----------+ 11 rows in set (0.00 sec) 2) modparam instruction used in my kamailio.cfg script: modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") 3) Result of < kamctl fifo sht_dump mbxrangeHash > command with kamailio version 3.3.x: [root@op52is4router1 ~]# kamctl fifo sht_dump mbxrangeHash Entry:: 0 6:: min=0600000000;max=0699999999;node=OPMVTS1VSE02 Entry:: 1 7:: min=0700000000;max=0799999999;node=OPMVTS1VSE02 Entry:: 2 4:: min=0400000000;max=0499999999;node=OPMVTS1VSE02 Entry:: 3 5:: min=0500000000;max=0599999999;node=OPMVTS1VSE02 Entry:: 4 2:: min=0200000000;max=0299999999;node=OPMVTS1VSE02 Entry:: 5 3:: min=0300000000;max=0399999999;node=OPMVTS1VSE02 VERSION:: 120104-173400 Entry:: 6 maxmbxrange:: 9 Entry:: 7 1:: min=0100000000;max=0199999999;node=OPMVTS1VSE02 Entry:: 14 9:: min=0900000000;max=0999999999;node=OPMVTS1VSE02 Entry:: 15 8:: min=0800000000;max=0899999999;node=OPMVTS1VSE02 Result of < kamcmd htable.dump mbxrangeHash > command with kamailio version 5.0.x : [root@vm-vse02-siprouter1 ~]# kamcmd htable.dump mbxrangeHash { entry: 0 size: 1 slot: { item: { name: 6 value: min=0600000000;max=0699999999;node=OPMMMS1VSE02 type: str } } } { entry: 1 size: 1 slot: { item: { name: 7 value: min=0700000000;max=0799999999;node=OPMMMS1VSE02 type: str } } } { entry: 2 size: 1 slot: { item: { name: 4 value: min=0400000000;max=0499999999;node=OPMMMS1VSE02 type: str } } } { entry: 3 size: 1 slot: { item: { name: 5 value: min=0500000000;max=0599999999;node=OPMMMS1VSE02 type: str } } } { entry: 4 size: 1 slot: { item: { name: 2 value: min=0200000000;max=0299999999;node=OPMMMS1VSE02 type: str } } } { entry: 5 size: 2 slot: { item: { name: 3 value: min=0300000000;max=0399999999;node=OPMMMS1VSE02 type: str } item: { name: VERSION value: 120104-173400 type: str } } } { entry: 6 size: 1 slot: { item: { name: maxmbxrange value: 9 type: str } } } { entry: 7 size: 1 slot: { item: { name: 1 value: min=0100000000;max=0199999999;node=OPMMMS1VSE02 type: str } } } { entry: 14 size: 1 slot: { item: { name: 9 value: min=0900000000;max=0999999999;node=OPMMMS1VSE02 type: str } } } { entry: 15 size: 1 slot: { item: { name: 8 value: min=0800000000;max=0899999999;node=OPMMMS1VSE02 type: str } } } So, I can see : - The entry number is not incremental 1 by 1 ; there are entry #1 to #7, then #14 and #15 (all have size=1) - For the entry #5, its size is of 2 and it contains 2 items I don't say here that this is a problem, but the way the htable is loaded seems strange to me. So if there people who can explain me if what I see is normal or not, and how the htable load process works, I would appreciate. Thanks in advance. Cordialement Patrick GINHOUX

2 4

Via header with i attribute
by DanB 04 Apr '17

04 Apr '17

Hey Guys, Any of you know which module is adding the "i" attribute and maybe if it can be removed safely? The reason would be for some Avaya PBXes discarding the Via completely due to this attribute. Sample header: """ Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKf0d6.72eca17f202d1e73a1f519dbecf7d6a6.0;i=8f1. """ Many thanks, DanB

2 1

nosql database
by Ján Füri 04 Apr '17

04 Apr '17

Hi community, I'd like to ask that witch nosql DB is the most supported by kamalio 5 ? or witch do you guys use in production ? Thanks, Jan

2 1

Re: [SR-Users] ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 04 Apr '17

04 Apr '17

Hi, A second update for people who started to look at this issue. I have identified the real cause of the problem by looking more in the kamailio log with the debug traces. I discover that it is related to the load of the properties table in the htable. I have the following entries in my mysql db : +-------+----------------+---------------------------------------------------------+------------+----------+ | id | key_name | key_value | value_type | key_type | +-------+----------------+---------------------------------------------------------+------------+----------+ | 0 | VERSION | 120104-1734 | 0 | 0 | | 1 | mmm-retry | 408=Y;502=Y;503=Y | 0 | 0 | | 2 | mgcf-retry | 408=Y;502=Y;503=Y | 0 | 0 | | 3 | max-mmm-retry | 2 | 0 | 0 | | 4 | max-mgcf-retry | 2 | 0 | 0 | | 5 | target-uri | mgcf=10.10.10.10;mmm-legacy=20.20.20.10;mmm=20.20.20.20 | 0 | 0 | | 6 | pike-status | 503 Service Overload | 0 | 0 | | 7 | mgcf-path | 99 | 0 | 0 | | 16 | mmm-nodeid | OPMMMS1VSE01=1;OPMMMS1VSE02=2 | 0 | 0 | | 17 | mmm-node-path | 1=1;2=2 | 0 | 0 | +-------+----------------+---------------------------------------------------------+------------+----------+ In the kamailio.cfg, the << modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") >> command causes the load of the above table in the htable. In the log we can see, that the load is done in the alphabetic order of the key_name and it fails to load the last entry "VERSION" : Mar 21 10:25:28 localhost /usr/sbin/kamailio[5784]: DEBUG: <core> [db_val.c:169]: db_str2val(): converting BLOB [VERSION] Mar 21 10:25:28 localhost /usr/sbin/kamailio[5784]: DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [0] Mar 21 10:25:28 localhost /usr/sbin/kamailio[5784]: DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [0] Mar 21 10:25:28 localhost /usr/sbin/kamailio[5784]: DEBUG: <core> [db_val.c:169]: db_str2val(): converting BLOB [120104-1734] Mar 21 10:25:28 localhost /usr/sbin/kamailio[5784]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) If I add a new entry named "WWWWW", then the failure occurs on this entry that is the last alphabetic entry : | 10000 | WWWWW | toto | 0 | 0 | Mar 21 10:28:45 localhost /usr/sbin/kamailio[5817]: DEBUG: <core> [db_val.c:169]: db_str2val(): converting BLOB [WWWWW] Mar 21 10:28:45 localhost /usr/sbin/kamailio[5817]: DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [0] Mar 21 10:28:45 localhost /usr/sbin/kamailio[5817]: DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [0] Mar 21 10:28:45 localhost /usr/sbin/kamailio[5817]: DEBUG: <core> [db_val.c:169]: db_str2val(): converting BLOB [toto] Mar 21 10:28:45 localhost /usr/sbin/kamailio[5817]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Attached the kamailio log for each test. I hope this help. Have anyone an idea of this problem ? Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : mardi 21 mars 2017 09:41 À : Kamailio users (sr-users(a)lists.sip-router.org) <sr-users(a)lists.sip-router.org> Objet : RE: ht_db_load_table(): key type must be string (type=6) Hi, There was a mistake in the file attached ; the short kamailio.cfg file (that makes kamailio to start) was not the correct one; attached the correct one. Have anyone an idea of this problem ? Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : lundi 20 mars 2017 17:58 À : Kamailio users (sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>) <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Objet : RE: ht_db_load_table(): key type must be string (type=6) Hi, To figure out what causes this issue, I work to shorten my kamailio.cfg to contains loadmodule/modparam instructions for the htable module only. My first attempt (see kamailio.cfg.ok) shows that the problem doesn't occur. In the second attempt (see kamailio.cfg.ko) I add instructions previously removed from the original .cfg file. I have determined that, depending on the instructions that follows the "modparam("htable", "fetch_rows", 1000)" line, the "key type must be string (type=6)" error occurs or not. In this ko file, if I comment out the following sequence, the error occurs : modparam("htable", "fetch_rows", 1000) #!!ifdef FIXED_ROUTING #modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") #!!else #modparam("htable", "htable", "abpqHash=>size=12;dbtable=abpq;") #modparam("htable", "htable", "zabpqHash=>size=12;dbtable=zabpq;") #!!endif modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") I hope my explanation is clearn and there are people who can help me on this problem that causes kamailio to not start. Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : vendredi 17 mars 2017 15:46 À : Kamailio users (sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>) <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Objet : ht_db_load_table(): key type must be string (type=6) Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX

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Re: [SR-Users] sr-users Digest, Vol 143, Issue 5
by Narayan P 04 Apr '17

04 Apr '17

Hi Juha, I mean to say that all of the headers in a SIP message. I need to replace all the headers with the headers received before and send to the client. Thanks, Narayan ________________________________ From: sr-users <sr-users-bounces(a)lists.sip-router.org> on behalf of sr-users-request(a)lists.sip-router.org <sr-users-request(a)lists.sip-router.org> Sent: Tuesday, April 4, 2017 10:00:02 AM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 143, Issue 5 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: Issue in replacing message header in kamailio (Narayan P) 2. Re: Issue in replacing message header in kamailio (Juha Heinanen) ---------------------------------------------------------------------- Message: 1 Date: Tue, 4 Apr 2017 09:02:28 +0000 From: Narayan P <narayana88(a)outlook.com> To: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Issue in replacing message header in kamailio Message-ID: <HK2PR0201MB1745D10E07415A9696765176D70B0(a)HK2PR0201MB1745.apcprd02.prod.outlook.com> Content-Type: text/plain; charset="iso-8859-1" Hi Can anybody help me on this.Any API to replace a message header completely in kamailio configuration file and its usage. Thanks, Narayan ________________________________ From: Narayan P Sent: Tuesday, April 4, 2017 6:31:14 AM To: sr-users(a)lists.sip-router.org Subject: Re: Issue in replacing message header in kamailio Hi All, How to replace message header completely that has been received previously in another 200 OK response message. I have used remove_body() and replace_body(re, txt) . Is there any API to replace these. Any suggestion will be highly appreciated. Thanks, Narayan ________________________________ From: Narayan P Sent: Monday, April 3, 2017 1:41:47 PM To: sr-users(a)lists.sip-router.org Subject: Issue in replacing message header in kamailio Hi All, How to replace total message header in another 200 ok response message. I have used remove_body() and replace_body(re, txt) . Is there any API to replace these. Any suggestion will be highly appreciated. Thanks, Narayan

3 2

Issue in replacing message header in kamailio
by Narayan P 04 Apr '17

04 Apr '17

Hi All, How to replace total message header in another 200 ok response message. I have used remove_body() and replace_body(re, txt) . Is there any API to replace these. Any suggestion will be highly appreciated. Thanks, Narayan

2 3

Doubt parsing multipart SDP
by Albert Petit 03 Apr '17

03 Apr '17

Hi list, I am having some issues parsing multipart INVITE [1] when calling force_rtp_proxy . I tried to upgrade to Kamailio 5 to see if maybe i was using too old code but seems not the case. Log of error is: Apr 3 18:31:21 mcptt /usr/sbin/kamailio[3232]: INFO: <script>: forcing RTP proxy for INVITE message Apr 3 18:31:21 mcptt /usr/sbin/kamailio[3232]: ERROR: <core> [core/parser/parse_content.c:370]: decode_mime_type(): ERROR:decode_mime_type: parse error near in [application/vnd.3gpp.mcptt-info+xml] char[51][3] offset=16 So seems there is some issue using Content Type application/vnd.3gpp.mcptt-info+xml in a multipart SIP message But this content type is accepted at following 3GPP spec: http://www.etsi.org/deliver/etsi_ts/129100_129199/129165/13.06.00_60/ts_129… Checking implementation of parse_content.c I see ". - _ +" characters are accepted so i am not sure what is failing. I did some test and if instead of that Content-Type: application/vnd.3gpp.mcptt-info+xml i use Content-Type: application/mcptt no errors are seen in Kamailio logs and ports are replaced succesfully. So do you see something wrong on what i am doing with attached info? INVITE sip:mcpttparticipating@demo.net SIP/2.0 Record-Route: <sip:10.1.1.123:6074;transport=tcp;lr=on;ftag=i3b01a;nat=yes> Call-ID: db6b10362447354bc9819df9a9046ecb(a)10.1.1.222 CSeq: 1 INVITE From: <sip:blue1@demo.net>;tag=i3b01a To: <sip:ccw@demo.net;poc-group=blues> Via: SIP/2.0/TCP 10.1.1.123:6074 ;branch=z9hG4bK15a9.1c063dec40e772152f44c026a1627082.0;i=f Via: SIP/2.0/TCP 10.1.1.222:5070 ;received=10.1.1.222;rport=55173;branch=z9hG4bK-363135-1750dc6e97437e4788f446f2d818b60a Max-Forwards: 69 Contact: <sip:blue1@10.1.1.222:55173 >;transport=tcp;+g.3gpp.mcptt;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mcptt";isfocus Accept-Contact: *;+g.3gpp.mcptt;require;explicit Accept-Contact: *;+g.3gpp.icsi-ref="urn:urn-7:3gpp-service.ims.icsi.mcptt";require;explicit P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mcptt P-Preferred-Identity: <sip:blue1@demo.net> User-Agent: PoC-client/3GPP-Rel13 DemoClient/v2.03.01/BV6000/SDK23 Session-Expires: 3600 Min-SE: 90 Supported: replaces, 100rel, timer, norefersub, timer, answermode, eventlist Resource-Priority: mcpttp.5 Content-Type: multipart/mixed;boundary=gnkboundary Content-Length: 997 --gnkboundary Content-Type: application/sdp v=0 o=- 3626700268 3626700268 IN IP4 10.1.1.222 s=spimedia c=IN IP4 10.1.1.222 t=0 0 a=X-nat:0 m=audio 10002 RTP/AVP 106 a=rtcp:10003 IN IP4 10.1.1.222 i=speech a=rtpmap:106 AMR-WB/16000 a=fmtp:106 mode-change-period=1; mode-change-capability=2; mode-change-neighbor=0; max-red=0 a=sendrecv a=ptime:100 a=maxptime:400 m=application 10003 udp MCPTT a=fmtp:MCPTT mc_queueing;mc_priority=1 --gnkboundary Content-Type: application/vnd.3gpp.mcptt-info+xml <?xml version="1.0" encoding="UTF-8"?> <mcpttinfo xmlns="urn:3gpp:ns:mcpttInfo:1.0" xmlns:xsi=" http://www.w3.org/2001/XMLSchema-instance"> <mcptt-Params> <session-type>chat</session-type> <mcptt-request-uri type="Normal"> <mcpttURI>sip:ccw@demo.net;poc-group=blues</mcpttURI> </mcptt-request-uri> <mcptt-client-id type="Normal"> <mcpttURI>urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF</mcpttURI> </mcptt-client-id> </mcptt-Params> </mcpttinfo> --gnkboundary-- Thanks! Albert

2 1

Doubt about MI Datagram module in Kamailio 5
by Albert Petit 03 Apr '17

03 Apr '17

Dear list, I've recently upgraded to Kamailio 5 in order to avoid issue parsing SDP multipart bodies which think are fixed in the update ( - symbol rejected in content type) . I could upgrade without issues from my kamailio 4 dev deployment but i cannot load anymore MI_Datagram module (seems not present in Kamailio 5 , MI Datagram module i refer to: http://www.kamailio.org/docs/modules/4.3.x/modules/mi_datagram.html ) Can i use MI Datagram module still in Kamailio 5 ? Thanks Albert

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Exclusive: Quitting the fast life of open-source SIP
by Alex Balashov 03 Apr '17

03 Apr '17

(Filed by UC Celeb Talk - 1 Apr 2017.) BERLIN (Tegel), GERMANY--In the now-infamous breakdown episode on the VIP tarmac at Berlin's Tegel Airport, Alex Balashov shocked the IP telecoms celebrity world last month. Shielding his eyes from the punishingly bright daylight of an overcast day in northern Germany, Balashov climbed out of his Gulfstream G650 jet, clutching a half-finished bottle of Woodford Reserve, and announced his sudden retirement from the "fast life" of open-source SIP consulting to a waiting gathering of techn business reporters. "I'm not with this 'live fast, die young' VoIP s*** anymore," he reportedly shouted, before collapsing and vomiting profusely on the red carpet. UC Celeb Talk's own Benjamin Brant sat down with Balashov in an exclusive interview to learn more about what happened, even as the telecoms industry continues to digest the shock meltdown. -- BRANT: Everyone's dying to know, what was going through you head at that moment? BALASHOV: Funny how you mention dying. I want to live to see 35, man. I can't handle the SIP biz anymore. BRANT: What do you mean? You seemed to have all the makings of an open-source SIP superstar. BALASHOV: I can't keep up with the drinks anymore, the three day-long parties, the paparazzi, always watchin' my back at ClueCon... BRANT: When'd you realise it might be time to call it? BALASHOV: I'll tell you when. It was Kamailio World '16. Man, those vodka + Club Mate drinks they pour down at Fokus are something else. They'll really get you thinking about things. BRANT: So you had a cocktail and thought better of your entire life as a SIP celebrity? The fans, the adulation, the glory? BALASHOV: When you wake up at 3 PM at the Park Inn on Alexanderplatz with a $31,000 room service bill on your AMEX, a goat wandering around, and reporters in your face, you tell me with a straight face you're not gonna take stock of your life and make some changes. BRANT: That really happened? BALASHOV: That's what I'm trying to tell you! That's Kamailio World, dude. That's the SIP software engineering culture. You wanna ITSP and WebRTC? You're gonna have to bro down! BRANT: Surely not everywhere. What about the IETF? BALASHOV: Those guys drink more than ANYONE. Have you seen the SIP Outbound RFC? BRANT: That's the work-product of an alcohol-fuelled creative orgy? BALASHOV: It's the work product of something-something. I don't know if I'm allowed to identify the uh, substance in question. BRANT: How are you going to live without the private jet? If I had a G6, I wouldn't just turn over the keys. BALASHOV: Yeah, I mean, open-source VoIP consulting has its upsides. It's a fine way to make a living. But it's just not worth it. Besides, hourly operating costs are through the roof. BRANT: But you were an icon, and quite a ladies' man too! How do you walk away from that? BALASHOV: Look man, of course women are gonna mob you when you're a SIP consulting star. That's just part of the open-source tech life in general. But it's fake, man. They don't love you. They don't want to know who you are inside. They just want you 'cause you know all about the SIP transaction state machine, loose vs. strict routing, cross-compiling Kamailio. They just got a crush on you 'cause you know presence and parallel forking race conditions. I want something real. BRANT: But that's part of the social bargain of being an open-source SIP celebrity. Didn't you once boast of having uh, lady friends, "in different geographic and non-geographic operator codes"? BALASHOV: Yeah, and that's cool, but I want to come home to a woman who loves me for me, not Mr. 3... ... 02, Moved Temporarily, but not happy necessarily. Feel me? BRANT: What happened at ClueCon? BALASHOV: You talking about the bullets Acme Packet ventilated me with? BRANT: Five of them, as I heard. BALASHOV: Well I'm still standin'. Took those shots and smiled. BRANT: I know you guys have a kind of omertà about this stuff, but I've got to ask: why? BALASHOV: I ain't worried, kids growing up in today's world need to learn the truth. They act like I was dealing transcoding licences on their block. O.G. 729. BRANT: But that wasn't what gave you a wake-up call that the SIP game is sharper than a razor blade? BALASHOV: Hell no, it wasn't their block. That fake-ass east coast channel partner crew can cash me outside howboudah? BRANT: How exclusive is Kamailio World? BALASHOV: It's one of the most exclusive clubs I've ever hit up. Lines wrapping around blocks. And the bouncers. Damn. BRANT: The bouncers? BALASHOV: Those dudes are harsh. BRANT: On a scale of 1 to Berghain, how bad is it? BALASHOV: All I know is, one time some squirrely guy in the audience asked a moderately new-blood question like, "If a Record-Route header has been added by an intermediate proxy, is it proper of a SIP User Agent to populate the Request URI of subsequent in-dialog requests with the Record-Route URI, or should it be the remote Contact?" Half the room got up like it was ready to curb-stomp that little face. Guy didn't know what he got himself into--a room full of elites. BRANT: So what happened? BALASHOV: Someone was like, "Bounce this fool! He don't know how to act." So these three muscular Kaiser Wilhelm-looking dudes--moustaches and all--in sunglasses roll up and drag this guy through the Atrium, out the doors and dump him straight out onto Anna-Louisa-Karsch-Straße. BRANT: Damn. Ejected out of the saloon. BALASHOV: Yo, that dude was dismissed from the premises. BRANT: So what do you see in your future post-SIP life? BALASHOV: I'm thinking about starting a bakery. A friend threw it out there as a way to kind of atone for all this and get straight with Jesus. Gonna bake really big cookies. You know, huge ones, like a small pizzas? But recognise--they're actually cookies. BRANT: Got any advice for those who bright young upstarts who haven't turned over a new leaf and thrown in the towel yet? BALASHOV: Come on down to Kamailio World #5, May 8-10 2017 in Berlin. You'll see your life in a new way. Registration and more information online at https://www.kamailioworld.com/k05/

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Kamailio 6.0 architecture
by Olle E. Johansson 03 Apr '17

03 Apr '17

Dear Kamailians! We got news for you! The Kamailio 6.0 architecture is finalised! It's something we've been working on for a long time in the project team and you've already seen part of it in the recent Kamailio 5 release - the KEMI interface for LUA. We've realised that we have too many modules, especially all those IMS* modules. And the core is getting slimmer and slimmer. This is what we're going to do: * All RPC, EVENTAPI,AUTH, JSON, TM, HTTP_CLIENT and TLS modules will be integrated back into the core. * The EVENT API will be the core API for external modules * The rest of the functionality in the modules will have to be rewritten in LUA and maintained as separate GITHUB projects outside of the core After successfully integrating LUA into the heart of Kamailio and exposing the Kamailio API we have decided to make life more simple for the core maintainers. Maintaining well over 100 modules took too much time, and who cares about some of the stuff? Billing, least cost routing and all the other crazy stuff in modules are going away anyhow as all services will be free. Calls really wants to be free. And presence is no longer needed, we'll just tap into the mobile phones from the core to know where everyone is. The removal of the modules will take place during the month of April 2017 and a new core will raise like a Fenix from the ashes, a much larger, monolithic and complex core with new functionality that will take Kamailio to new areas, like a message bus for games, an API server for mobile yoga apps and a control plane for embedded IoT devices. "Kamailio everywhere" is the new slogan for the Kamailio project. - "We will all remember April 1st 2017 as a new start of the project" says Olle-Carsten Mierla, the new elected project leader. "From now on, Kamailio aims to be a standard part of every operating system distribution, defining the new core of the Internet.” We hope all of you join us in this effort and refrain from reporting any bugs or other disturbing issues while we rebuild the new Kamailio. /O

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Re: [SR-Users] Exclusive: Quitting the fast life of open-source
by JR Richardson 01 Apr '17

01 Apr '17

Brilliant! Bravo! Bravo! Brovo! Gonna miss you Bro! Get in this neck of the woods, got some dope a$$ hooch and BBQ waitin! Peace out! JR Richardson Engineering for the Masses Chasing the Azeotrope ---------------------------------------------------------------------- Message: 1 Date: Sat, 1 Apr 2017 01:02:58 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Exclusive: Quitting the fast life of open-source SIP BRANT: But you were an icon, and quite a ladies' man too! How do you walk away from that? BALASHOV: Look man, of course women are gonna mob you when you're a SIP consulting star. That's just part of the open-source tech life in general. But it's fake, man. They don't love you. They don't want to know who you are inside. They just want you 'cause you know all about the SIP transaction state machine, loose vs. strict routing, cross-compiling Kamailio. They just got a crush on you 'cause you know presence and parallel forking race conditions. I want something real. BRANT: But that's part of the social bargain of being an open-source SIP celebrity. Didn't you once boast of having uh, lady friends, "in different geographic and non-geographic operator codes"? BALASHOV: Yeah, and that's cool, but I want to come home to a woman who loves me for me, not Mr. 3... ... 02, Moved Temporarily, but not happy necessarily. Feel me?

1 0

Configuring Kamailio as a "VoIP traffic reflector"
by Tahiro Hashizume 31 Mar '17

31 Mar '17

Hi there, This e-mail is going to be lengthy and I hope you will not get frustrated (like I am, for different reasons described later) by reading it all from the beginning to the end since I have just warned you :P You have absolutely no need for reading the entire post unless you are crazy enough read about what I do for fun in given circumstances. Basically, I would like Kamailio and Asterisk to run on the same box in such a way that A.)Asterisk listens on eth0 (on 10.1.1.15/24 with ProxyARP enabled) for incoming SIP register requests from VoIP-ATAs B.)Kamailio listens on five IP addresses on dummyN which has 10.1.1.1N/32 (N=0-4, so dummy0 has 10.1.1.10/32, for example) making five SIP+RTP sessions to the same remote host to appear as if they are from five different hosts - so that a single instance of Asterisk running on debian may be registered (through Kamailio) on the same remote host by using five different accounts and source IP addresses. I know Asterisk alone cannot do this (even by running multiple instances of it) and that there have to be five instances of both SIP-Proxy and RTP-Proxy so that the source address can be literally spoofed. Should be simple, but so far cannot find good sources of reference. I need your advice. Thanks in advance. << Backgrounds - read only if you are curious >> The so-called frustration is when your needs are not realized due to technological limitations. I have years worth of frustration on the VoIP setup at home and that's totally NOT your fault. Written below for those who happen to be curious should sound interesting, so here is the situation I'm facing; I'm in Japan and our infrastructure is well within the definition of NGN which in my case includes a low-latency VoIP to replace the traditional PSTN. (NTT will end ISDN in 2020, copper-line PSTN in 2025, as announced so far.) The service is provided by NTT (former state-run telecom) of Japan and works on FTTH which carries: a.)PPPoE for those who wish to get IPv4&IPv6 connectivity - which for the most part is so that users get to choose their favorite ISP. b.)WAN-reachable IPv6 traffic c.)WAN-unreachable IPv4 for SIP based services (address blocks reserved and announced by AS9595) Given the fact that b.) and c.) are only applicable (optionally and respectively) to those who signed up for the native-IPv6 (we call it IPoE, oddly) service and to those who signed up for the VoIP service, it's essentially an L2 transport service for PPPoE to your ISP - very much like the way we had our phone-line and ISP (dial-up) separately signed up for in the old days (hence I call it the access-line service). I've made a switch over to VoIP-based PSTN from the copper-line in April 2011 after seeing that copper-line PSTN served no better in case of emergency like 3.11 when people want to call others to see if their loved one are okay - on the evening through the night of March 11 2011, our traditional PSTN stayed at its capacity limits while VoIP-based means of communication (incl. Skype, surprisingly) stayed intact. That was the wake-up call. Now, upon signing up for the VoIP service, the user receives a box so called HGW (stands for Home Gateway) which is very much a SIP+RTP UA on its WAN side and SIP+RTP server on LAN side (up to 5 UAs may register if the UA supports "sip session-timer"). And this HGW is being the pain that drove me to writing this e-mail - it only allows one SIP-register per IP address. Further, looking into the specs requirements for HGW published by NTT (=there's a centralized documentation for it since the designs and manufacturing of HGW is split across several companies), there's a part which basically says: "One SIP register to HGW from a UA on LAN should be able to handle (but not limited to) one data session. An UA on HGW's LAN side may support multiple-data session per SIP register(optional)." - which to me reasonably explains why my HGW only allows one SIP-register per IP address. My gyess is that the unwritten assumption here is "one sip register"="one physical phone", which is quite understandable. Regarding the fact that NTT optionally provides services like: 1.)additional phone numbers 2.)multi-data sessions(RTP for voice, for example) ...on one trunk. This, and also in accordance with the technical specs. requirements for HGW above means supporting multiple RTP sessions either terminated or originated by an Asterisk box which sits on the HGW's LAN side requires that a single instance of Asterisk on a debian box somehow uses multiple IP addresses (with or without the use of other software). The simplest way of doing this (to me) seemd like running five instances of RTP(proxy)+SIP(proxy) which will each listen on one of five IP addresses, to which the Asterisk instance in question could establish one session of SIP+RTP each. The idea seemed simple enough, but there did not seem like good selections of software years ago. I have once given up on the implementation years ago, altered the NTT-supplied HGW with a third-party product (NVR500 by YAMAHA which on its LAN behaves as a MGCP+RTP server) in order to maximize the stability of the phone system. This was due to the fact that the x86-box I was using was hardware-unstable and that the HGW's behavior was something I did not like dealing with (recently solved). I've decided to make a come-back to Asterisk recently since I recently started using hylafax running on a RasPi with RS232 modem hooked up to it but with missing support for CID since the Japanese equivalent of CID (we call it Number-Display) does not follow the Bellcore standards and the modem I use only supports Bellcore standards- going VoIP again and running iaxmodem could solve this issue. -------------------------------------------- Tahiro Hashizume Department of Electrical and Electronic Engineering at Tokyo Denki University 5 Senju-Asahi-cho Adachi-ku Tokyo, JP1208551 Japan --------------------------------------------

2 4

[SIP Transaction] Handle Cancel message
by Hai Bui Duc Ha 31 Mar '17

31 Mar '17

Hi all, I have problem when use module pusher on Kazoo. The user after receive call from pusher can not receive the Cancel message if have another people pick up this phone. I trace log and debug on Kamailio and see the Freeswitch sent Cancel message but Kamailio reply 481 - Call leg/transaction does not exist. The configure on Kamailio: default.cfg ======================================================= *route[PREPARE_INITIAL_REQUESTS]* *{* * if (is_method("CANCEL")) {* * if(t_lookup_cancel()) xlog("L_INFO","$ci|log|======== t_lookup_cancel()");* * xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CAPTURED IN MAIN---\n");* * if (t_check_trans()) {* * xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CHECK TRANS TRUE---\n");* * //xlog("L_INFO", "$ci|log|====================================");* * route(RELAY);* * } else {* * xlog("L_INFO","$rm from $fu (IP:$si:$sp) ---CHECK TRANS FALSE---\n");* * sl_send_reply("481", "Call leg/transaction does not exist");* * }* * exit();* * } else if (is_method("ACK")) {* * if (t_check_trans()) {* * route(RELAY);* * }* * exit();* * }* ======================================================= As I read on http://kamailio.org/docs/modules/4.3.x/modules/tm.html#tm.f.t_check_trans Kamailio see the cancel message not same the transaction INVITE message. INVITE message: ------------------------------------------------------------------------ INVITE sip:100@quydang.htk.cvoice SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:11000;rport;branch=z9hG4bKK9jyp8mXUZHgS Route: <sip:192.168.1.77:5060> Max-Forwards: 49 From: "02 quy" <sip:102@quydang.htk.cvoice>;tag=04K2SappeyDtj To: <sip:100@quydang.htk.cvoice> Call-ID: 19fa9228-ef95-11e6-b473-1f645951fd7b CSeq: 103030653 INVITE Contact: <sip:mod_sofia@192.168.1.77:11000> User-Agent: Anttel Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-AUTH-IP: 192.168.1.141 X-AUTH-PORT: 64799 X-KAZOO-AOR: sip:100@quydang.htk.cvoice X-KAZOO-PUSHER-Token-Proxy: sip:192.168.1.77:5060 X-KAZOO-PUSHER-Token-ID: 803F0D6610AC5979C6F0513A3A4BE62E0BAF1530DC7A98C2C0DC13D784585FBE X-KAZOO-PUSHER-Token-Type: apple X-KAZOO-PUSHER-Token-App: com.htkinc X-FS-Support: update_display,send_info Remote-Party-ID: "02 quy" <sip:102@quydang.htk.cvoice >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1486708670 1486708671 IN IP4 192.168.1.77 s=FreeSWITCH c=IN IP4 192.168.1.77 t=0 0 m=audio 24764 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ CANCEL message: ------------------------------------------------------------------------ CANCEL sip:100@quydang.htk.cvoice SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:11000;rport;branch=z9hG4bKK9jyp8mXUZHgS Route: <sip:192.168.1.77:5060> Max-Forwards: 49 From: "02 quy" <sip:102@quydang.htk.cvoice>;tag=04K2SappeyDtj To: <sip:100@quydang.htk.cvoice> Call-ID: 19fa9228-ef95-11e6-b473-1f645951fd7b CSeq: 103030653 CANCEL Reason: SIP;cause=200;text="Call completed elsewhere" Content-Length: 0 ------------------------------------------------------------------------ How can I know it same transaction ? We have any idea to solve this problem ? Thank for advice ! Regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

2 9

$hep(dst_ip)
by David Villasmil 30 Mar '17

30 Mar '17

Hello guys, the merged https://github.com/kamailio/kamailio/pull/995 is not available on 5.0 and I'm wondering if/when it will be added to 5.0? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 6

Adding two time prefix to $rU
by Yebe diomande Keffa 29 Mar '17

29 Mar '17

Hello I am new with kamailio, and trying to add prefix to Request URI ($rU). I use branch_route, for $rU=”prefix”+$rU, i got two time prefix adding => prefix_prefix_original$rU This is my code small code: request_route { if (is_method("INVITE")) { $var(ip1)="51.254.245.58"; $var(port1)="33"; if(!t_is_set("branch_route")) route(FWD); xlog("L_ALERT","Route FWD is executed from ip:$si to ip:$rd \n"); } exit; } route[FWD] { t_on_branch("CHECK"); t_relay(); } branch_route[CHECK] { $rd=$var(ip1); $rU=$var(port1)+$rU; xlog("L_ALERT","dst rewrited: $rU \n"); } LOG FILES : Mar 28 19:34:45 ns3033789 systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Mar 28 19:34:54 ns3033789 /usr/sbin/kamailio[2859]: ALERT: <script>: dst rewrited: 3322508983432 Mar 28 19:34:54 ns3033789 /usr/sbin/kamailio[2859]: ALERT: <script>: Route FWD is executed from ip:51.254.245.57 to ip:5.135.177.203 Mar 28 19:34:54 ns3033789 /usr/sbin/kamailio[2859]: ALERT: <script>: dst rewrited: 333322508983432 Mar 28 19:34:54 ns3033789 /usr/sbin/kamailio[2859]: ALERT: <script>: Route FWD is executed from ip:5.135.177.203 to ip:51.254.245.58 Mar 28 19:35:01 ns3033789 CRON[2915]: (root) CMD (/usr/local/rtm/bin/rtm 8 > /dev/null 2> /dev/null) ^C root@ns3033789:~# May someone advise me Thank you in advance

2 1

log_custom module question
by Anonim Stefan 29 Mar '17

29 Mar '17

Hi, I'm trying some logging via UDP, using log_custom module. I've noticed that the logs are always sent through the first interface in the config "listen=" rows. Now, is there a way to listen on that interface, in order to log_udp() on it, without being able to send/receive any SIP on it? (i.e. disabling SIP capability and use it only for logging via UDP) Thank you, Stefan

2 1

Nightly build config error
by David Villasmil 28 Mar '17

28 Mar '17

Hello guys, I have in my apt the nightly build repo, which i use fairly regularly for testing. Today i tried installing kamailio 5 and got the following: # apt-get install -y kamailio Reading package lists... Done Building dependency tree Reading state information... Done The following packages were automatically installed and are no longer required: libdbd-mysql-perl libdbi-perl libev4 libevent-2.0-5 libjansson4 libmysqlclient18 libsctp1 libterm-readkey-perl libunistring0 mysql-client mysql-client-5.5 mysql-common Use 'apt-get autoremove' to remove them. Suggested packages: kamailio-berkeley-modules kamailio-carrierroute-modules kamailio-cpl-modules kamailio-dbg kamailio-ldap-modules kamailio-lua-modules kamailio-mysql-modules kamailio-perl-modules kamailio-postgres-modules kamailio-presence-modules kamailio-python-modules kamailio-radius-modules kamailio-snmpstats-modules kamailio-tls-modules kamailio-unixodbc-modules kamailio-xml-modules kamailio-xmpp-modules The following NEW packages will be installed: kamailio 0 upgraded, 1 newly installed, 0 to remove and 42 not upgraded. Need to get 0 B/5304 kB of archives. After this operation, 24.5 MB of additional disk space will be used. Selecting previously unselected package kamailio. (Reading database ... 37888 files and directories currently installed.) Preparing to unpack .../kamailio_5.0.0+0~20170322012605.24+jessie_amd64.deb ... Unpacking kamailio (5.0.0+0~20170322012605.24+jessie) ... Processing triggers for systemd (215-17+deb8u4) ... Processing triggers for man-db (2.7.0.2-5) ... Setting up kamailio (5.0.0+0~20170322012605.24+jessie) ... Job for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. invoke-rc.d: initscript kamailio, action "start" failed. dpkg: error processing package kamailio (--configure): subprocess installed post-installation script returned error exit status 1 Processing triggers for systemd (215-17+deb8u4) ... Errors were encountered while processing: kamailio [master 5dad0af] committing changes in /etc after apt run 16 files changed, 2992 insertions(+) create mode 100644 default/kamailio create mode 100755 init.d/kamailio create mode 100644 kamailio/dictionary.kamailio create mode 100644 kamailio/kamailio-advanced.cfg create mode 100644 kamailio/kamailio-basic.cfg create mode 100644 kamailio/kamailio.cfg create mode 100644 kamailio/kamctlrc create mode 120000 rc0.d/K02kamailio create mode 120000 rc1.d/K02kamailio create mode 120000 rc2.d/S03kamailio create mode 120000 rc3.d/S03kamailio create mode 120000 rc4.d/S03kamailio create mode 120000 rc5.d/S03kamailio create mode 120000 rc6.d/K02kamailio create mode 120000 systemd/system/multi-user.target.wants/kamailio.service Counting objects: 1321, done. Compressing objects: 100% (749/749), done. Writing objects: 100% (1321/1321), done. Total 1321 (delta 127), reused 1320 (delta 126) E: Sub-process /usr/bin/dpkg returned an error code (1) Checking up: root@homer-dev ~/homer-docker# systemctl status kamailio.service * kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled) Active: failed (Result: exit-code) since Mon 2017-03-27 16:58:09 UTC; 7s ago Mar 27 16:58:09 homer-dev kamailio[9438]: ERROR: <core> [core/cfg.y:3264]: yyparse(): cfg. parser: failed to find command t_precheck_trans (params 0) Mar 27 16:58:09 homer-dev kamailio[9438]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 479,...oadmodule? Mar 27 16:58:09 homer-dev kamailio[9438]: ERROR: <core> [core/cfg.y:3264]: yyparse(): cfg. parser: failed to find command t_check_trans (params 0) Mar 27 16:58:09 homer-dev kamailio[9438]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 480,...oadmodule? Mar 27 16:58:09 homer-dev kamailio[9438]: ERROR: <core> [core/cfg.y:3264]: yyparse(): cfg. parser: failed to find command t_check_trans (params 0) Mar 27 16:58:09 homer-dev kamailio[9438]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 483,...oadmodule? Mar 27 16:58:09 homer-dev systemd[1]: kamailio.service: control process exited, code=exited status=255 Mar 27 16:58:09 homer-dev systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Mar 27 16:58:09 homer-dev systemd[1]: Unit kamailio.service entered failed state. Mar 27 16:58:09 homer-dev kamailio[9438]: ERROR: bad config file (46 errors) Hint: Some lines were ellipsized, use -l to show in full. All modules are there, the problem is the startup script can't find where the modules are installed? If i just un-comment the "mpath" in the config file, kamailio starts normally. /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" I'm not 100% sure "mpath" was set before, but i don't remember having to set that up in the near past... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ

2 2

Kamailio Registration load Issue
by Jade SZ 28 Mar '17

28 Mar '17

Hi Guys, I am running a simple REGISTER load test on: 1) Kamailio sever with 2 cores - mem 5G 2) Kamailio server with 4 cores - mem 16G Both are EC2 instances. At -r = 500 i.e. 500 reg/sec sipP test works fine with very few re-trans. But when i increase it to 800 reg/sec it starts retransmissions in bulk. I don't see server's CPU, load-avg or memory shooting. Running everything by default, even using kamctl to start the instance. So I have not tuned any params yet. My main concern is how can I make server choke and get its actual capacity, and avoid these retransmissions as apparently kamailio is not even utilizing 2 cores and CPU usage is under 10 always. Results of both server is same i.e. 500 reg/sec max so I am sure there is some problem that needs a fix, but need some hints here. Also used multiple SIPp's to rule out if it is SIPp issue, but after 500 I see same problem. Regards, JSZ

4 6

kamailio 4.4.1 crash on bad sip response
by David Escartín Almudévar 28 Mar '17

28 Mar '17

hello all we have experienced a crash and tracing the logs and the core, seems it was because a sip response from an endpoint. a UDP receiver (26248) crashed and the last message we see on it is a 487 quite bad formed Mar 24 01:58:02 mia-proxy-inout-1-stby /usr/local/kamailio/sbin/kamailio[26248]: ERROR: tm [t_lookup.c:1055]: t_check_msg(): ERROR: reply doesn't have a via, cseq or call-id header Mar 24 01:58:17 mia-proxy-inout-1-stby /usr/local/kamailio/sbin/kamailio[26230]: ALERT: <core> [main.c:739]: handle_sigs(): child process 26248 exited by a signal 11 the backtrace of the core (gdb) backtrace #0 0x0000000000457ab9 in get_hash1_case_raw (s=0x0, len=0) at hashes.h:210 #1 0x000000000045b472 in _dns_hash_find (name=0x7f6906943188, type=1, h=0x7fff120793cc, err=0x7fff120793ac) at dns_cache.c:535 #2 0x0000000000461285 in dns_hash_get (name=0x7f6906943188, type=1, h=0x7fff120793cc, err=0x7fff120793ac) at dns_cache.c:762 #3 0x0000000000467194 in dns_get_entry (name=0x7f6906943188, type=1) at dns_cache.c:2102 #4 0x0000000000468a05 in dns_a_get_he (name=0x7f6906943188) at dns_cache.c:2432 #5 0x0000000000468bb9 in dns_get_he (name=0x7f6906943188, flags=1) at dns_cache.c:2505 #6 0x00000000004696c4 in dns_srv_sip_resolvehost (name=0x7f6906943188, port=0x7fff120795e2, proto=0x7fff120795e1 "\001\330\023") at dns_cache.c:2679 #7 0x000000000046aa37 in dns_sip_resolvehost (name=0x7f6906943188, port=0x7fff120795e2, proto=0x7fff120795e1 "\001\330\023") at dns_cache.c:2849 #8 0x000000000049519e in update_sock_struct_from_via (to=0x7fff12079708, msg=0x7f69069a1dd8, via=0x7f69068f82a8) at forward.c:704 #9 0x0000000000495ee5 in do_forward_reply (msg=0x7f69069a1dd8, mode=0) at forward.c:766 #10 0x00000000004970af in forward_reply (msg=0x7f69069a1dd8) at forward.c:849 #11 0x00000000005197ef in receive_msg ( buf=0xab0d80 "SIP/2.0 487 Request Terminated\r\nFrom: \"8888888888 \"<sip:8888888888@7.7.7.7;user=phone>;tag=B7jgc8jQ4m5pB\r\nTo: <sip:555555555@8.8.8.8:5060>;tag=e0d50be-13c4-58d47cba-a2ed9808-36fa\r \nl\337K\016"..., len=367, rcv_info=0x7fff12079a10) at receive.c:299 #12 0x0000000000627b43 in udp_rcv_loop () at udp_server.c:495 #13 0x00000000004b107a in main_loop () at main.c:1600 #14 0x00000000004b842f in main (argc=13, argv=0x7fff12079fb8) at main.c:2616 i have tried to duplicate the issue, but i dont know how to translate l \337K\016 to a xml notation i guess this is some weird that cannot be processed for kamailio could you please take a look and let me know if you know how to duplicate and fix this crash? thanks a lot and regards david

2 9

new keepalive module ?
by Guillaume Bour 28 Mar '17

28 Mar '17

Hi all, I would like to implement destinations keepalive in drouting module (as it is done in dispatcher). But instead of duplicating what's implemented in dispatcher, I think it would be more clever to create a new module dedicated to pinging destinations, and to plug drouting, dispatcher and other modules to this new one What's you opinion about that ? Cheers, Guillaume

2 5

Presence + RLS + XCAP
by Paul Smith 27 Mar '17

27 Mar '17

Hi I am struggling to figure out how to build and test an integrated presence server with Resource-List (rls.so) and XCAP (xcap_server.so). I am confused about what is meant to happen and how to debug it. Not sure if I am suffering from incompatibilities, configuration errors, or bugs… I have not found an up to date guide on configuring an integrated presence, rls, xcap server so I have used old tutorials, module documentation and some course code review … but I’m pretty confused now! I am using Snom phones and Jitsi (on Mac) to test. I have a kamailio Registrar which proxies all presence messages to the separate kamailio 4.2 Presence server. REQUIREMENT : Contact Presence ============================= I would like to get a SUBSCRIBE from the Snom phone to subscribe to a contact-list resource list, and to receive a NOTIFY with all of the presence states for those contacts. Relevant snom settings? Contact List URI: this generates a SUBSCRIBE with event:presence I believe that: ========== presence module should be managing the individual user presence and dialog states xcap module should allow me to use HTTP / curl to edit the resource lists rls module should be managing the mapping from a resource list subscription to the individual presence states and generating the aggregated NOTIFY messages. each device should then be able to SUBSCRIBE to a resource list uri and then receive an aggregated NOTIFY for all of the watched entities. Current status: =========== SUBSCRIBE sip:org1a-contacts@*****.co.uk SIP/2.0 v: SIP/2.0/UDP 192.168.1.15:59594;branch=z9hG4bK-qllzlgs3xf43;rport f: <sip:org1a@*****.co.uk>;tag=wwasqc6lmb t: <sip:org1a-contacts@****.co.uk> i: 313439303236383530323234313530-x3hlb3j6b5ef CSeq: 1 SUBSCRIBE Max-Forwards: 70 User-Agent: snomD375/8.9.3.46 m: <sip:org1a@192.168.1.15:59594>;reg-id=1 Event: presence Accept: application/pidf+xml, application/rlmi+xml, multipart/related Supported: eventlist Expires: 3600 l: 0 … auth handshake SIP/2.0 202 OK Record-Route: <sip:10.210.160.13;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes> Record-Route: <sip:**pubip**;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes> v: SIP/2.0/UDP 192.168.1.15:59594;received=**recvip**;branch=z9hG4bK-8wufiv1ua3ne;rport=60554 f: <sip:org1a@****.co.uk>;tag=wwasqc6lmb t: <sip:org1a-contacts@*****.co.uk>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-d30e i: 313439303236383530323234313530-x3hlb3j6b5ef CSeq: 2 SUBSCRIBE Expires: 3600 Contact: <sip:10.210.160.16:5060;transport=udp> Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 … and then the snom receives an empty NOTIFY. NOTIFY sip:org1a@192.168.1.15:58325 SIP/2.0 Via: SIP/2.0/UDP 109.104.108.18;branch=z9hG4bKc54a.20db45097fb44595ccd5a30827ca2a62.0 Via: SIP/2.0/UDP 10.210.160.16;rport=5060;branch=z9hG4bKc54a.40150886000000000000000000000000.0 To: <sip:org1a@****.co.uk>;tag=sx4ne9jpcl From: <sip:org1a-contacts@*****.co.uk>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-ed89 CSeq: 2 NOTIFY Call-ID: 313439303237323938323235353133-2du221e6okvf Content-Length: 0 User-Agent: kamailio (4.2.0 (x86_64/linux)) Max-Forwards: 69 Event: presence Contact: <sip:10.210.160.16:5060;transport=udp;alias=10.210.160.16~5060~1> Subscription-State: active;expires=3600 The Subscription is showing up in the watchers table, not the rls_watchers table, so I believe it is being rejected by the rls_handle_subscribe() and then parsed by the presence module. Is that correct? There is noting in rls_watchers or rls_presentity tables. Should there be? I have a rows in the XCAP table for the Resource-List curl -d @/tmp/contacts-resource-list.xml -vvv -X PUT http://109.104.108.23:5060/xcap-root/resource-lists/users/sip:org1a-contact… relevant row in xcap table: | 3 | org1a-contacts | *****.co.uk | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="team"><entry uri="sip:1002@****.co.uk"/><entry uri="sip:1001@****.co.uk"><display-name>bob</display-name></entry><entry uri="sip:1003@****.co.uk"><display-name>test1</display-name></entry><entry uri="sip:1006@****co.uk"><display-name>test1006</display-name></entry><entry uri="sip:1007@****co.uk"><display-name>test1007</display-name></entry><entry uri="sip:1008@h****.co.uk"><display-name>test1008</display-name></entry><entry uri="sip:1009@****.co.uk"><display-name>test1009</display-name></entry> <entry uri="sip:1005@****.co.uk"><display-name>testing1005</display-name></entry> <entry uri=“sip:org1a@****.co.uk"><display-name>testingorg1a</display-name></entry> <entry uri=“sip:org1c@****.co.uk"><display-name>testingorg1c</display-name></entry></list></resource-lists> | 4 | sr-1490270411-10965-2 | 0 | /xcap-root/resource-lists/users/sip:org1a-contacts@****co.uk/index | 0 | I have rows in the presence table for the entities that I want to track e.g. | | | 2889 | org1c | ****.co.uk | presence | a.1490270411.10942.42.1083 | 1490277099 | 1490273499 | <?xml version="1.0 " encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:im="urn:ietf:params:xml:ns:pidf:im" entity=“pres:org1c@*****.co.uk"> <tuple id="snom370-000413260935"> <status><basic>open</basic> <im:im>Available</im:im> </status> <contact priority="1.00">sip:org1c@****.co.uk</contact> <note xml:lang="en">Available</note> </tuple></presence> I am running kamaiio 4.2 Question : Should I have an resource-services row too? Or does rls.so infer the relationship using the URI? When I use jitsi to manipulate its contact list it only inserts 2 rows into the xcap table and I cannot get a SIP Subscribe to connect to that resource list either. Question : What should the curl PUT commands and XML docs look like to build a valid RLS service (is there a good example anywhere)? Question : Is there an example config or doc anywhere showing how to build integrated presence, rls, xcap? Question : Has anyone got this working with Snom and or other handsets… the standards and implementations seem vague / inconsistent? Note: There is another Snom setting (Extension Monitoring Call Pickup List URI:) which produces a SUBSCRIBE with Event:dialog but which should allow a button/lamp/callpickup/BLF for the buddy contact. I’d like to get that working too, but recognise that rls.so does not handle dialog messages by default, so I’m starting with Presence... Paul

3 2

Important problem with this test : if ($var(i:maxRangeInx)==$null)
by Ginhoux, Patrick 27 Mar '17

27 Mar '17

Hi, I'm continue my job to 'migrate' my 3.3.1 kamailio.cfg script on a RHEL 6.2 server to the 5.0.x kamailio version on a RHEL 7.1 server, and I encounter a major problem with the '== $null' test in the script. My original script (3.3.1) contains this sequence : $var(i:rangeInx)=0; $var(i:maxRangeInx) = $sht(mbxrangeHash=>"maxrange"); xlog(" imbxRangeInx value before test : '$var(i:maxRangeInx)' "); if ($var(i:maxRangeInx)==$null) { $var(i:maxRangeInx)=99; } xlog(" imbxRangeInx after test : '$var(i:maxRangeInx)' "); ==> The results are : imbxRangeInx value before test : '0' imbxRangeInx value after test : '99' I run the same script on my new server and I get '0' before and after the test. I attempted another kind of test to try to figure out where the problem is. So I write the simple test (with kamailio 5.0.x) : $var(i:rangeInx)=0; $var(i:maxRangeInx) = $sht(mbxrangeHash=>"maxrange"); $var(toto) = $null; if ($var(toto) == $null) xlog ("toto true"); else xlog ("toto false"); The result I got is always 'toto false'. So it appears to me that the test '== $null' doens' that gives a good result with kamailio 3.3.1, doesn't return the same with kamailio 5.0.x. That is a major issue for me and it let me confused about the possibility to move to kamailio 5.0.x. Have people an idea / explanation, why this issue ? Cordialement Patrick GINHOUX

2 2

Module cnxcc
by m9rt8ii 27 Mar '17

27 Mar '17

Please can anybody give me basic steps how to set up cnxcc module on kamailio 4.4 ? I will be very very grateful. http://kamailio.org/docs/modules/4.4.x/modules/cnxcc.html Thank you in advance -- View this message in context: http://sip-router.1086192.n5.nabble.com/Module-cnxcc-tp157107.html Sent from the Users mailing list archive at Nabble.com.

2 1

Cannot yum download from download.opensuse.org, no mirrors
by Shadowy Man 27 Mar '17

27 Mar '17

This is only happening for one particular block of IP's. I can download from other blocks. I suspect it is being done on the download.opensuse.org end for some reason. It appears to be the entire xx.xx block of which I only use a few addresses from. I will probably have to try contact them about it. What I would like to ask is why Kamailio does not have any mirrors? If there were then this would not be a problem. It would just failover.

2 1

wiresharking TLS problem
by Rex Lin (林昱頡) 27 Mar '17

27 Mar '17

hi all wondering if any of you has the experience of TLS decrypting, SIP client has problem while calling each other, but we are using TLS for communication protocol.....facing problem debugging it..... Best Regards, Rex Lin

3 3

Regex module having trouble with large regex expressions in kamailio 4.4.5
by Tim Balmer 24 Mar '17

24 Mar '17

Hi, I discovered some strange behavior from the regex module yesterday (kamailio version 4.4.5). For example we have the phone number 0033755500381 that we try to match with the following kamailio code: #!define REGEX_FF_NEG_INTERNATIONAL "6" ... if (pcre_match_group("$rU", REGEX_FF_NEG_INTERNATIONAL)) { # Do something } When the content of our regex groups file is like this: [6] ^(\+|00)3(0690|0693|0694|0695|0697|0698|0699|36|37|4160|46|47|51160|511691|511693|511696|511891|511893|511896|513|516091|5160921|516093|516096|516391|516392|516393|516396|516591|516592|516593|516596|516691|5166921|516693|516696|5191|5192|5193|5196|538|54388|543 89|546|5475|5477|548|567117|5672|5677|5679|5692|5694|5696|5698|5699|5795|5796|5797|5799|58299|584|5850|5948|5987|5988|5989|59988|59989|620|630|631|670|706|712|725|728|774|776|8591|8592|8595|8597|8598|8599|8620|8621|8630|8631|8640|8641|8643|8649|8650|8651|8660| 8661|8664|8670|8671|869814|906698|93) The above if statment is not matching, although it should match according to any regex validator (for example https://regex101.com/ ) But when I split the above regex in smaller parts, like this: [6] ^(\+|00)306(90|93|94|95|97|98|99) ^(\+|00)336 ^(\+|00)337 ^(\+|00)34160 ^(\+|00)346 ^(\+|00)347 ^(\+|00)351(160|1691|1693|1696|1891|1893|1896|3|6091|60921|6093|6096|6391|6392|6393|6396|6591|6592|6593|6596|6691|66921|6693|6696|91|92|93|96) ^(\+|00)3538 ^(\+|00)354(388|389|6|75|77|8) ^(\+|00)356(7117|72|77|79|92|94|96|98|99) ^(\+|00)357(95|96|97|99) ^(\+|00)358(299|4|50) ^(\+|00)359(48|87|88|89|988|989) ^(\+|00)3620 ^(\+|00)363(0|1) ^(\+|00)3670 ^(\+|00)3706 ^(\+|00)3712 ^(\+|00)372(5|8) ^(\+|00)377(4|6) ^(\+|00)385(91|92|95|97|98|99) ^(\+|00)386(20|21|30|31|40|41|43|49|50|51|60|61|64|70|71|9814) ^(\+|00)3906698 ^(\+|00)393 Then the exact same regex module call as before does result in a match as expected. So to me it looks like the regex module is having trouble with large regex expressions, I did check the logs but there is nothing strange there, that means also the large expression is correctly loaded according to the logs while in fact it is not working. Checked the regex module documentation but there I cannot find any limits regarding to the size of invididual expressions. Is it me missing something or did I discover a bug in de regex module? Best Regards, Tim Balmer Edutel BV

2 2

Re: [SR-Users] Question on the debug change level command with 5.0.x
by Federico Cabiddu 24 Mar '17

24 Mar '17

No, you don't. The debugger module, among other things, allows to set the debug level per module. Regards, Federico On 23 Mar 2017 6:31 pm, "Ginhoux, Patrick" <patrick.ginhoux(a)fr.unisys.com> wrote: Hi, Thanks you so much; it works fine as expected. Now have I to keep the ‘debugger’ module lines in my config ? Cordialement Patrick GINHOUX *De :* sr-users [mailto:sr-users-bounces@lists.sip-router.org] *De la part de* Federico Cabiddu *Envoyé :* jeudi 23 mars 2017 17:38 *À :* Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> *Objet :* Re: [SR-Users] Question on the debug change level command with 5.0.x Hi, you have to load the cfg_rpc module (http://www.kamailio.org/docs/ modules/5.0.x/modules/cfg_rpc.html). Then you can run kamcmd cfg.set_now_int core debug <LEVEL>. Regards, Federico On Thu, Mar 23, 2017 at 5:24 PM, Ginhoux, Patrick < patrick.ginhoux(a)fr.unisys.com> wrote: Hi, Previously, in my kamailio.cfg 3.3.1, the debug level is set using the ‘debug’ option. Then with the MI_FIFO module it was simple to change debug level on the flight using the simple command ‘kamctl fifo debug <new value>’. Moving to kamailio 5.0.x, there is no more MI_FIFO module. In the kamailio.cfg, I still have the ‘debug’ option set; so I’m looking for the command to use to change the debug level. I look at the debugger module that seems the solution. So I add the following lines in y script: debug=4 loadmodule "debugger.so" modparam("debugger", "log_level", 1) modparam("debugger", "mod_hash_size", 5) modparam("debugger", "mod_level_mode", 1) Then I try to use the following commands : This one retrieves the debug level set by the debug=4) : [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level core 4 This one changed the debug level for the core module [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level core 2 200 ok Same commands can be used for each of the module loaded: [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level tm 4 [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level tm 2 200 ok [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level dispatcher 4 [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level dispatcher 2 200 ok Each command only applies to a specific module. I don’t find a ‘global’ command that I could use in the same manner as the old ‘kamctl fifo debug’ command. Is there something I missed in my config ? Cordialement Patrick GINHOUX _______________________________________________ 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 _______________________________________________ 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

2 1

Question on the debug change level command with 5.0.x
by Ginhoux, Patrick 23 Mar '17

23 Mar '17

Hi, Previously, in my kamailio.cfg 3.3.1, the debug level is set using the 'debug' option. Then with the MI_FIFO module it was simple to change debug level on the flight using the simple command 'kamctl fifo debug <new value>'. Moving to kamailio 5.0.x, there is no more MI_FIFO module. In the kamailio.cfg, I still have the 'debug' option set; so I'm looking for the command to use to change the debug level. I look at the debugger module that seems the solution. So I add the following lines in y script: debug=4 loadmodule "debugger.so" modparam("debugger", "log_level", 1) modparam("debugger", "mod_hash_size", 5) modparam("debugger", "mod_level_mode", 1) Then I try to use the following commands : This one retrieves the debug level set by the debug=4) : [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level core 4 This one changed the debug level for the core module [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level core 2 200 ok Same commands can be used for each of the module loaded: [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level tm 4 [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level tm 2 200 ok [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.get_mod_level dispatcher 4 [root@vm-vse02-siprouter1 kamailio]# kamcmd dbg.set_mod_level dispatcher 2 200 ok Each command only applies to a specific module. I don't find a 'global' command that I could use in the same manner as the old 'kamctl fifo debug' command. Is there something I missed in my config ? Cordialement Patrick GINHOUX

2 2

Presence + RLS + XCAP
by Paul Smith 23 Mar '17

23 Mar '17

Longer version of this got blocked by the mail list earlier today… and I have dug a bit deeper since then… I am trying to SUBSCRIBE to a resource-list (rls.so) but the rls_handle_subscribe() is returning with a “list not found" Mar 23 16:36:11 presence kamailio[12536]: DEBUG: rls [subscribe.c:211]: rls_get_service_list(): No rl document found Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:81]: db_free_columns(): freeing 2 columns Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[0] at 0x7f70b50ebee8 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[1] at 0x7f70b50afb90 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:94]: db_free_columns(): freeing result names at 0x7f70b50eaf98 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:99]: db_free_columns(): freeing result types at 0x7f70b50cc0a0 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: <core> [db_res.c:136]: db_free_result(): freeing result set at 0x7f70b50cc2d0 Mar 23 16:36:11 presence kamailio[12536]: DEBUG: rls [subscribe.c:634]: rls_handle_subscribe(): list not found - searched for uri <sip:org1a-contacts@**MYDOMAIN**> I have tried various permutations of xml docs in the xcap table for the rls-service and resource-list but nothing have not yet managed to get a success from the rls_handle_subscribe() Latest version of data in my mysql database / xcap table: | 3 | org1a-contacts | **MYDOMAIN** | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="team"><entry uri="sip:1002@**MYDOMAIN**"/><entry uri="sip:1001@**MYDOMAIN**"><display-name>bob</display-name></entry><entry uri="sip:1003@**MYDOMAIN**"><display-name>test1</display-name></entry><entry uri="sip:1006@**MYDOMAIN**"><display-name>test1006</display-name></entry><entry uri="sip:1007@**MYDOMAIN**"><display-name>test1007</display-name></entry><entry uri="sip:1008@**MYDOMAIN**"><display-name>test1008</display-name></entry><entry uri="sip:1009@**MYDOMAIN**"><display-name>test1009</display-name></entry> <entry uri="sip:1005@**MYDOMAIN**"><display-name>testing1005</display-name></entry> <entry uri="sip:org1a@**MYDOMAIN**"><display-name>testingorg1a</display-name></entry> <entry uri="sip:org1c@**MYDOMAIN**"><display-name>testingorg1c</display-name></entry> <entry uri="sip:dnd1002-6@presence.byphone.co.uk"><display-name>1002 dnd</display-name></entry></list></resource-lists> | 4 | sr-1490270411-10960-2 | 0 | /xcap-root/resource-lists/users/sip:org1a-contacts@**MYDOMAIN**/index | 0 | | 4 | org1a-contacts | **MYDOMAIN** | <?xml version="1.0" encoding="UTF-8" standalone="no"?><rls-services xmlns="urn:ietf:params:xml:ns:rls-services"><service uri="sip:org1a-contacts@**MYDOMAIN**"><list name="org1a-contacts"><rl:entry uri="1000_6@**MYDOMAIN**"/><rl:entry uri="org1c@**MYDOMAIN**"/></list><packages><package>presence</package></packages></service></rls-services> | 8 | sr-1490285288-12551-1 | 0 | /xcap-root/rls-services/users/sip:org1a-contacts@**MYDOMAIN**/index | 0 | Is there a reference doc somewhere with an example of a successful SUBSCRIBE and associated XCAP docs for a resource-list? Thanks Paul

1 0

failed to find append_branch
by Anuradha Selvarajan 23 Mar '17

23 Mar '17

Hello, I am a new user to kamailio. I am trying to use the fork.cfg to send sip registration to 2 destination. I get the following error but I have added loadmodule tm.so, why am I still getting this error? Mar 22 15:53:48 emps116 kamailio[14065]: ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command append_branch (params 1) Mar 22 15:53:48 emps116 kamailio[14065]: : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/fork.cfg, line 45, column 43: unknown command, mi Mar 22 15:53:48 emps116 kamailio[14065]: ERROR: bad config file (1 errors) Mar 22 15:53:48 emps116 kamailio[14065]: loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules Mar 22 15:53:48 emps116 kamailio[14065]: loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Mar 22 15:53:48 emps116 kamailio[14065]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 22 15:53:48 emps116 systemd[1]: kamailio.service: Control process exited, code=exited status=255 Mar 22 15:53:48 emps116 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. My script looks like this (default one): mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules" loadmodule "sl.so" loadmodule "tm.so" . . . . route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); exit; }; # try these two destinations first in parallel; the second # destination is targeted to sink port -- that will make ser # wait until timer hits seturi("sip:nobody@kamailio.org"); append_branch("sip:parallel@kamailio.org"); # if we do not get a positive reply, continue at reply_route[1] #t_on_failure("1"); # forward the request to all destinations in destination set now t_relay(); } Appreciate any help! Thanks Anu

3 3

Presence + RLS + XCAP
by Paul Smith 23 Mar '17

23 Mar '17

Hi I am struggling to figure out how to build and test an integrated presence server with Resource-List (rls.so) and XCAP (xcap_server.so). I am confused about what is meant to happen and how to debug it. Not sure if I am suffering from incompatibilities, configuration errors, or bugs… I have not found an up to date guide on configuring an integrated presence, rls, xcap server so I have used old tutorials, module documentation and some course code review … but I’m pretty confused now! I am using Snom phones and Jitsi (on Mac) to test. I have a kamailio Registrar which proxies all presence messages to the separate kamailio 4.2 Presence server. REQUIREMENT : Contact Presence ============================= I would like to get a SUBSCRIBE from the Snom phone to subscribe to a contact-list resource list, and to receive a NOTIFY with all of the presence states for those contacts. Relevant snom settings? Contact List URI: this generates a SUBSCRIBE with event:presence I believe that: ========== presence module should be managing the individual user presence and dialog states xcap module should allow me to use HTTP / curl to edit the resource lists rls module should be managing the mapping from a resource list subscription to the individual presence states and generating the aggregated NOTIFY messages. each device should then be able to SUBSCRIBE to a resource list uri and then receive an aggregated NOTIFY for all of the watched entities. Current status: =========== SUBSCRIBE sip:org1a-contacts@*****.co.uk <sip:org1a-contacts@*****.co.uk> SIP/2.0 v: SIP/2.0/UDP 192.168.1.15:59594;branch=z9hG4bK-qllzlgs3xf43;rport f: <sip:org1a@*****.co.uk <sip:org1a@*****.co.uk>>;tag=wwasqc6lmb t: <sip:org1a-contacts@****.co.uk <sip:org1a-contacts@****.co.uk>> i: 313439303236383530323234313530-x3hlb3j6b5ef CSeq: 1 SUBSCRIBE Max-Forwards: 70 User-Agent: snomD375/8.9.3.46 m: <sip:org1a@192.168.1.15:59594 <sip:org1a@192.168.1.15:59594>>;reg-id=1 Event: presence Accept: application/pidf+xml, application/rlmi+xml, multipart/related Supported: eventlist Expires: 3600 l: 0 … auth handshake SIP/2.0 202 OK Record-Route: <sip:10.210.160.13;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes <sip:10.210.160.13;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes>> Record-Route: <sip:**pubip**;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes <sip:**pubip**;r2=on;lr=on;ftag=wwasqc6lmb;nat=yes>> v: SIP/2.0/UDP 192.168.1.15:59594;received=**recvip**;branch=z9hG4bK-8wufiv1ua3ne;rport=60554 f: <sip:org1a@****.co.uk <sip:org1a@****.co.uk>>;tag=wwasqc6lmb t: <sip:org1a-contacts@*****.co.uk <sip:org1a-contacts@*****.co.uk>>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-d30e i: 313439303236383530323234313530-x3hlb3j6b5ef CSeq: 2 SUBSCRIBE Expires: 3600 Contact: <sip:10.210.160.16:5060;transport=udp <sip:10.210.160.16:5060;transport=udp>> Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 … and then the snom receives an empty NOTIFY. NOTIFY sip:org1a@192.168.1.15:58325 <sip:org1a@192.168.1.15:58325> SIP/2.0 Via: SIP/2.0/UDP 109.104.108.18;branch=z9hG4bKc54a.20db45097fb44595ccd5a30827ca2a62.0 Via: SIP/2.0/UDP 10.210.160.16;rport=5060;branch=z9hG4bKc54a.40150886000000000000000000000000.0 To: <sip:org1a@****.co.uk <sip:org1a@****.co.uk>>;tag=sx4ne9jpcl From: <sip:org1a-contacts@*****.co.uk <sip:org1a-contacts@*****.co.uk>>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-ed89 CSeq: 2 NOTIFY Call-ID: 313439303237323938323235353133-2du221e6okvf Content-Length: 0 User-Agent: kamailio (4.2.0 (x86_64/linux)) Max-Forwards: 69 Event: presence Contact: <sip:10.210.160.16:5060;transport=udp;alias=10.210.160.16~5060~1 <sip:10.210.160.16:5060;transport=udp;alias=10.210.160.16~5060~1>> Subscription-State: active;expires=3600 The Subscription is showing up in the watchers table, not the rls_watchers table, so I believe it is being rejected by the rls_handle_subscribe() and then parsed by the presence module. Is that correct? There is noting in rls_watchers or rls_presentity tables. Should there be? I have a rows in the XCAP table for the Resource-List curl -d @/tmp/contacts-resource-list.xml -vvv -X PUT http://109.104.108.23:5060/xcap-root/resource-lists/users/sip:org1a-contact… <http://109.104.108.23:5060/xcap-root/resource-lists/users/sip:org1a-contact…> relevant row in xcap table: | 3 | org1a-contacts | *****.co.uk <http://co.uk/> | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="team"><entry uri="sip:1002@****.co.uk <http://co.uk/>"/><entry uri="sip:1001@****.co.uk <http://co.uk/>"><display-name>bob</display-name></entry><entry uri="sip:1003@****.co.uk <http://co.uk/>"><display-name>test1</display-name></entry><entry uri="sip:1006@****co.uk <http://co.uk/>"><display-name>test1006</display-name></entry><entry uri="sip:1007@****co.uk <http://co.uk/>"><display-name>test1007</display-name></entry><entry uri="sip:1008@h****.co.uk <http://co.uk/>"><display-name>test1008</display-name></entry><entry uri="sip:1009@****.co.uk <http://co.uk/>"><display-name>test1009</display-name></entry> <entry uri="sip:1005@****.co.uk <http://co.uk/>"><display-name>testing1005</display-name></entry> <entry uri=“sip:org1a@****.co.uk <http://co.uk/>"><display-name>testingorg1a</display-name></entry> <entry uri=“sip:org1c@****.co.uk <http://co.uk/>"><display-name>testingorg1c</display-name></entry></list></resource-lists> | 4 | sr-1490270411-10965-2 | 0 | /xcap-root/resource-lists/users/sip:org1a-contacts@****co.uk/index <http://co.uk/index> | 0 | I have rows in the presence table for the entities that I want to track e.g. | | | 2889 | org1c | ****.co.uk <http://co.uk/> | presence | a.1490270411.10942.42.1083 | 1490277099 | 1490273499 | <?xml version="1.0 " encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:im="urn:ietf:params:xml:ns:pidf:im" entity=“pres:org1c@*****.co.uk <http://co.uk/>"> <tuple id="snom370-000413260935"> <status><basic>open</basic> <im:im <im:im>>Available</im:im> </status> <contact priority="1.00">sip:org1c@****.co.uk <http://co.uk/></contact> <note xml:lang="en">Available</note> </tuple></presence> I am running kamaiio 4.2 Question : Should I have an resource-services row too? Or does rls.so infer the relationship using the URI? When I use jitsi to manipulate its contact list it only inserts 2 rows into the xcap table and I cannot get a SIP Subscribe to connect to that resource list either. Question : What should the curl PUT commands and XML docs look like to build a valid RLS service (is there a good example anywhere)? Question : Is there an example config or doc anywhere showing how to build integrated presence, rls, xcap? Question : Has anyone got this working with Snom and or other handsets… the standards and implementations seem vague / inconsistent? Note: There is another Snom setting (Extension Monitoring Call Pickup List URI:) which produces a SUBSCRIBE with Event:dialog but which should allow a button/lamp/callpickup/BLF for the buddy contact. I’d like to get that working too, but recognise that rls.so does not handle dialog messages by default, so I’m starting with Presence... Paul Paul Smith 1 Lanyon Quay, Belfast, BT1 3LG UK: 02890270050 RoI: 1800855111 paul.smith(a)claritytele.com <mailto:paul.smith@claritytele.com%0b> www.claritytele.com <http://claritytele.com/> This email and any attachment, is confidential. If you have received this in error, please delete it from your system, do not use or disclose the information in any way, and notify me immediately. The contents of this message may contain personal views which are not the views of Clarity Telecom, unless specifically stated. > >> >>

1 0

Status of db_cassandra
by Markus Bönke 23 Mar '17

23 Mar '17

Hello, We are thinking about to use kamailio as a sip registrar with cassandra as db backend. As I can see in the documentation for kamailio 5.0 it is only tested with Casandra 1.1.6 and 1.0.1, current version is 3.10 in the meantime. Is anyone using this module with newer versions of Cassandra? Is it stable? Thanks and regards Markus

4 6

looks like dp_translate with pvs in repl_exp is badly broken
by Juha Heinanen 23 Mar '17

23 Mar '17

when match_exp of dp 100 is ^#test and repl_exp is sip:$rU@$rd this code: xlog("L_INFO", "****** ru = $ru\n"); dp_translate(100, "$rU/$var(res)"); xlog("L_INFO", "****** res = $var(res)\n"); produces: Mar 22 08:21:17 lohi /usr/bin/sip-proxy[7543]: INFO: ****** ru = sip:#test@test.tutpro.com Mar 22 08:21:17 lohi /usr/bin/sip-proxy[7543]: INFO: ****** res = sip:$rU@$rd -- juha

4 6

getting source IP from Contact header
by Ali Taher 22 Mar '17

22 Mar '17

Hello everyone, I'm trying to get the host part from the contact header in the sip invite packet. Following is the contact header format: Contact: <sip:user_name@192.168.1.1:5060> What I want as result is 192.168.1.1 I searched among the pseudovairables and found $ct variable which reference to the body of contact header. (<sip:user_name@192.168.1.1:5060>) Is there a way to get directly the host (192.168.1.1) without applying transformation at the level of $ct. If not what transformation can achieve it. Thanks in advance. Regards, Ali

3 2

Alternative of SIPp for WebRTC2SIP load-test
by Jade SZ 21 Mar '17

21 Mar '17

Hi Guys, SIPp is great in testing pure SIP udp/tcp/tls but I was wondering if there is a similar tool for stress testing Kamailio websocket2sip. I am trying to load-test webrtc2sip kamailio's implementation closest to the real scenario. One way would be to fork multiple process of webrtc2sip client in loop. But would really like to hear how are kamailio webrtc users/community load-testing it for benchmarking their servers. Also roughly how much CPS they have achieved using single kamailio with webrtc enabled (share server specs as well) Thanks in advance. Regards, JSZ

2 1

do we need high available IP for S-CSCF?
by Valentin Christoph 21 Mar '17

21 Mar '17

Hi all, Maybe I need a small hint. I have got a configuration of 2 S-CSCF instances (both kamailio 4.4.5 with S-CSCF configuration, each on a separate virtual machine) in load sharing mode (50:50). Both instances work on the same "scscf" data base, so that it is only one S-CSCF from a logical point of view. Now I get into troubles with registration: * P-CSCF sends REGISTER to first S-CSCF * First S-CSCF answers with 401 Unauthorized an challenges the UE. So far everything OK. * Second REGISTER is sent to second S-CSCF (DNS SRV Load Balancing). * Second S-CSCF answers again with 401 Unauthorized (Error) * This goes back and forth until UE gives up to register This problem exists sporadically (approx. every fifth registration). Now the question: is it worth to debug this configuration or should we rather switch to high available IP configuration (if we want to use kamailio)? What's your opinion/experiences? Thanks Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

Siremis 4.3.0 with Apache 2.4
by kai.ohnacker＠cbc.de 21 Mar '17

21 Mar '17

Hello Community, I installed the Kamailio 4.4 and try to install the Siremis GUI based on apache 2.4. If I try to start the Siremis Wizard, the Webserver display the following line: "location.href='$url'"; exit; } ?>". The starting page of apache 2.4 is working fine, but the /siremis page doesn't. Due the installation I take the dedicated apache 2.4 commands. Are there some further requirements? Some ideas? OS: Ubuntu 16.04 Thanks a lot. Cheers, Kai

2 2

No cdr after Kamailio 5.0 restart
by Kordován Szabolcs 21 Mar '17

21 Mar '17

Hi, I use acc module for create cdr into db and also make acc log into db. After restart Kamailio cdr won't be created but acc log will be inserted into db. In the syslog also I see just the acc log. The dialog will be remove from db. I have read many threads. I know that it should work. Thanks for help! Regards, Szabolcs >From my configs: #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 modparam("acc", "db_url", DBURL) modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") modparam("acc", "log_level", 3) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "early_media", 1) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "acc_prepare_always", 1) modparam("acc", "time_mode", 1) modparam("acc", "time_attr", "seconds") modparam("acc", "time_exten", "microsecs") modparam("acc", "reason_from_hf", 1) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdr_extra", "callid=$ci;src_user=$dlg_var(fU);src_domain=$dlg_var(fd);dst_ouser=$dlg_var(tU);dst_user=$dlg_var(rU);dst_domain=$dlg_var(rd);src_ip=$dlg_var(si);peer_in=$dlg_var(peerin);peer_out=$dlg_var(peerout)") request_route { <------> <------>$var(loose) = loose_route(); <------>$var(check_trans) = t_check_trans(); <------>sip_trace(); <------>setflag(FLT_SIPTRACE); <------>dlg_manage(); <------># handle retransmissions <------>if(t_precheck_trans()) { <------> xlogl("LOG_LOCAL0", "L_ALERT", "================== kamaty $rm route $ci ==== $rm ==== precheck_trans exit\n"); <------> exit; <------>} <------>if(is_method("OPTIONS")) {. <------> # send reply for each options request. <------> sl_send_reply("200", "ok");. <------> exit();. <------>} <------># CANCEL processing <------>if (is_method("CANCEL")) { <------><------>if (t_check_trans()) { <------><------> route(RELAY); <------><------>} <------><------>exit; <------>} <------>if ( is_method("ACK") ) { <------><------>if ( $var(check_trans) ) { <------><------> route(RELAY); <------><------>} <------><------>exit; <------>} <------>if (is_method("BYE")) { <------> if ($var(loose)) { <------><------>setflag(FLT_ACC); # do accounting ... <------><------>setflag(FLT_ACCFAILED); # ... even if the transaction fails <------><------>setflag(FLT_ACCMISSED); <------><------>route(RELAY); <------> <-->exit; <------> } <------> exit; <------>} etc.

5 10

Re: [SR-Users] ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 21 Mar '17

21 Mar '17

Hi, There was a mistake in the file attached ; the short kamailio.cfg file (that makes kamailio to start) was not the correct one; attached the correct one. Have anyone an idea of this problem ? Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : lundi 20 mars 2017 17:58 À : Kamailio users (sr-users(a)lists.sip-router.org) <sr-users(a)lists.sip-router.org> Objet : RE: ht_db_load_table(): key type must be string (type=6) Hi, To figure out what causes this issue, I work to shorten my kamailio.cfg to contains loadmodule/modparam instructions for the htable module only. My first attempt (see kamailio.cfg.ok) shows that the problem doesn't occur. In the second attempt (see kamailio.cfg.ko) I add instructions previously removed from the original .cfg file. I have determined that, depending on the instructions that follows the "modparam("htable", "fetch_rows", 1000)" line, the "key type must be string (type=6)" error occurs or not. In this ko file, if I comment out the following sequence, the error occurs : modparam("htable", "fetch_rows", 1000) #!!ifdef FIXED_ROUTING #modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") #!!else #modparam("htable", "htable", "abpqHash=>size=12;dbtable=abpq;") #modparam("htable", "htable", "zabpqHash=>size=12;dbtable=zabpq;") #!!endif modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") I hope my explanation is clearn and there are people who can help me on this problem that causes kamailio to not start. Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : vendredi 17 mars 2017 15:46 À : Kamailio users (sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>) <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Objet : ht_db_load_table(): key type must be string (type=6) Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX

1 0

Re: [SR-Users] ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 20 Mar '17

20 Mar '17

Hi, To figure out what causes this issue, I work to shorten my kamailio.cfg to contains loadmodule/modparam instructions for the htable module only. My first attempt (see kamailio.cfg.ok) shows that the problem doesn't occur. In the second attempt (see kamailio.cfg.ko) I add instructions previously removed from the original .cfg file. I have determined that, depending on the instructions that follows the "modparam("htable", "fetch_rows", 1000)" line, the "key type must be string (type=6)" error occurs or not. In this ko file, if I comment out the following sequence, the error occurs : modparam("htable", "fetch_rows", 1000) #!!ifdef FIXED_ROUTING #modparam("htable", "htable", "mbxrangeHash=>size=4;dbtable=mbxrange;") #!!else #modparam("htable", "htable", "abpqHash=>size=12;dbtable=abpq;") #modparam("htable", "htable", "zabpqHash=>size=12;dbtable=zabpq;") #!!endif modparam("htable", "htable", "propertiesHash=>size=4;dbtable=properties;") I hope my explanation is clearn and there are people who can help me on this problem that causes kamailio to not start. Cordialement Patrick GINHOUX De : Ginhoux, Patrick Envoyé : vendredi 17 mars 2017 15:46 À : Kamailio users (sr-users(a)lists.sip-router.org) <sr-users(a)lists.sip-router.org> Objet : ht_db_load_table(): key type must be string (type=6) Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX

1 0

ACC - Cdr enabled
by Diego Nadares 20 Mar '17

20 Mar '17

Hi guys, I have configured to use cdr with extra fields modparam("acc", "cdr_extra", * "sip_code=$rs;sip_reason=$rr;"* "hangup_src=$dlg_var(hangup_disposition);" "prefix=$dlg_var(prefix);gos=$dlg_var(gos);endpoint_type=$ dlg_var(endpoint_type);" "ip_a=$dlg_var(ip_a);ip_b=$dlg_var(ip_b);" Is this the right way to always save the hangup cause? Or is another better way? Because, when call is finished with a bye both fields are empty. The other thing is that I need to save, start_time and end_time with seconds.microseconds (1490013250.601707). Is this possible? Thanks! Diego

2 5

Is File Sharing between two Extensions Possible?
by Shahid Mehmood 20 Mar '17

20 Mar '17

Dear all, I want to share (any kind of) files between two extensions. Is it possible? If it is, then how ??? please elaborate with example. -- Shahid Mehmood

1 0

cisco spa502g issues
by ben＠wtf.com.au 19 Mar '17

19 Mar '17

Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@<our domain> Contact:: sip:5546@<OUR OFFICE IP>:1032 Q= Expires:: 180 Callid:: 4754c4f9-c67e1018(a)10.0.41.29 Cseq:: 43112 User-agent:: Cisco/SPA502G-7.6.2a State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:<OUR SR IP>:5060 Methods:: 6815 AOR:: 5590(a)sip.skunkworks.net.au Contact:: sip:5590@10.0.41.14:5060 Q= Expires:: 1154 Callid:: 24435738-224b06db(a)10.0.41.14 Cseq:: 8012 User-agent:: Linksys/SPA921-5.1.8 Received:: sip:<OUR OFFICE IP>:1026 State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: udp:<OUR SR IP>:5060 Methods:: 4767 The older 921 has its private IP in the contact, where as the newer 502 has the external IP of our office in the contact. Our file wall is a Watchguard T-10 (latest updates etc) with the SIP-ALG running. Any thoughts on where to start looking ? Cheers, Ben.

3 4

Generate extra cdr per dialog
by Diego Nadares 18 Mar '17

18 Mar '17

Hi Guys, I'm using acc module for accounting with the following conf: .... modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_facility", "LOG_LOCAL1") modparam("acc", "cdr_end_id", "end_time_dt") modparam("acc", "cdr_extra", "sip_code=$rs;sip_reason=$rr;" "hangup_src=$dlg_var(hangup_disposition);" "prefix=$dlg_var(prefix);gos=$dlg_var(gos);endpoint_type=$dlg_var(endpoint_type);" "ip_a=$dlg_var(ip_a);ip_b=$dlg_var(ip_b);" .... It works fine. At the end of the call the cdr is generated ok. BUT In MANAGE_FAILURE, If the first call failed and if there are more dests, we are generating a new call in the same dialog (Some of the dialog vars like ip_a, ip_b can change for the new dest ) I would like to generate both cdrs, the one that failed and the new one (with some dlg vars with new data). if(rtjson_next_route() && t_check_status("503")) { $avp(current_route) = $avp(current_route)+1; t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); exit; } Right now just the last call cdr is saved. I all ready made this work with db_acc (transaction oriented) but I think in my case it will be more reliable with cdr (dialog oriented). Thanks in advance! Diego.

1 1

f_malloc WARNINGS in syslog
by Diego Nadares 17 Mar '17

17 Mar '17

Hi Guys, I'm seeing a lot of warnings in my syslog. Is this normal? Or maybe I am doing something wrong in my cfg? .... Mar 16 17:11:47 dwsipm03 ./kamailio[26226]: WARNING: <core> [mem/f_malloc.c:444]: fm_malloc(): fm_malloc(0x7f68e5199010, 8) called from nathelper: nathelper.c: nh_timer(2078) Mar 16 17:11:47 dwsipm03 ./kamailio[26226]: WARNING: <core> [mem/f_malloc.c:527]: fm_malloc(): fm_malloc(0x7f68e5199010, 8) returns address 0x7f68e52a2d50 Mar 16 17:11:47 dwsipm03 ./kamailio[26226]: WARNING: <core> [mem/f_malloc.c:588]: fm_free(): fm_free(0x7f68e5199010, 0x7f68e52a2d50), called from nathelper: nathelper.c: nh_timer(2222) Mar 16 17:11:47 dwsipm03 ./kamailio[26226]: WARNING: <core> [mem/f_malloc.c:607]: fm_free(): fm_free: freeing block alloc'ed from nathelper: nathelper.c: nh_timer(2078) ..... Mar 16 17:15:44 dwsipm03 ./kamailio[26210]: WARNING: <core> [mem/f_malloc.c:607]: fm_free(): fm_free: freeing block alloc'ed from core: xavp.c: xavp_new_value(94) Mar 16 17:15:44 dwsipm03 ./kamailio[26210]: WARNING: <core> [mem/f_malloc.c:588]: fm_free(): fm_free(0x7f68d9d51000, 0x7f68da069ca8), called from tm: h_table.c: free_cell_helper(246) Mar 16 17:15:44 dwsipm03 ./kamailio[26210]: WARNING: <core> [mem/f_malloc.c:607]: fm_free(): fm_free: freeing block alloc'ed from tm: h_table.c: build_cell(317) Mar 16 17:15:44 dwsipm03 ./kamailio[26210]: WARNING: <core> [mem/f_malloc.c:588]: fm_free(): fm_free(0x7f68d9d51000, 0x7f68da031f00), called from tm: h_table.c: free_cell_helper(150) .... Thanks in advance. Diego

2 2

ht_db_load_table(): key type must be string (type=6)
by Ginhoux, Patrick 17 Mar '17

17 Mar '17

Hi, Now the kamailio installation is completed, I try to start kamailio but got an error on a modparam line. By looking at the kamailio.cfg (with debug mode) it looks like the instruction is not correct : modparam("htable", "fetch_rows", 1000) [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:38:13 CET; 17s ago Process: 1526 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [pike.c:99]: pike_init(): PIKE - initializing Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: cfgutils [cfgutils.c:767]: mod_init(): no hash_file given, disable hash functionality Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: htable [ht_db.c:276]: ht_db_load_table(): key type must be string (type=6) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: ERROR: <core> [core/sr_module.c:968]: init_mod(): Error while initializing module htable (/usr/lib64/kamailio/modules/htable.so) Mar 17 15:38:13 vm-vse02-siprouter1 /usr/sbin/kamailio[1539]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:38:13 vm-vse02-siprouter1 kamailio[1526]: Starting kamailio: Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:38:13 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# modparam("htable", "fetch_rows", "1000") [root@vm-vse02-siprouter1 ~]# systemctl status kamailio.service -l kamailio.service - SYSV: Kamailio is a fast, reliable and flexible SIP Server. Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Fri 2017-03-17 15:26:21 CET; 3s ago Process: 1452 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Mar 17 15:26:20 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: Not starting : invalid configuration file! Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <fetch_rows> of type <1> not found in module <htable> Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 308, column 41: Can't set module parameter Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: ERROR: bad config file (1 errors) Mar 17 15:26:21 vm-vse02-siprouter1 kamailio[1452]: 0(1460) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 17 15:26:21 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. [root@vm-vse02-siprouter1 ~]# The documentation shows that fetch_rows is an integer : fetch_rows (integer) How many rows to fetch at once from database. Default value is 100. Example 1.12. Set fetch_rows parameter ... modparam("htable", "fetch_rows", 1000) So how to explain the first error. Have you any thoughts on this case ? Cordialement Patrick GINHOUX

1 0

Error on <ds_probing_threshhold> of type <2> not found in module <dispatcher> and SCTP API Not initialized
by Ginhoux, Patrick 17 Mar '17

17 Mar '17

Hi, I'm migrating my kamailio.cfg from the 3.3.1 release to 5.0.0 and encounter the following errors : Mar 16 12:56:45 vm-vse02-siprouter1 systemd[1]: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: Not starting : invalid configuration file! Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: 0(6652) ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <ds_probing_threshhold> of type <2> not found in module <dispatcher> Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: 0(6652) CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 295, column 49: Can't set module parameter Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: 0(6652) WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 971, column 6-13: constant value in if(...) Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: ERROR: bad config file (1 errors) Mar 16 12:56:45 vm-vse02-siprouter1 kamailio[6644]: 0(6652) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 16 12:56:45 vm-vse02-siprouter1 systemd[1]: kamailio.service: control process exited, code=exited status=1 Mar 16 12:56:45 vm-vse02-siprouter1 systemd[1]: Failed to start SYSV: Kamailio is a fast, reliable and flexible SIP Server.. Mar 16 12:56:45 vm-vse02-siprouter1 systemd[1]: Unit kamailio.service entered failed state. Below the extract of my kamailio.cfg file : modparam("dispatcher", "list_file", DISPATCHER_FILE) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshhold",3) modparam("dispatcher", "flags", 2) modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "ds_hash_size", 8) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "force_dst", 1) For the dispatcher error : I don't find any change (eg in the Changelog for 5.0.0.) regarding the dispatcher module. So is it a bug in the dispatcher module or have I to change it to something else ? For the SCTP API error : I don't know what to do here. Thanks to people that can help me on these 2 errors Cordialement Patrick GINHOUX

2 3

Configuration to fetch SIP response body.
by vivek＠advaitamtech.com 17 Mar '17

17 Mar '17

Hi, Is there any configuration to fetch SIP response body received?. My requirement is to fetch the SIP response body and save it into database. Thanks, Vivek

3 3

Search for kamailio PERL module for 5.0 level
by Ginhoux, Patrick 17 Mar '17

17 Mar '17

Hi, I'm a beginner in the use of kamailio product, so I try below to explain my problem with my words: 1) PERL module issue : Some years ago a SIP Router proxy running RedHat 6.2 was developped using the kamailio version 3.3 and the following packages used : kamailio-3.3.2-1.1.1.x86_64.rpm kamailio-snmpstats-3.3.2-1.1.1.x86_64.rpm kamailio-mysql-3.3.2-1.1.1.x86_64.rpm kamailio-perl-3.3.2-1.1.1.x86_64.rpm We have the project to migrate this SIP Proxy to RedHat 7.1 and used the latest Kamailio v5.0.0 Release: kamailio-5.0.0-24.1.x86_64.rpm kamailio-mysql-5.0.0-24.1.x86_64.rpm kamailio-snmpstats-5.0.0-24.1.x86_64.rpm However I don't find the kamalio-perl package for RedHat 7.1. I see that the perl module has been renamed to 'app_perl' but I don't find it either for kamailio 5.0.0. 2) Kamailio.cfg script Our proxy used the kamailio.cfg from the sample of the 3.3.3 release. So can I reuse the same 3.3.3 kamailio.cfg file with the 5.0.0 release or should I migrate (rewrite it) from the 5.0.0 sample ? So I'm looking for some help on these issues. Is there people that had this case ? Thanks in advance for any answer. Cordialement Patrick GINHOUX PS : I just create my account today but don't get the email to validate it.

2 6

(no subject)
by Alberto Llamas 17 Mar '17

17 Mar '17

Hi Guys, Who knows and can recommend a company a company where I could order Toll free numbers for EU?. I really appreciate any hint and sorry for the noise. Regards, -- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA *"Internet is all about share"*

2 1

cisco spa502g issues
by Ben Kramer 17 Mar '17

17 Mar '17

Hi guys, we have a bunch of SIP phones behind a fire wall, with our kamalio server out on the internet. Most of them are the older SPA92x series, but we have some new SPA502g's. We have no problems calling between 92x and 502's. How ever the 502's calling each other do not get voice path. I have noticed that the phones REGISTER differently: AOR:: 5546@<our domain> Contact:: sip:5546@<OUR OFFICE IP>:1032 Q= Expires:: 180 Callid:: 4754c4f9-c67e1018(a)10.0.41.29 Cseq:: 43112 User-agent:: Cisco/SPA502G-7.6.2a State:: CS_DIRTY Flags:: 0 Cflag:: 0 Socket:: udp:<OUR SR IP>:5060 Methods:: 6815 AOR:: 5590(a)sip.skunkworks.net.au Contact:: sip:5590@10.0.41.14:5060 Q= Expires:: 1154 Callid:: 24435738-224b06db(a)10.0.41.14 Cseq:: 8012 User-agent:: Linksys/SPA921-5.1.8 Received:: sip:<OUR OFFICE IP>:1026 State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: udp:<OUR SR IP>:5060 Methods:: 4767 The older 921 has its private IP in the contact, where as the newer 502 has the external IP of our office in the contact. Our file wall is a Watchguard T-10 (latest updates etc) with the SIP-ALG running. Any thoughts on where to start looking ? Cheers, Ben.

1 0

send_all blocks indefinitely
by Robert 16 Mar '17

16 Mar '17

H all, Recently, when I'm trying to load test Kamailio with a single proxy, it hung at around 15000 TPS. I attached gdb to one of the kamailio processes and I saw this: (gdb) where #0 0x00007fa8872366ad in __libc_send (fd=13, buf=0x7ffe6e0e5350, n=16, flags=-2027723091) at ../sysdeps/unix/sysv/linux/x86_64/send.c:27 #1 0x00000000004eb635 in send_all (socket=13, data=0x7ffe6e0e5350, data_len=16) at pass_fd.c:133 #2 0x00007fa4155a1b69 in tcp_close_connection (conn_id=28022) at tcp_funcs.c:199 #3 0x00007fa4155a363f in cleanup (start=0x7fa4540c3ab8, end=0x7fa42f92dca0) at tcp_funcs.c:337 #4 0x00007fa4155a1496 in timer_clean_tcp_connections (ticks=18583555, param=0x0) at tcp_funcs.c:165 #5 0x0000000000617f79 in fork_basic_timer (child_id=-1, desc=0x7fa4155a88e0 "Connections cleanup timer", make_sock=1, f=0x7fa4155a0ac6 <timer_clean_tcp_connections>, param=0x0, interval=30) at timer_proc.c:79 #6 0x00007fa4155a5816 in child_init (rank=0) at tcp_mod.c:176 tcp_func.c is something I wrote myself, which is basically a timer that runs periodically to try to clean up tcp connections that I no longer need. The code for tcp_close_connection is taken directly from usrloc > urecord.c > close_connection() >From the gdb output above, it seems like send_all is blocked and never returned, which led to my proxy to hang. In kamailio/src/core/pass_fd.c, I saw this: ... /** sends all data (takes care of signals) (assumes blocking fd) * returns number of bytes sent or < 0 for an error */ int send_all(int socket, void* data, int data_len) But in kamailio/doc/misc/TODO, I saw this: ... x [tcp] make send_all, send non-blocking ready ? x [tcp] tcp_send w/ timeout (to avoid waiting forever to send DOS) So, my question is, is send_all() suppose to block or not? I'm fairly new to how tcp and socket connection works. Any help will be appreciated! Thanks, Robert -- View this message in context: http://sip-router.1086192.n5.nabble.com/send-all-blocks-indefinitely-tp1569… Sent from the Users mailing list archive at Nabble.com.

1 0

Cdr on shutdown
by Diego Nadares 16 Mar '17

16 Mar '17

Hi Guys, Is it possible to generate cdrs or to save dialogs to db when kamailio receives a kill signal from terminal? Thanks in advance. Diego

3 5

siptrace duplicate both incoming and outgoing messages
by Grant Bagdasarian 16 Mar '17

16 Mar '17

Hi, Is it possible to instruct Kamailio to duplicate both incoming and outgoing messages automatically? If not, if I set the flag and explicitly call sip_trace, will both messages be duplicated to the capture server? Regards, Grant Bagdasarian CM

2 1

mi_fifo: No module matching <jsonrpcs> found
by Mostafa Zare 15 Mar '17

15 Mar '17

hi guys i'm newbie to kamailio. i've installed kamailio version 5 (and all the modules) from pkgs on a jessie debian. i have the below config on my kamailio.cfg when i try to start kamailio service it throws the following error. what's the problem? - systemctl status kamailio: kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled) Active: failed (Result: exit-code) since Wed 2017-03-15 19:01:17 IRST; 10s ago Process: 9377 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255) Main PID: 1576 (code=exited, status=0/SUCCESS) Mar 15 19:01:17 kamailio-debian kamailio[9377]: ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <jsonrpcs> found Mar 15 19:01:17 kamailio-debian kamailio[9377]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 103, column 72: Can't set module parameter Mar 15 19:01:17 kamailio-debian kamailio[9377]: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <ds_probing_threshhold> of type <2> not found in module <dispatcher> Mar 15 19:01:17 kamailio-debian kamailio[9377]: CRITICAL: <core> [core/cfg.y:3404]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 135, column 50: Can't set module parameter Mar 15 19:01:17 kamailio-debian kamailio[9377]: ERROR: bad config file (4 errors) Mar 15 19:01:17 kamailio-debian kamailio[9377]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Mar 15 19:01:17 kamailio-debian kamailio[9377]: loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Mar 15 19:01:17 kamailio-debian systemd[1]: kamailio.service: control process exited, code=exited status=255 Mar 15 19:01:17 kamailio-debian systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Mar 15 19:01:17 kamailio-debian systemd[1]: Unit kamailio.service entered failed state. - kamailio.cfg: # *** To run in debug mode: # - define WITH_DEBUG # #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 # log_facility=LOG_LOCAL0 # Ds increasing debug... debug=2 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:0.0.0.0:5060 # listen=127.0.0.1:5060 sip_warning=no ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- # modparam("dispatcher", "db_url", DBURL) # modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") ####### Routing Logic ######## # main request routing logic route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } # dispatch destinations route(DISPATCH); # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; sl_send_reply("404", "No registrar"); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { # Destination types: # http://kamailio.org/docs/modules/4.2.x/modules/dispatcher.html#dispatcher.f… # 4 = round robin # 9 = weight-based # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "9")) { send_reply("404", "No destination"); exit; } t_set_fr(0,2000); t_on_failure("ROUTE_ADVANCE"); route(RELAY); exit; } event_route[dispatcher:dst-down] { xlog("L_ERR", "Destination down: $rm $ru ($du)\n"); } event_route[dispatcher:dst-up] { xlog("L_ERR", "Destination up: $rm $ru\n"); } failure_route[ROUTE_ADVANCE] { if (t_is_canceled()) { exit; } xlog("L_NOTICE", "Media server $du failed to answer, selecting other one!\n"); if (t_check_status("503") || (t_branch_timeout() and !t_branch_replied())) { ds_mark_dst("ip"); # select the new destination if(ds_next_dst()) { # again set local timeout for reply # xlog("L_NOTICE", "Selecting next route upon failure\n"); t_set_fr(0,2000); t_on_failure("ROUTE_ADVANCE"); route(RELAY); exit; } else { #last available node failed to reply, no other destinations available send_reply("404", "No destination"); xlog("L_NOTICE","No destination, sent 404\n"); exit; } } else { xlog("L_NOTICE","Unhandled failured route!\n"); } }

2 1

SIP contact header question
by Jack Davis 15 Mar '17

15 Mar '17

Greetings, I have a general question about the usage of SIP contact headers in the context of using Kamailio as a SIP proxy. ________________________________ [ A ] --> [ Kamailio B ] ---> [ C ] Node A originates a SIP invite, containing a valid via header and URI while setting the contact address to a user@itself and delivers it to Kamailio B which is acting as a SIP proxy. Kamailio B then uses dispatcher routing to direct the Invite to node C, adding a via line with its own information as well as a record-route header with its own proxy information but retaining the same contact address from A. Node C establishes the call and then sends a re-invite to the Kamailio B proxy which is in turn sent to Node A. Node A responds with a 200 OK The problem arises when Node C tries to send an Ack in response to this 200 OK. The ack is being sent to the Contact address, rather than the routing already established in the initial dialog. ________________________________ My question is: should kamailio be rewriting this contact address with its own? Is that the best practice? My understanding is that the contact header is more so related to future requests within the same dialog ONLY when a record-route is not used. I would appreciate any clarification on the RFC or best practices in this scenario. Thank you, Jack Davis

5 12

Kamailio does'nt respond to packets from specific addresses
by עדי ויזגן 15 Mar '17

15 Mar '17

Hi everyone, I have the following problem: I have a Kamailio server that is connected to several Asterisk servers for load balacing in a LAN network with the IP's of 192.168.16.X. I recently connected with another company that is supposed to send me calls through the Kamailio server. The problem is this: I am connected to then with a cable from their network to mine. My port has the address 10.9.18.198 and they come from the address 172.18.177.18. When I get packets from their address (172...) I can see them getting to my Kamailio server, but the Kamailio does not respond. It only happens when I get packets from 172 source address. If I get from 192 or 10 adresses from my LAN network it works, and if I get packets from real world IP's through my WAN it works. Does anyone have any idea why the Kamailio does'nt respond to packets from 172 adresses? And how to fix this? Thanks in advance!

2 3

Kamailio World 2017 - A selection of presentations
by Daniel-Constantin Mierla 15 Mar '17

15 Mar '17

Hello, I would like to announce that many of the presentations at Kamailio World 2017 (May 8-10, in Berlin, Germany) are now listed at: - https://www.kamailioworld.com/k05/schedule/ Just several are not published yet, working to get all their details in place and expect to have the first draft of the schedule very soon. This edition takes place again at the same location in the famous city center of Berlin, the number of seats is limited, last year the event was fully booked. Do not delay too much your registration if you want to attend this year: - https://www.kamailioworld.com/k05/registration/ Thanks to the sponsors (https://www.kamailioworld.com/k05/sponsors/), it looks like we will be able to offer again some grants for students or underrepresented groups. More details will be sent out in the near future. Looking forward to meeting many of you in Berlin! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Re: [SR-Users] rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hi Daniel, yes, I hear the audio going both ways ok. In the meantime I succeeded to record something by rtpproxy but in best case only one direction, I can hear just the callee, but not the caller. In the meantime I have used start_recording in route[NATMANAGE] but before rtpproxy_manage("co"): -------------------------------------------------- if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); -------------------------------------------------- After reading of your message I tested also with start_recording after rtpproxy_manage("co"): -------------------------------------------------- rtpproxy_manage("co"); if (is_method("INVITE") and (status=="200")) { start_recording(); } -------------------------------------------------- But the result was the same: I have 2 SIP users: 31 (X-Lite on Windows, static address 192.168.0.11) 35 (Android Zoiper, IP address 192.168.0.29 (by DHCP)) Kamailio has static address 192.168.0.13. When 35 calls 31 then I hear 1 direction in the recorded rtp file, the voice of callee. When 31 calls 35 then I don't hear anything in the recorded file. I don't understand why this difference when user 31 or 35 initiates a call!? But I hear audio conversation in both directions, just from the recorded file in one direction, or even nothing. I get 2 files per one call: <call-id>.rtp and <call-id>.rtcp How many files should rtpproxy produce for one call? Should also a file or two for 2nd channel be generated? How can I record both audio ways? Thank you Marko The whole function route[NATMANAGE]: # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } ----- Original Message ----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Marko Tirs <marko.tirs(a)yahoo.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 14, 2017 5:23 PM Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording Hello, I think you can call the start recording function just after rtpproxy_manage(). If you don't use the start recording function, is the audio going both ways without problems? Cheers, Daniel

1 0

Re: [SR-Users] rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hi Daniel, yes, I hear the audio going both ways ok. In the meantime I succeeded to record something by rtpproxy but in best case only one direction, I can hear just the callee, but not the caller. In the meantime I have used start_recording in route[NATMANAGE] but before rtpproxy_manage("co"): -------------------------------------------------- if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); -------------------------------------------------- After reading of your message I tested also with start_recording after rtpproxy_manage("co"): -------------------------------------------------- rtpproxy_manage("co"); if (is_method("INVITE") and (status=="200")) { start_recording(); } -------------------------------------------------- But the result was the same: I have 2 SIP users: 31 (X-Lite on Windows, static address 192.168.0.11) 35 (Android Zoiper, IP address 192.168.0.29 (by DHCP)) Kamailio has static address 192.168.0.13. When 35 calls 31 then I hear 1 direction in the recorded rtp file, the voice of callee. When 31 calls 35 then I don't hear anything in the recorded file. I don't understand why this difference when user 31 or 35 initiates a call!? But I hear audio conversation in both directions, just from the recorded file in one direction, or even nothing. I get 2 files per one call: <call-id>.rtp and <call-id>.rtcp How many files should rtpproxy produce for one call? Should also a file or two for 2nd channel be generated? How can I record both audio ways? Thank you Marko The whole function route[NATMANAGE]: # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } ----- Original Message ----- From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Marko Tirs <marko.tirs(a)yahoo.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 14, 2017 5:23 PM Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording Hello, I think you can call the start recording function just after rtpproxy_manage(). If you don't use the start recording function, is the audio going both ways without problems? Cheers, Daniel

1 0

rtpproxy doesn't record after start_recording
by Marko Tirs 14 Mar '17

14 Mar '17

Hello experts, I'm newbie in Kamailio and have a problem with my 1st installation. I just want to control the calls between SIP clients registered by Kamailio and record the conversations. I installed rtpproxy and started with rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.0.13 -r /home/user1/rtpproxy -d DBUG:LOG_LOCAL0 After start_recording() in the function onreply_route[MANAGE_REPLY] rtpproxy creates the files <call-id>.rtp and <call-id>.rtcp but they remain at zero length. 2nd problem is that the media stream is broken in one direction after start_recording() has been executed, the callee doesn't hear the caller. How is the right Kamailio configuration for recording conversations? Thank you Regards Marko Modified kamailio-basic.cfg partly: #!KAMAILIO #!define WITH_NAT #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (is_method("INVITE") and (status=="200")) { xlog("START_RECORDING #1\n"); start_recording(); xlog("START_RECORDING #2\n"); } else if(status=~"[12][0-9][0-9]") route(NATMANAGE); }

2 1

Is kamailio support load balancing for websockets
by SureshTummala 14 Mar '17

14 Mar '17

Hi, I want to use kamailio as a load balancer for web sockets. Is kamailio supports LB functionality for web-sockets? I can able to use kamailio as a proxy and can make calls with websockets. Now i want to add LB functionality. Setup: UA1 -->LB -->Proxy UA2 <--LB <--Proxy LB is not forwarding the packets from Proxy to UA Please help me Regards,Suresh Tummala

1 0

Several Asterisk on this same IP
by przeqpiciel 14 Mar '17

14 Mar '17

I would like to ask you how you deal with several asterisks and kamailio on that same IP address, I have installation where i route 5060 to internal server with kamailio and I would like to route RTP to asterisks, but is any way to get around a problem with RTP ports collisions?

6 19

how to unregister an remote registration at uac module
by José Lopes 13 Mar '17

13 Mar '17

Hello, I am using the module uac to do remote registration. But sometimes i need to do unregister, to control or debug the registration process. I tried using kamcmd with uac.reg_disable, but this doesn't unregister. There is any way to do it? Best regards, Jose Lopes

2 2

Segmentation fault - dialog RPC
by pars3c 13 Mar '17

13 Mar '17

Dear all, many times, when i call dialog_list on dialog module with jsonrpc, kamailio crash. The version i use is: 4.4.2 Thanks for any suggestion. #0 0x00007f2a706eb00a in internal_rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940, profile_name=0x7fffb9fa8b10, value=0x0) at dialog.c:1666 1666 if ((!value || (STR_EQ(*value, ph->value))) && ph->dlg) { The dump , with bt full is this : #0 0x00007f2a706eb00a in internal_rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940, profile_name=0x7fffb9fa8b10, value=0x0) at dialog.c:1666 profile = 0x7f2a5098fb40 ph = 0x0 i = 10 #1 0x00007f2a706eb6e4 in rpc_profile_print_dlgs (rpc=0x7f2a4fdf99a0, c=0x7f2a4fdf9940) at dialog.c:1772 profile_name = {s = 0x1c78b00 "outbound", len = 8} value = {s = 0x0, len = 0} #2 0x00007f2a4fbe45c6 in jsonrpc_dispatch (msg=0x7fffb9fa9f10, s1=0x0, s2=0x0) at jsonrpc-s_mod.c:1091 rpce = 0x7f2a75609360 ctx = 0x7f2a4fdf9940 ret = 0 nj = 0x1c76af0 val = {s = 0x1c77b20 "dlg.profile_list", len = 16} __FUNCTION__ = "jsonrpc_dispatch" #3 0x000000000041f4ad in do_action (h=0x7fffb9fa9e30, a=0x7f2a75584700, msg=0x7fffb9fa9f10) at action.c:1054 ret = -5 v = 4288256 dst = {send_sock = 0x1b9fa9b23, to = {s = {sa_family = 14080, sa_data = "Xu*\177\000\000\b7Xu*\177\000"}, sin = { sin_family = 14080, sin_port = 30040, sin_addr = {s_addr = 32554}, sin_zero = "\b7Xu*\177\000"}, sin6 = { sin6_family = 14080, sin6_port = 30040, sin6_flowinfo = 32554, sin6_addr = {__in6_u = { __u6_addr8 = "\b7Xu*\177\000\000S \205\000\060\000\000", __u6_addr16 = {14088, 30040, 32554, 0, 8275, 133, 48, 0}, __u6_addr32 = {1968715528, 32554, 8724563, 48}}}, sin6_scope_id = 26}}, id = 0, proto = 16 '\020', send_flags = { f = 155 '\233', blst_imask = 250 '\372'}} tmp = 0x7fffb9fab700 "\t" new_uri = 0x0 end = 0x0 crt = 0x7fffb9fa92d0 "" cmd = 0x7f2a754adcc0 len = 0 user = 0 uri = {user = {s = 0x12c <Address 0x12c out of bounds>, len = 0}, passwd = {s = 0x7fffb9fa92d0 "", len = 1969341184}, host = { s = 0x7f2a75475010 "\001", len = 4742114}, port = {s = 0x0, len = 0}, params = { s = 0x98206f7473 <Address 0x98206f7473 out of bounds>, len = 1969341256}, sip_params = { s = 0x7fffb9fa91e0 "`\222\372\271\377\177", len = 6538286}, headers = {s = 0x1c2b6f0 "\300\300\302\001", len = 0}, port_no = 50838, proto = 30049, type = 32554, flags = 0, transport = {s = 0x0, len = -940795965}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x6222286e00000000 <Address 0x6222286e00000000 out of bounds>, len = 577270895}, maddr = { s = 0x3a746e65696c6309 <Address 0x3a746e65696c6309 out of bounds>, len = 0}, method = {s = 0x75c5f0 "core", len = -1174760864}, lr = {s = 0x764661 "rval_destroy", len = 7718524}, r2 = {s = 0x7f2a7561c348 "\002", len = 1967607824}, gr = {s = 0x7fffb9fa91b8 "H\303au*\177", len = 1967607824}, transport_val = {s = 0x7f2a7561c300 "\300\004", len = 1969341256}, ttl_val = {s = 0x7f2a7557e200 "\020", len = -1174760316}, user_param_val = {s = 0x7f2a75583700 "\002", len = 1969341256}, maddr_val = {s = 0x416f00 "1\355I\211\321^H\211\342H\203\344\360PTI\307\300\060\336q", len = -1174751488}, method_val = { ---Type <return> to continue, or q <return> to quit--- s = 0x0, len = 0}, lr_val = {s = 0x7fffb9fa9260 "Ð\372\271\377\177", len = 5581478}, r2_val = {s = 0x0, len = 0}, gr_val = { s = 0x3a28d <Address 0x3a28d out of bounds>, len = 1969177728}} next_hop = {user = {s = 0x1c2b6f0 "\300\300\302\001", len = 1969330344}, passwd = { s = 0x69692c22 <Address 0x69692c22 out of bounds>, len = 29537760}, host = { s = 0x400000000 <Address 0x400000000 out of bounds>, len = -1174761304}, port = { s = 0x71657228 <Address 0x71657228 out of bounds>, len = 1969342102}, params = {s = 0x0, len = 1}, sip_params = {s = 0x0, len = -1174761328}, headers = {s = 0x1 <Address 0x1 out of bounds>, len = 2}, port_no = 40672, proto = 47610, type = 32767, flags = 3120206016, transport = {s = 0x7a6563697220616c <Address 0x7a6563697220616c out of bounds>, len = 1701736297}, ttl = { s = 0x7f2a75580848 "\001", len = -1174760828}, user_param = {s = 0x7fffb9fa9f10 "\b\005", len = -1174757840}, maddr = { s = 0x65722e2e223a7069 <Address 0x65722e2e223a7069 out of bounds>, len = 1702129517}, method = {s = 0x7f2a7561c696 "/sip", len = 29855024}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x500000004 <Address 0x500000004 out of bounds>, len = 4}, transport_val = {s = 0x400000004 <Address 0x400000004 out of bounds>, len = 6552774}, ttl_val = { s = 0x7fffb9fa9060 "", len = 1967607824}, user_param_val = {s = 0x1000001 <Address 0x1000001 out of bounds>, len = 1967607824}, maddr_val = {s = 0x7fffb9fa90b0 "\020PGu*\177", len = 1956545302}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14985000}, r2_val = { s = 0x165e80 <Address 0x165e80 out of bounds>, len = 1792216}, gr_val = {s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 8}} u = 0x7f2a75583f30 port = 0 dst_host = 0x7f2a75583700 i = 32767 flags = 0 avp = 0x7fffb9fa9e30 st = {flags = 151652617, id = 29193, name = {n = 0, s = {s = 0x6e65090a00000000 <Address 0x6e65090a00000000 out of bounds>, len = -1174761256}, re = 0x6e65090a00000000}, avp = 0x7fffb9fa9000} sct = 0x7fffb9fa94d0 sjt = 0x7fffb9fa9f10 rve = 0x416f00 mct = 0x7fffb9fab700 rv = 0x92922726f727265 rv1 = 0x0 c1 = {cache_type = 3120205776, val_type = 32767, c = {avp_val = {n = -940803561, s = { s = 0x31c7ec7a17 "\213D$8H\201\304\070\002", len = -1174757616}, re = 0x31c7ec7a17}, pval = {rs = { s = 0x31c7ec7a17 "\213D$8H\201\304\070\002", len = -1174757616}, ri = -1174757840, flags = 32767}}, i2s = "vp(s:inserver_erPor)\","} s = {s = 0x1c2b6f0 "\300\300\302\001", len = 0} srevp = {0x0, 0x0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, ---Type <return> to continue, or q <return> to quit--- select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action" #4 0x000000000042c218 in run_actions (h=0x7fffb9fa9e30, a=0x7f2a75584700, msg=0x7fffb9fa9f10) at action.c:1549 t = 0x7f2a75584700 ret = -1 mod = 0x0 ms = 0 __FUNCTION__ = "run_actions" #5 0x000000000041f416 in do_action (h=0x7fffb9fa9e30, a=0x7f2a75585b10, msg=0x7fffb9fa9f10) at action.c:1045 ret = 1 v = 1 dst = {send_sock = 0x7fffb9fa9b50, to = {s = {sa_family = 64710, sa_data = "c\000\000\000\000\000\060\233\372\271\377\177\000"}, sin = {sin_family = 64710, sin_port = 99, sin_addr = {s_addr = 0}, sin_zero = "0\233\372\271\377\177\000"}, sin6 = { sin6_family = 64710, sin6_port = 99, sin6_flowinfo = 0, sin6_addr = {__in6_u = { __u6_addr8 = "0\233\372\271\377\177\000\000Ù¨bu*\177\000", __u6_addr16 = {39728, 47610, 32767, 0, 43225, 30050, 32554, 0}, __u6_addr32 = {3120208688, 32767, 1969400025, 32554}}}, sin6_scope_id = 366}}, id = 10, proto = -24 '\350', send_flags = {f = 234 '\352', blst_imask = 98 'b'}} tmp = 0x1467562aa20 <Address 0x1467562aa20 out of bounds> new_uri = 0x7a0a02 "parse_headers" end = 0x79db5c "core: parser/msg_parser.c" crt = 0x40 <Address 0x40 out of bounds> cmd = 0x7f2a75475010 len = 1967607824 user = 32554 uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x7fffb9fa9b70 "@\234\372\271\004", len = 6529985}, host = { s = 0x416f00 "1\355I\211\321^H\211\342H\203\344\360PTI\307\300\060\336q", len = -1174751488}, port = {s = 0x0, len = 0}, params = {s = 0x7fffb9fa9ba0 "p\234\372\271\377\177", len = 6661320}, sip_params = {s = 0x7fffb9fa99f0 "", len = 1967607824}, headers = {s = 0x0, len = 1967607824}, port_no = 39488, proto = 47610, type = 32767, flags = 6552774, transport = { s = 0x7fffb9fa9a20 "", len = 1967607824}, ttl = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 1967607824}, user_param = {s = 0x7fffb9fa9a70 "\240\232\372\271\377\177", len = 1956545302}, maddr = { s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 0}, method = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14986448}, lr = {s = 0x1658d8 <Address 0x1658d8 out of bounds>, len = 1790768}, r2 = { s = 0x1ed710 <Address 0x1ed710 out of bounds>, len = 8}, gr = {s = 0x12e <Address 0x12e out of bounds>, len = 1967607824}, transport_val = {s = 0x14600000000 <Address 0x14600000000 out of bounds>, len = 1969417808}, ttl_val = { s = 0x7fffb9fa9aa0 "\240\233\372\271\377\177", len = 4742114}, user_param_val = {s = 0x0, len = 1967607824}, maddr_val = { s = 0x0, len = 0}, method_val = {s = 0x7fffb9fa9ba0 "p\234\372\271\377\177", len = 6534031}, lr_val = {s = 0x79d630 "core", len = 6552774}, r2_val = {s = 0x7fffb9fa9b00 "n\001", len = 1969417880}, gr_val = {s = 0x7f2a7562aa20 "", len = 1969399988}} next_hop = {user = {s = 0x7fffb9fa9820 "", len = 1967607824}, passwd = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 1967607824}, host = {s = 0x7fffb9fa9870 "\240\230", <incomplete sequence \372\271>, len = 1956545302}, port = { ---Type <return> to continue, or q <return> to quit--- s = 0x1c588e1 "\311\305\001", len = 0}, params = {s = 0x1000000 <Address 0x1000000 out of bounds>, len = 14986216}, sip_params = {s = 0x1659c0 <Address 0x1659c0 out of bounds>, len = 1969416936}, headers = {s = 0x7f2a7562aa20 "", len = 1969399993}, port_no = 301, proto = 0, type = ERROR_URI_T, flags = 0, transport = { s = 0x7f2a7562a8d9 "Content-Type: application/json-rpc\r\nCache-Control: no-cache\r\nPragma: no-cache\r\nUser-Agent: Java/1.8.0_121\r\nHost: \r\nAccept: text/html, image/gif, image/jpeg, *; q=.2, */*; q=.2\r\nConn"..., len = 0}, ttl = { s = 0xb9fa98a0 <Address 0xb9fa98a0 out of bounds>, len = 1969416936}, user_param = {s = 0x0, len = 1967607824}, maddr = { s = 0x0, len = 0}, method = {s = 0x7fffb9fa99a0 "\240\233\372\271\377\177", len = 6534031}, lr = {s = 0x79d630 "core", len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x7fffb9fa9e7c "\260\f\222\017", len = -1174757736}, transport_val = { s = 0x79d630 "core", len = -1174757704}, ttl_val = {s = 0x7a0a10 "get_hdr_field", len = 7986012}, user_param_val = { s = 0xe8 <Address 0xe8 out of bounds>, len = 1967607824}, maddr_val = {s = 0x79d630 "core", len = 1969416864}, method_val = { s = 0x7a0a10 "get_hdr_field", len = 7986012}, lr_val = {s = 0xe8 <Address 0xe8 out of bounds>, len = 1967607824}, r2_val = { s = 0x7f2a75475010 "\001", len = 1969416864}, gr_val = {s = 0x7f2a7562eaa0 <incomplete sequence \350>, len = 1969399988}} u = 0x7f2a7562ee50 port = 0 dst_host = 0x80067e961 i = 0 flags = 0 avp = 0x7f2a75475010 st = {flags = 3120207856, id = 32767, name = {n = 1967607824, s = {s = 0x7f2a75475010 "\001", len = 1969239958}, re = 0x7f2a75475010}, avp = 0x7f2a75475010} sct = 0x4b9fa9c40 sjt = 0x79d630 rve = 0x7f2a75582f20 mct = 0x7fffb9fab700 rv = 0x63fcc6 rv1 = 0x0 c1 = {cache_type = 43, val_type = RV_NONE, c = {avp_val = {n = 0, s = {s = 0x0, len = 7718384}, re = 0x0}, pval = {rs = {s = 0x0, len = 7718384}, ri = 26, flags = 0}}, i2s = "\t\000\000\000\000\000\000\000huTu*\177\000\000\000\000\000\000\000"} s = {s = 0x7fffb9fa99f0 "", len = 1969239512} srevp = {0x3000000018, 0x7fffb9fa9ab0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __FUNCTION__ = "do_action"

2 2

unable to receive response from freeswitch
by ysuryas . 13 Mar '17

13 Mar '17

I have configured kamailio (192.168.56.102) and freeswitch(192.168.56.103) in separate virtualboxes when making a call, the invite goes to proxy and it is forwarded to freeswitch but response from freeswitch is "unreachable port" attached the wireshark trace: I don't why 10.0.2.15 is making the Invite to freeswitch, i was expecting 192.168.56.102 to do that. UA -> 192.168.56.102 -> 192.168.56.103 Any suggestions on this? [image: Inline image 1] Thanks, Surya

2 1

rtpproxy doesn't record after start_recording
by Marko Tirs 11 Mar '17

11 Mar '17

Hello experts, I'm newbie in Kamailio and have a problem with my 1st installation. I just want to control the calls between SIP clients registered by Kamailio and record the conversations. I installed rtpproxy and started with rtpproxy -F -s udp:127.0.0.1:7722 -l 192.168.0.13 -r /home/user1/rtpproxy -d DBUG:LOG_LOCAL0 After start_recording() in the function onreply_route[MANAGE_REPLY] rtpproxy creates the files <call-id>.rtp and <call-id>.rtcp but they remain at zero length. 2nd problem is that the media stream is broken in one direction after start_recording() has been executed, the callee doesn't hear the caller. How is the right Kamailio configuration for recording conversations? Thank you Regards Marko Modified kamailio-basic.cfg partly: #!KAMAILIO #!define WITH_NAT #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (is_method("INVITE") and (status=="200")) { xlog("START_RECORDING #1\n"); start_recording(); xlog("START_RECORDING #2\n"); } else if(status=~"[12][0-9][0-9]") route(NATMANAGE); }

1 0

Dispatcher Module - Hava limit?
by Rodrigo Moreira 11 Mar '17

11 Mar '17

Hello, I am experiencing the following problem: my dispatcher module is losing many calls when the scenario for balancing is about 175 CPS or more. Is there a limit that the module can do? Where do I find this information? Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069

3 6

Multiple parallel branches, one invalid
by Nathan Ward 10 Mar '17

10 Mar '17

Hi, I have a scenario I’m not sure of the best way to solve. I use alias_db and have multiple destinations for a single number, and have append_branches set to 1. I then use lookup_branches to get details from the location table for each branch. I then do t_load_contacts and t_next_contacts, then t_relay to parallel fork the calls to the destination, and all the phones ring at once. This all works great if all the destinations from the dbaliases table are online. However, if one of them is offline, lookup_branches says "Not found in usrloc”. This branch is still there, and the call is attempted. In my environment, this then does a DNS lookup, can’t resolve the domain, and the branch fails. This is the first branch that is tried, and when this branch has an error like this, it causes t_relay() to fail - I check for the output of t_relay() and reply with an error - per the default config that Kamailio ships with. I guess what I need to figure out, is if there’s a way to have t_relay fail only if all branches fail? Or, is there a more elegant way to handle this? I should add, I only allow a single registration per subscriber - so, lookup_branches only returns one branch per alias from alias_db_lookup. -- Nathan Ward

2 1

404 not here for BYE
by Mikhail Khomenkov 10 Mar '17

10 Mar '17

Hi i have some kind of issue with BYE Asterisk send invite to Kamailio1, Kamailio1 proxy to Kamailio2, Kamailio2 found UAC and send invite to it. UAC response 200OK -> Kamailio2 -> Kamailio1 -> Asterisk. Session established. But then UAC sending BYE to Kamailio2 , Kamailio2 sending it to Kamailio1 and Kamailio1 response 404 not here (not sending BYE to Asterisk) Any suggestions? 54.236.97.30 107.21.18.6 10.0.0.68 108.41.170.187 Asterisk Kamailio1 Kamailio2 UAC | invite | | | |--------------------------------> | invite | | | |------------------------------->| invite | | | |----------------------------------->| | | | 200OK | | | 200OK |<-----------------------------------| | 200OK |<-------------------------------| | |<--------------------------------| | | BYE | | BYE |<-----------------------------------| | |<-------------------------------| | | | 404 Not here | | | |------------------------------->| 404 Not here | | | |----------------------------------->| # U 107.21.18.6:5060 -> 10.0.0.68:5060 INVITE sip:88919640010@sip7.keku.com SIP/2.0. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Max-Forwards: 68. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 237. . v=0. o=root 39119558 39119558 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 54.236.97.30. t=0 0. m=audio 53626 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0;rport=5060. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 INVITE sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Max-Forwards: 66. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295 <(609)%20375-6295>@54.236.97.30: 5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 645. . v=0. o=root 39119558 39119558 IN IP4 107.23.47.88. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 107.23.47.88. t=0 0. m=audio 10054 RTP/SAVPF 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:10055. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:v7hy4Ix2Re87vG/ Fiymv20UGZ89xoWPp/dFzmiHq. a=setup:actpass. a=fingerprint:sha-1 18:F3:50:26:22:26:81:88:FB:95: F8:4B:3A:A7:9F:A6:F5:B7:B8:E3. a=ice-ufrag:ZgtwCEvD. a=ice-pwd:a0Tt54oToZFj9b42QXOM2Fq1VR. a=candidate:RzNgC40JM1mFcCUB 1 UDP 2130706431 107.23.47.88 10054 typ host. a=candidate:RzNgC40JM1mFcCUB 2 UDP 2130706430 107.23.47.88 10055 typ host. # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. To: <sip:88919640010@sip6.keku.com>. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 283. . v=0. o=- 620006012 3 IN IP4 172.22.1.2. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 172.22.1.2. t=0 0. m=audio 56064 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoNS33aJgrSvBRBRtHHhjlA+ rLvd9ldUiUxXzoem. a=sendrecv. # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. 83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 215. . v=0. o=- 620006012 3 IN IP4 107.23.47.88. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 107.23.47.88. t=0 0. m=audio 10072 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:10073. # U 107.21.18.6:5060 -> 10.0.0.68:5060 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.21.18.6:5060;branch=z9hG4bK67a4. ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK4e4d5063;rport=5060. Route: <sip:107.23.47.88;lr>. Max-Forwards: 68. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bK67a4. 3a3af8e184f57334c9d42b8ce7d77849.0. Via: SIP/2.0/UDP 107.21.18.6:5060;rport=5060;branch=z9hG4bK67a4. ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060;received=54.236.97.30;branch= z9hG4bK4e4d5063;rport=5060. Max-Forwards: 66. From: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295 <(609)%20375-6295>@54.236.97.30: 5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 . . # U 108.41.170.187:53487 -> 10.0.0.68:5060 BYE sip:6093756295 <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.22.1.2:53487;branch=z9hG4bK-524287-1--- 1034af445c791a63;rport. Max-Forwards: 70. Route: <sip:107.23.47.88;lr>. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 BYE sip:6093756295 <(609)%20375-6295>@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060;branch=z9hG4bKbdb2. c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. Max-Forwards: 68. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 107.21.18.6:5060 -> 10.0.0.68:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 107.23.47.88:5060;rport=5060;branch=z9hG4bKbdb2. c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.22.1.2:53487;received=108.41.170.187;branch=z9hG4bK- 524287-1---1034af445c791a63;rport=53487. To: "6093756295 <(609)%20375-6295>" <sip:6093756295@54.236.97.30>; tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0.

2 1

sip6 BYE
by Mikhail Khomenkov 10 Mar '17

10 Mar '17

Hi i have some kind of issue with BYE Asterisk send invite to Kamailio1, Kamailio1 proxy to Kamailio2, Kamailio2 found UAC and send invite to it. UAC response 200OK -> Kamailio2 -> Kamailio1 -> Asterisk. Session established. But then UAC sending BYE to Kamailio2 , Kamailio2 sending it to Kamailio1 and Kamailio1 response 404 not here (not sending BYE to Asterisk) Any suggestions? 54.236.97.30 107.21.18.6 10.0.0.68 108.41.170.187 Asterisk Kamailio1 Kamailio2 UAC | invite | | | |--------------------------------> | invite | | | |------------------------------->| invite | | | |----------------------------------->| | | | 200OK | | | 200OK |<-----------------------------------| | 200OK |<-------------------------------| | |<--------------------------------| | | BYE | | BYE |<-----------------------------------| | |<-------------------------------| | | | 404 Not here | | | |------------------------------->| 404 Not here | | | |----------------------------------->| # U 107.21.18.6:5060 -> 10.0.0.68:5060 INVITE sip:88919640010@sip7.keku.com SIP/2.0. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Max-Forwards: 68. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 237. . v=0. o=root 39119558 39119558 IN IP4 54.236.97.30. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 54.236.97.30. t=0 0. m=audio 53626 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0;rport=5060. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 INVITE sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Max-Forwards: 66. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip6.keku.com>. Contact: <sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Date: Fri, 10 Mar 2017 02:30:12 GMT. Min-SE: 120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Cisco-Guid: 1241711126_45859092(a)23.29.27.22. Content-Type: application/sdp. Content-Length: 645. . v=0. o=root 39119558 39119558 IN IP4 107.23.47.88. s=Asterisk PBX 1.8.28-cert5. c=IN IP4 107.23.47.88. t=0 0. m=audio 10054 RTP/SAVPF 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:10055. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:v7hy4Ix2Re87vG/Fiymv20UGZ89xoWPp/dFzmiHq. a=setup:actpass. a=fingerprint:sha-1 18:F3:50:26:22:26:81:88:FB:95:F8:4B:3A:A7:9F:A6:F5:B7:B8:E3. a=ice-ufrag:ZgtwCEvD. a=ice-pwd:a0Tt54oToZFj9b42QXOM2Fq1VR. a=candidate:RzNgC40JM1mFcCUB 1 UDP 2130706431 107.23.47.88 10054 typ host. a=candidate:RzNgC40JM1mFcCUB 2 UDP 2130706430 107.23.47.88 10055 typ host. # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. To: <sip:88919640010@sip6.keku.com>. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "88919640010"<sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. User-Agent: X-Lite release 4.9.6 stamp 82167. Allow-Events: talk, hold. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.d6f7e6b1f007e0958a99c6a6342e4e48.1. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 283. . v=0. o=- 620006012 3 IN IP4 172.22.1.2. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 172.22.1.2. t=0 0. m=audio 56064 RTP/SAVPF 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoNS33aJgrSvBRBRtHHhjlA+rLvd9ldUiUxXzoem. a=sendrecv. # U 10.0.0.68:5060 -> 107.21.18.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.83b3107c510ddfe551896134a5e82554.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK3d97649d;rport=5060. Record-Route: <sip:107.23.47.88;lr>. Record-Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: <sip:88919640010@sip6.keku.com>;tag=d6323b02. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 215. . v=0. o=- 620006012 3 IN IP4 107.23.47.88. s=X-Lite release 4.9.6 stamp 82167. c=IN IP4 107.23.47.88. t=0 0. m=audio 10072 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:10073. # U 107.21.18.6:5060 -> 10.0.0.68:5060 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.21.18.6:5060 ;branch=z9hG4bK67a4.ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK4e4d5063;rport=5060. Route: <sip:107.23.47.88;lr>. Max-Forwards: 68. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 ACK sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bK67a4.3a3af8e184f57334c9d42b8ce7d77849.0. Via: SIP/2.0/UDP 107.21.18.6:5060 ;rport=5060;branch=z9hG4bK67a4.ff7b9cc9b887d892d7fd83e6f046b4d4.0. Via: SIP/2.0/UDP 54.236.97.30:5060 ;received=54.236.97.30;branch=z9hG4bK4e4d5063;rport=5060. Max-Forwards: 66. From: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. To: <sip:88919640010@sip7.keku.com>;tag=d6323b02. Contact: <sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1>. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. . # U 108.41.170.187:53487 -> 10.0.0.68:5060 . . # U 108.41.170.187:53487 -> 10.0.0.68:5060 BYE sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;branch=z9hG4bK-524287-1---1034af445c791a63;rport. Max-Forwards: 70. Route: <sip:107.23.47.88;lr>. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 10.0.0.68:5060 -> 107.21.18.6:5060 BYE sip:6093756295@54.236.97.30:5060;alias=107.21.18.6~5060~1 SIP/2.0. Via: SIP/2.0/UDP 107.23.47.88:5060 ;branch=z9hG4bKbdb2.c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. Max-Forwards: 68. Route: <sip:107.21.18.6;lr=on>. Contact: <sip:88919640010@108.41.170.187:53487;rinstance=37d7b85846910961>. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. User-Agent: X-Lite release 4.9.6 stamp 82167. Content-Length: 0. . # U 107.21.18.6:5060 -> 10.0.0.68:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 107.23.47.88:5060 ;rport=5060;branch=z9hG4bKbdb2.c896050feb9a2151548b9fe6c3f776fb.0. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0. . # U 10.0.0.68:5060 -> 108.41.170.187:53487 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 172.22.1.2:53487 ;received=108.41.170.187;branch=z9hG4bK-524287-1---1034af445c791a63;rport=53487. To: "6093756295" <sip:6093756295@54.236.97.30>;tag=as372f68e0. From: <sip:88919640010@sip6.keku.com>;tag=d6323b02. Call-ID: 01e9e3e03dd2995d720bff297a226e83@54.236.97.30:5060. CSeq: 2 BYE. Server: kamailio (4.2.5 (x86_64/linux)). Content-Length: 0.

3 2

Bounty for topos fix
by Trevor Peirce 09 Mar '17

09 Mar '17

Hello, I would like to offer a $100 USD bounty if a developer is able to provide a fix for the two topos problems I have experienced. I will add another $50 if this can be solved within 48 hours of the time stamp of this message. The bug report is at: https://github.com/kamailio/kamailio/issues/1005 You may find the most relevant information in my last few updates. Please refer to mar1-nat.txt and mar1-reinvite.txt for the most recent SIP captures, and the post below that for example database records. Steps to reproduce the NAT problem can be found at https://github.com/kamailio/kamailio/issues/1005#issuecomment-283255720 Some fine print to clarify the offer: Payment is contingent on resolving both described use-cases. Payment will be sent via your choice of PayPal, cheque drawn on a US bank, or cheque drawn on a Canadian bank. Payment will be issued once the patch is accepted by the Kamailio project, although the bonus for a quick resolution simply requires a suitable patch to be submitted to the bug tracker within 48 hours from now. Thanks! -- Trevor Peirce AcroVoice Solutions Inc

1 1

Changes in LCR Target List doesn't takes place immediately
by Nitesh Lohchab 09 Mar '17

09 Mar '17

Hello, I believe LCR Target list is stored in the database and any changes made here should affect the behaviour of the calls imedialtely. I have changed one of my Target List Rule , where I changed my gateway Id to another gateway. But the calls are still being routed via previous gateway. Is there something I can do to make the changes in production hours. Sometimes one of carriers delivers bad call quality I would like to switch the carriers. -- Regards, Nitesh Lohchab

2 1

rtpengine sending rtp to wrong endpoint after reinvite
by Grant Bagdasarian 09 Mar '17

09 Mar '17

Hi, One of our customers is using a SEMS box to place two outbound calls using our sip trunk. Once the first call is connected a second call is placed and when the second call answers their server sends a re-invite to switch audio ports so the rtp traffic doesn't flow through their server anymore but is routed inside our platform. Basically, they just switch SDP's of both calls. It seems like a random issue, and is not really reproducible, except for placing multiple calls and sometimes both parties can hear each other, other times they can't, because rtpengine fails (I think) to update the endpoint and keeps sending rtp back to their server for one of the call legs. We tried to reproduce the case using a freeswitch box and it worked every time. After the reinvite, the rtp remained within our platform. The signaling in both cases still goes through the freeswitch or sems for call control. Does anyone have experience with this case? Or seen the issue before where rtpengine keeps sending rtp to the original endpoint? Regards, Grant Bagdasarian CM

2 3

Access avp between sip transactions
by Grant Bagdasarian 09 Mar '17

09 Mar '17

Hi, Is it possible to access an avp variable from a different sip transaction? On an INVITE is set an avp, but on ReINVITE I would like to access this avp again and get the value. Currently the value is null, which makes sense. Not sure if this is possible and if it violates any rules inside the Kamailio engine? I guess I could use a htable, but I'm just wondering if there is a built-in way for doing this. Regards, Grant Bagdasarian CM

2 1

Kamailio Presence help required.
by SamyGo 08 Mar '17

08 Mar '17

Hi, I'm in need of making/tweak an existing Kamailio Presence setup which is giving some tough time. *Whats already working:* BLF dialog states changes are already sent across the users. SCA is working as well. *What isn't working:* When a User comes online then it sends SUBSCRIBE with *Event: dialog* and don't get notified of its subscribers state right then unless the monitored extensions make a call (BLF works) *Why is it not working:* As evident from wireshark traces, the user IP phones (Test sets: Polycoms, Yealink, CISCO, Grandstream) don't send our *Event: presence* rather only *Event:dialog* and Kamailio do not send NOTIFY out to everyone. Though yes there is an internally generated PUBLISH seen and handled properly upon registration state changes. Jitsi has been tested and Kamailio send out these registration state change info to jitsi, somce jitsi sends *Event:presence *in SUBSCRIBE. I need dialog based NOTIFY to be sent out on registration state-change. Need pointers and help on the topic, looking forward to some feedback. Regards, Sammy

4 8

TCP stack is not doing load distribution among children
by Surendra Pullaiah 08 Mar '17

08 Mar '17

Hi Daniel, Thanks for your suggestions. The approaches which you have suggested are good, event I thought before initiating this query (since am working with kamailio for last three years), but my requirement is to distribute on single connection. Since it requires code change, started working on it. Really thanks for understanding my concern. @juha: Dear juha, it is not broken TCP implementation, kamailio TCP stack well implemented. If you would like to build a server to handle HTTP and MSRP protocols at higher rate, here is the place. But for SIP we have options to handle either UDP/TCP. Thanks surendra

1 0

Kamailio push for Android VoIP Application and VoIP Contacts update
by sscc 08 Mar '17

08 Mar '17

good day, i have a running kamailio server having 200 users. I maintain the user ids in a vcf (vCard Format) form. i have developed an Android VoIP application. Android application is given to the user along with the contact vcf. The contact list get increased day by day and consequently the user ids in vcf (vCard Format) as well. the application is also updated. Aim is to update application and the contact list by sending push notification to user. (we don't want to place our application at play store). i want to do push notification from Kamilio server for application and contact list update. Is there any way to do push notification from server to users to update the app and VoIP contact list. regards sscc -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-push-for-Android-VoIP-Appl… Sent from the Users mailing list archive at Nabble.com.

1 0

TCP stack is not doing load distribution among children for SIP
by Surendra Pullaiah 08 Mar '17

08 Mar '17

Dear Daniel, A small concern in kamailio, existing TCP stack is not doing load distribution among children for SIP messages. For example UE-->kamailio1 (TCP listen (4 children)) ----->kamailio2 (TCP Listen 4 children) As per stack if UE opens a connection with kamailio1 based on free children it is sharing the FD, so no problem towards from UE to kamailio1, then need requirement to forward same sip message to kamailio2, here kamailio1 is opening connection with kamailio2 and then forwarding. Suppose if we forward two requests to kamailio2, only one children is getting used among four. With this design suppose if we want to run our kamailio as two sip servers we may not achieve good results when performance matters. Please suggest something on this to move further. Regards surendra

3 5

new keepalive module ?
by Guillaume Bour 07 Mar '17

07 Mar '17

Hi all, I would like to implement destinations keepalive in drouting module (as it is done in dispatcher). But instead of duplicating what's implemented in dispatcher, I think it would be more clever to create a new module dedicated to pinging destinations, and to plug drouting, dispatcher and other modules to this new one What's you opinion about that ? Cheers, Guillaume

1 0

Kamailio core dumped
by Alexey V. Panfilov 07 Mar '17

07 Mar '17

Hi there, I've kamailio 4.2.3 (i386/freebsd) which works perfectly with 200-300 cps for years. Now I need migrate to freebsd amd64. I've installed kamailio 4.4.5 (amd64/freebsd), copied kamailio.cfg, made a test call - it was all right. But when I start work load (200-300 cps), kamailio core dumped and exited. In logs (skipped "<script>:" having): ======================= Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x809937000, 976); Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x809937000, 976) called from core: msg_translator.c: build_req_buf_from_sip_req(2154), module: core; Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: <core> [msg_translator.c:2159]: build_req_buf_from_sip_req(): out of memory Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: tm [t_fwd.c:462]: prepare_new_uac(): could not build request Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: tm [t_fwd.c:1723]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32034]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (2/SL) Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32035]: ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x809937000, 864); Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32035]: ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x809937000, 864) called from tm: t_reply.c: relay_reply(1885), module: tm; Free fragment not found! Mar 1 12:39:32 sip7 /usr/local/sbin/kamailio[32035]: ERROR: tm [t_reply.c:1887]: relay_reply(): ERROR: relay_reply: cannot alloc reply shmem Mar 1 12:39:37 sip7 /usr/local/sbin/kamailio[32038]: : <core> [mem/f_malloc.c:598]: fm_free(): BUG: fm_free: bad pointer 0x61743b3e6d6f632e (out of memory block!), called from tm: h_table.c: free_cell_helper(226) - aborting Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32031]: ALERT: <core> [main.c:740]: handle_sigs(): child process 32034 exited by a signal 10 Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32031]: ALERT: <core> [main.c:743]: handle_sigs(): core was generated Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32031]: INFO: <core> [main.c:755]: handle_sigs(): terminating due to SIGCHLD Mar 1 12:39:39 sip7 /usr/local/sbin/kamailio[32044]: INFO: <core> [main.c:810]: sig_usr(): signal 15 received ======================= I've tried to start with -m 512 - nothing changes. Hardware: Xeon(R) CPU E5-2660, 16Gb RAM, 1Tb HDD. Software: OS: FreeBSD 10.3-STABLE (r313760) amd64 version: kamailio 4.4.5 (amd64/freebsd) f98162 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, select, kqueue. id: f98162 compiled on 12:25:48 Mar 1 2017 with clang 3.4 bt is here: http://tmp.lehis.ru/kam_gdb.txt I can't enable debug symbols for gdb (tried gmake mode=debug, which added EXTRA_DEBUG to flags). How can I do it? I need in direction for resolve this issue, please, help. Thanks. -- Best regards, Alexey V. Panfilov

2 3

Kamailio 4.2 crashed
by Igor Potjevlesch 07 Mar '17

07 Mar '17

Hello, I got two coredumps: (gdb) bt full #0 0x000000000061ab8e in qm_insert_free (qm=0x7f0f7fca7000, frag=0x7f0f8f6f6268) at mem/q_malloc.c:180 f = 0x7f0f8f6c45e8 prev = 0x0 hash = 5 And (gdb) bt full #0 0x000000000061b935 in qm_detach_free (qm=0x7f0f7fca7000, frag=0x7f0f8f6d70b8) at mem/q_malloc.c:271 prev = 0x7f0f7fca7248 next = 0x7f0f8f6c45e8 What are necessary to diagnose how this issue occurs? Thank you for your support. Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 3

SCTP support for CDP
by Sergey Zyrianov 07 Mar '17

07 Mar '17

Hello, Can I enable use of SCTP for diameter connections by CDP module? Thanks, Sergey.

3 6

Installation of Siremi 4.3.0 on Centos 7.3 ?
by Lucas Viallon 07 Mar '17

07 Mar '17

Hi i want install Siremis 4.3.0 on a Centos 7.3 with Php 7.1 when i want access to the web site, i have this error in httpd logs: Mon Mar 06 18:35:06.833569 2017] [php7:warn] [pid 22018] [client 172.16.1.150:58375] PHP Warning: require_once(SMARTY_CORE_DIRcore.assemble_plugin_filepath.php): failed to open stream: No such file or directory in /var/www/siremis-4.3.0/openbiz/others/Smarty/libs/Smarty.class.php on line 1368 [Mon Mar 06 18:35:06.833641 2017] [php7:error] [pid 22018] [client 172.16.1.150:58375] PHP Fatal error: require_once(): Failed opening required 'SMARTY_CORE_DIRcore.assemble_plugin_filepath.php' (include_path='.:/usr/share/pear:/usr/share/php:/var/www/siremis-4.3.0/openbiz/others/') in /var/www/siremis-4.3.0/openbiz/others/Smarty/libs/Smarty.class.php on line 1368 Anyone know this errors ? regards Lucas

2 1

Error in kamailio (Need for help)
by Ghanshyam Variya 06 Mar '17

06 Mar '17

Mar 2 11:55:33 Ubuntu kamailio: INFO: <core> [core/sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Mar 2 11:55:33 Ubuntu kamailio: INFO: <core> [core/tcp_main.c:4665]: init_tcp(): using epoll_lt as the io watch method (auto detected) Mar 2 11:55:33 Ubuntu kamailio[6479]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Mar 2 11:55:33 Ubuntu kamailio[6479]: INFO: rr [rr_mod.c:175]: mod_init(): outbound module not available Mar 2 11:55:33 Ubuntu kamailio[6479]: INFO: sipcapture [sipcapture.c:454]: parse_table_names(): INFO: table name:sip_capture Mar 2 11:55:33 Ubuntu kamailio[6479]: ERROR: <core> [db.c:203]: db_bind_mod(): Module db_mysql not found. Missing loadmodule? Mar 2 11:55:33 Ubuntu kamailio[6479]: ERROR: sipcapture [sipcapture.c:526]: capture_mode_set_params(): parsing capture_mode: could not bind db funcs for url:[mysql://kamailio:kamailiorw@localhost/kamailio] Mar 2 11:55:33 Ubuntu kamailio[6479]: ERROR: sipcapture [sipcapture.c:637]: capture_mode_init(): capture mode parsing failed Mar 2 11:55:33 Ubuntu kamailio[6479]: WARNING: sipcapture [sipcapture.c:769]: mod_init(): Default capture mode configuration failed. Suppose sip_capture calls will use other defined capture modes. Mar 2 11:55:33 Ubuntu kamailio[6479]: : <core> [core/mem/q_malloc.c:145]: qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten(0, 0)[0x7fea6ae64ae0:0x7fea6ae64b18]! Memory allocator was called from sipcapture: sipcapture.c:831. Fragment marked by sipcapture: sipcapture.c:788. Mar 2 11:55:33 Ubuntu kamailio: ERROR: <core> [core/daemonize.c:303]: daemonize(): Main process exited before writing to pipe This type of error are put's my kamailio so can you tell me why ? Thank you, Ghanshyam variya

3 2

Re: [SR-Users] Unknown $dlg(h_entry) and $dlg(h_id) in dialog
by Никитенко Виталий 06 Mar '17

06 Mar '17

Hi, very thanks! Before use dlg_get i lost 3-4 hour calls every day. Now one day - fly is good! > Hello, > > can you try using dlg_get(...) before the xlog? > > Cheers, > Daniel > > On 27/02/2017 08:33, Никитенко Виталий wrote: >> Hi i have kamailio 4.2, dialog module with parametr >> modparam("dialog", "dlg_flag", 5) >> modparam("dialog", "enable_stats", 1) >> modparam("dialog", "profiles_with_value", "caller") >> modparam("dialog", "dlg_match_mode", 1) >> modparam("dialog", "default_timeout", 3600) >> modparam("dialog", "timeout_noreset", 1) >> modparam("dialog", "send_bye", 1) >> modparam("dialog", "timeout_avp", "$avp(i:4242)") >> modparam("dialog", "db_url", DBURL) >> modparam("dialog", "db_mode", 0) >> >> in event_route[tm:local-request] i use >> if(is_known_dlg()) { >> xlog("L_INFO","DIALOG IS KNOWN $dlg(h_entry)::$dlg(h_id)) I >> } >> >> And when call drop by timeout, sometimes in log i see >> /usr/local/sbin/kamailio[21144]: INFO: <script>: DIALOG IS KNOWN <null>::<null> >> >> But usually this string in log file i see as "DIALOG IS KNOWN 3452::3334" >> >> Why it may be? when the $dlg(h_entry)::$dlg(h_id) is not known, I can not track $DLG_lifetime. >> Different between unsuccessful and successful call i dont see >> >> _______________________________________________ >> 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 > > -- > Daniel-Constantin Mierla > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > ------------------------------

2 1

Topoh module & call leg/transaction does not exist
by Olli Attila 06 Mar '17

06 Mar '17

Hello, I recently started testing topology hiding module on Kamailio and encountered a problem where B party hangs up, Kamailio can't identify the ongoing transaction to which the BYE should be related to and instead of tearing the call down Kamailio just sends back "481 Call leg/Transaction does not exist" to B party. On Kamailio config I have only enabled the topoh.so -module with default options and configured this to enable masking: modparam("topoh", "mask_callid", 1) Here is what the INVITE to B party looks like and the BYE message coming from the B party: http://pastebin.com/raw/f63zUmid When debugging on Kamailio end i get: http://pastebin.com/raw/ZefGAz30 Is this a Kamailio issue or a softphone issue on B party end? Cheers, --Olli

3 6

kamailio re.subst question.
by José Seabra 02 Mar '17

02 Mar '17

Hello There, I'm trying using the following regexp on re.subst function: (.*)Calling-Name: ?([\W|0-9A-Za-z_]{5,500} ?[<>:@.|0-9A-Za-z_]{5,500})(.*)$ And the string used for match is the following: #015#012Calling-Name-Status: available#015#012Calling-Name: "josé" < sip:5555555552@10.10.10.10>#015#012Presentation-Indicator: allowed#015#012 The Kamailio re.subst function is constructed as the following: $sht(cnam=>$ci::calling_name)=$(var(rb){re.subst,/(.*)Calling-Name: ?([\W|0-9A-Za-z_]{5,500} ?[<>:@.|0-9A-Za-z_]{5,500})(.*)$/\2/s}); The issue here is that this re.subst function doesn't match the accents, I have tested this regex on regex101.com and it does match. My question is how i can get the same result with re.subst? Anyone can help? Thank you Regards -- José Seabra

2 5

Dimensioning of shared memory for carrierroute
by Igor Potjevlesch 02 Mar '17

02 Mar '17

Hello, I'd like to dimension shared memory to use "kamctl cr reload" without issue. Currently, my table "carrierroute" has 204k entries and if I do "kamctl cr reload", the reload fails. Just after, I have thousands of these errors: Mar 2 12:10:52 /usr/local/sbin/kamailio[28606]: ERROR: <core> [sip_msg_clone.c:507]: sip_msg_shm_clone(): cannot allocate memory Mar 2 12:10:52 /usr/local/sbin/kamailio[28606]: ERROR: tm [t_lookup.c:1332]: new_t(): ERROR: new_t: out of mem: Mar 2 12:10:52 /usr/local/sbin/kamailio[28606]: ERROR: tm [t_lookup.c:1472]: t_newtran(): ERROR: t_newtran: new_t failed After a reboot, Kamailio back to normal. Actually, I start Kamailio with -m 704 and -M 128. Thank you for your advises. Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 2

Help with KEEP-ALIVE method
by Sergey Basov 02 Mar '17

02 Mar '17

Hi All. Today I have problem with connection from 1 of the clients. Their PBX sends KEEP-ALIVE after some time after REGISTER. I have next error in kamailio log Feb 28 14:26:19 sbc2 /usr/sbin/kamailio[3657]: ERROR: <core> [tcp_read.c:1354]: tcp_read_req(): bad request, state=7, error=4 buf:#012KEEP-ALIVE sip:1.2.3.4 SIP/2.0#015#012#015#012#012parsed :#012KEEP-ALIVE sip:1.2.3.4 SIP/2.0 After which my kamailio closes TCP session.... This connection is done with sip TLS. Thank you in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

3 4

Lost of connection with MySQL?
by Igor Potjevlesch 02 Mar '17

02 Mar '17

Hello, Two times in the last week, Kamailio reports MySQL errors like these: /usr/local/sbin/kamailio[19225]: ERROR: db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Can't connect to MySQL server on '192.168.2.6' (4) (2003) /usr/local/sbin/kamailio[19225]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query /usr/local/sbin/kamailio[19225]: ERROR: acc [acc.c:504]: acc_db_request(): failed to insert into database These errors are repeated many times and cause Kamailio to hang and don't reply to SIP requests. A reboot of Kamailio solves the issue. Anyone has already had this issue and had solved it? Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

3 17

CentOS 6 RPMs don't provide utils.so nor http_client.so
by Trevor Peirce 01 Mar '17

01 Mar '17

Hello, Is there any reason the CentOS 6 RPMs don't build utils.so or http_client.so? Looking back to 4.3 it appears it was being built: kamailio-utils-4.3.6-1.1.x86_64.rpm However it is no longer available in 4.4 nor 5.0 RPMs for either CentOS 6 or 7. Thanks, -- Trevor Peirce AcroVoice Solutions Inc

2 5

Update Kamailio tables
by Andy Chen 01 Mar '17

01 Mar '17

Hi all, What is the quickest way to update Kamailio tables? I wanted to check out the SIPTRACE module and I noticed the table is not there. Thanks. -- Andy Chen

2 2

Kamailio -can we change 478 to 503 when dns fails?
by José Seabra 01 Mar '17

01 Mar '17

Hello there, Some times dns server fails and my kamailio server start replying "Unresolvable destination (478/SL)". The propose of my email is to know if is there any possibility to change this reply (478) to for example one of 5XX replies. I have modparam("sl", "default_code", 503) but in situations where dns fails, kamailio doesn't reply 503. Thank you for your support Regards José Seabra

2 4

Figuring out why topos stops script from working
by Trevor Peirce 01 Mar '17

01 Mar '17

Hello, I'm having some trouble with topos not working well when mixed with nathelper (and maybe other cases, too). There is a bug that describes exactly the problem at https://github.com/kamailio/kamailio/issues/1005 Has anyone else had trouble with this module? Anything I'm missing? Thanks, -- Trevor Peirce AcroVoice Solutions Inc

3 2

Unknown $dlg(h_entry) and $dlg(h_id) in dialog
by Никитенко Виталий 01 Mar '17

01 Mar '17

Hi i have kamailio 4.2, dialog module with parametr modparam("dialog", "dlg_flag", 5) modparam("dialog", "enable_stats", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "timeout_noreset", 1) modparam("dialog", "send_bye", 1) modparam("dialog", "timeout_avp", "$avp(i:4242)") modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 0) in event_route[tm:local-request] i use if(is_known_dlg()) { xlog("L_INFO","DIALOG IS KNOWN $dlg(h_entry)::$dlg(h_id)) I } And when call drop by timeout, sometimes in log i see /usr/local/sbin/kamailio[21144]: INFO: <script>: DIALOG IS KNOWN <null>::<null> But usually this string in log file i see as "DIALOG IS KNOWN 3452::3334" Why it may be? when the $dlg(h_entry)::$dlg(h_id) is not known, I can not track $DLG_lifetime. Different between unsuccessful and successful call i dont see

2 1

Scaling websockets over SIP
by Jade SZ 01 Mar '17

01 Mar '17

Hi Guys, I am using kamailio/rtpengine and webrtc over sip client for a/v calls and all works fine. Now I want to scale it to 2 or more than 2 kamailio servers. As per search on mailing list I could: 1) Websocket Outbound/Edge Proxy 2) Each UA client to register with its Regards, Jade SZ

1 2

pushing alarms/counters other than with SNMPstats
by Valentin Christoph 28 Feb '17

28 Feb '17

Hi all I'm looking for ways to push stats/alarms from kamailio to some external destination other than using the SNMPstats module. Any ideas? Thanks Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

5 5

sip over websockets sip uri
by Jade SZ 28 Feb '17

28 Feb '17

Hi Team, Question: Just wanted to clarify regarding SIP URI in webrtc (over sip) connection. e.g. if we have a scenario where Kamailio is hosted with websocket support. Websocket URI is used to send packets to to wss/ws address and the SIP URI goes with it. I have tested it with any SIP URI including any domain and it works if I authenticate that domain, regardless that domain is real or not. // client side SIP URI config var configuration = { uri : 'sip:alice@example.com', }; Can we use any central domain for sip uri similar to realm concept and send traffic to multiple kamailio servers through websocket LB. dummy.domain - for ws/wss SIP URI kamailio-a.domain.com kamailio-b.domain.com Thanks in advance. Regards, Jade

2 4

Newbie starting - looking for docs / tutorials
by Antony Stone 28 Feb '17

28 Feb '17

Hi. I'm just starting with Kamailio - I've been to a few presentations about it, at Fosdem and KamailioWorld, but never used it so far. I'm pretty familiar with Asterisk, and I want to use them in combination. My first use case is to handle SIP registrations (both as a client, registering to a remote SIP server, and as a server, accepting registrations from handsets) in Kamailio instead of Asterisk, and still be able to route calls to/from the registered accounts through the Asterisk dialplan. I suspect this is a pretty common starting point for people who've used Asterisk and want to add Kamailio to the setup? So far I've been unsuccessful finding any Howto or Tutorial explaining how to do this - I can't help feeling I must simply be looking in the wrong place or in the wrong way. Searching for topics like "Asterisk Kamailio Getting Started" I find: http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf but with 12 chapters and 85 pages I'm not sure which bits are the essential parts to do what I need. I've also found http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x- asterisk-11.3.0-astdb but this seems to focus on a database backend, which I guess I'll want to use in due course, but doesn't seem the simplest place to start just to learn how Asterisk & Kamailio work together? So, please can someone suggest a beginner's guide to introducing Kamailio into an Asterisk setup, keeping things as simple as possible to start with, so I can learn about Kamailio one bit at a time (since there do seem to be a lot of bits to it) from a familiar environment of using it with Asterisk? Thanks in advance, Antony. -- What do you get when you cross a joke with a rhetorical question? Please reply to the list; please *don't* CC me.

2 3

multi domain kamailio
by przeqpiciel 28 Feb '17

28 Feb '17

Let's suppose that i have two machines with installed asterisk and one with kamailio. I would like have routing based on sip domain. for domain sip.domain1.com send sip signalling to asterisk#1 server and for sip.domain2.com send to asterisk#2 server. What i should use? Domain module or what ? Tomasz

3 4

Kamailio v5.0.0 Released - new major version is out
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Kamailio v5.0.0 is out – it comes with 6 new modules and a consistent set of improvements touching more than 50 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-0-0-release-notes/ Many thanks to all developers and community members that made possible this release. A consistent effort was directed to restructuring the source code tree to meet more modern patterns; introducing the KEMI framework that allows writing SIP routing blocks in other embedded interpreters such as Lua, JavaScript or Python; removal of obsoleted MI control mechanism, being replaced by RPC. Enjoy Kamailio v5.0.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference 2017 - http://www.kamailioworld.com

1 0

problem with avpops and mongodb modules
by hdssdsdsdsfsdf hdssdsdsdsfsdf 27 Feb '17

27 Feb '17

1 1

Releasing Kamailio v5.0.0
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Hello, I am considering to release Kamailio v5.0.0 at the beginning of next week, depending how things evolve in the next days, either on Monday, Feb 27, or Tuesday, Feb 28. I haven't noticed any new issues specific for the new features added for 5.0 lately, so it looks pretty good so far. If you are aware of issues not reported to the bug tracker, register them as soon as possible at: - https://github.com/kamailio/kamailio/issues There are still some parts of documentation that have to be updated, I am planning to work on them during the next days, but help from others are very appreciated, especially details of what you had to do changes while upgrading from 4.4 to 5.0. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Parallel Forking with Different Call-Id
by SamyGo 24 Feb '17

24 Feb '17

Hi, I've a very strange scenario to work on which requires me to parallel fork the call to the same Destination provider. The only problem here is that they think that the second INVITE with different branch tag is a re-transmission and hence only take one call forward. I do not have to modify any R-URI or any headers, hence just 'append_branch() before t_relay() is in the code. ... $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); append_branch(); route(RELAY); ... *Question:* Is there anyway possible I can change the CallID of the forked INVITEs ? I have tried using Topoh module, but it still puts the same CallID before sending out. Different from the A-leg but the Sent out INVITEs have same Call-ID value. In other weird scenario, I've also tried branching, and looping call within Kamailio before sending out to carrier hoping that Kamailio would treat the two different calls and Topoh would change the Call-ID on both INVITEs before sending out.. ... if(is_present_hf("X-FORKED")) { $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); route(RELAY); }else { append_hf("X-FORKED: 1\r\n"); append_branch(); route(RELAY); } ... Still I get same Call-ID on outgoing branched call. I could branch out one INVITE to a MediaServer, say FreeSWITCH/Asterisk but again I don't want to have that component bottleneck the throughput. That could be my very last option. Looking for some ideas. Regards, Sammy

3 6

Kamailio 4.3.x location_attrs update issue
by Mridul B 24 Feb '17

24 Feb '17

Hi, We've noticed a case where the *location_attrs* table isn't updated when there's an unexpected shutdown of a SIP device (here there'll be no REGISTER message sent with expires=0). We're using Kamailio v4.3.x When the SIP device shuts down ungracefully. The location table is updated using DELETE query This is done by the timer which deletes the expired contacts, by default the timer_interval is set as 60. Here, the logs of auto-removal of expired contacts entries by the timer > Feb 17 17:16:13 localhost kamailio[23273]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:14' AND expires<>'1970-01-01 00:00:00'] > Feb 17 17:16:14 localhost kamailio[23272]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:15' AND expires<>'1970-01-01 00:00:00'] The delete operation is only on location table, shouldn't we have the same implementation on the location_attrs table based upon ruid or username? We're getting duplicate entries in the location_attrs table of the same sip user after it registers. But when the SIP device un-registers, the entries in the location as well as in the location_attrs table is deleted Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: usrloc [ucontact.c:1725]: > uldb_delete_attrs(): trying to delete location attributes > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location_attrs where username='mridul' > AND ruid='uloc-58a72243-5ace-2'] > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location where username='mridul' AND > contact='sip:mridul@127.0.0.1:37881;rinstance=b841a8c916e47dfa'] I think the usrloc module should have the implementation of deleting the entries of expired sip users based on its ruid/username in the location_attrs table also. Thanks for the help. Best regards, MRIDUL

2 3

want to see active users SIP users
by Mandar Nagarkar 24 Feb '17

24 Feb '17

Hello Everyone, I want to see active users SIP users for this purpose I checked the location table of kamailio database it does have active users entries however as per my observation there is 45 secs to 1 min delay before the entry actually added to this table i.e. suppose if I am registering the user through pjspi,jssip or any client I cannot see entry for that user in location table. I have to wait up-to 1 min to see the user in location table. Is there any way we can Identify if user is online without delayed entry? Regards, Mandar nagarkar The information in this email is confidential and may be legally privileged. It is intended solely for the addressee and access to it by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken based on it, is strictly prohibited and may be unlawful.

2 1

Fetching proxy-autheticate header from 407 Response
by vivek＠advaitamtech.com 23 Feb '17

23 Feb '17

Hi All, Am not using the authentication in the kamailio. So I will be receiving the "407 Proxy Authentication Required" response from the third party server for the first INVITE. I want to fetch the Proxy-Authenticate header present in the 407 response. I browsed and tried using the pseudo-variables "$adu/$aa/$ar". It dint work. Please let me know if there is a way to fetch this header. Thanks, Vivek.

3 4

How to release call without having to use fromTag and toTag, and resume a suspended call
by Cassandra Yurekli 23 Feb '17

23 Feb '17

To whom it may concern, There are several points I don’t quite understand regarding a couple of kamailio modules. I have a main application which controls calls by sending command messages via a socket (for example reject this sip invites). I communicate with the socket client via evapi module and using evapi_asynch_relay() in order to be sure that communication are suspended and kamailio waiting an order for doing something with them (Accepte, reject, release, ...). I want to release a call, using a command “release this call”. I’m actually doing this with dialog module (using get_dialog(callid,fromTag,toTag) and dlg_bye(‘all’)) but is there anything else (like another module) which could release a call without necessarily having to use fromTag and toTag to get the context ? I didn’t find it on modules documentations. Also after an INVITE I’m waiting for the main app to accept or reject the call. For accepting and continuing the transaction I found “t_continue(tindex,tlabel,route)” on tmx module, but at this moment $T(id_index) and $T(id_label) are null, is it normal or I did something wrong ? How can I simply resume the suspended call ? Thanks; Best regards,

2 1

Need help with tcpops
by Abdoul Osséni 22 Feb '17

22 Feb '17

Hi, My call flow is: Softphone --> TLS ---> Kamailio --> UDP --> ASTERISK --> PSTN I want to use tcpops module between the softphone and kamailio. http://www.kamailio.org/docs/modules/4.4.x/modules/tcpops - enable tcpops : no problem - disable tcpops : two cases - cancel or bye from softphone: no problem. - cancel or bye to softphone via kamailio: how disable tcpops? I can't use $avp(bye_conid) because it is asterisk thant sending the sip message. Can we use $avp(caller_conid)? In my case, the softphone is always at the origin of the call establishment. Regards Abdoul.

2 1

Automated integration tests for individual modules
by A Lai 22 Feb '17

22 Feb '17

Hi, I just started working with the Kamailio code base. I am wondering if there are some common strategies in having automated tests on proper functioning of modules when loaded into cfg.

2 1

problem with avpops and mongodb modules
by hsdsdssdfsdf dsasd 22 Feb '17

22 Feb '17

2 1

How to handle CANCEL if corresponding INVITE transaction is missing
by Venkatesh Macha 22 Feb '17

22 Feb '17

Hi all, I am using Kamailio 4.4.5, It is working fine. But in one case. If i make a video call between the two users and cancels the call then Call is not disconnecting on the callee side. Please look the following scenario. 1000 made one video call to 2000 2000 is ringing. Now 1000 disconnected the call by sending CANCEL. But 2000 is not disconnecting. This is happening because Kamailio is unable to find the corresponding INVITE transaction for the CANCEL so it is simply dropping the requests. Why Kamailio unable to find the transaction. And is there any possibility to handle this case and send the CANCEL to callee side. I tried with forwad() and someother things but it is just generating the loops. I am also attaching the ngrep trace. Thank you in Advance. CANCEL_not_working_nabble.CANCEL_not_working_nabble <http://sip-router.1086192.n5.nabble.com/file/n156243/CANCEL_not_working_nab…> -- View this message in context: http://sip-router.1086192.n5.nabble.com/How-to-handle-CANCEL-if-correspondi… Sent from the Users mailing list archive at Nabble.com.

2 2

kamailio memory leak in private memory
by david 22 Feb '17

22 Feb '17

Hello all We are observing an issue in one kamailio proxy we use. The consecuence of an unknow cause is that the private memory of some UDP receiver processes in where we listen from a particular IP, seems to be draining along the time while processing traffic. Quite "slowly" but maybe aroung 200K in 3-4hours, depending on the moment of the day. This is quite weird for us, because it seemed like it was caused by some particular traffic profile, since we have several IP listening (8 children each) and we only see the issue in the UDP receiver processes of some of those IPs. Besides we have moved the kamailio service listening on the same IP to several different physical servers, and the behaviour is the same. We look at it ussing pkg.stats kamcmd command and for instance if we track one of the process affected pid: 23784 rank: 53 used: 2838616 free: 13005408 real_used: 3771808 pid: 23784 rank: 53 used: 2842584 free: 12999280 real_used: 3777936 This 2 outputs were taken with 10min diff between each other. But we dont know where or how to look at it. We have been these last days trying to search for patterns to allocate some particular traffic source causing something weird, but no luck. We have checked all the SIP INVITE messages looking for anything weird at any field but no luck, besides we sometimes see some bad headers with blank spaces in the uri or thing like that but those cases we have an error and we dont think it can cause a memory leak. Shared memory seems to be stable. We also have checked the config we are using and we have been discarding possible issues since we dont think the $var variables we are using could cause that issue (but you never know....) We tried to debug the kamailio on one affected IP compiling it with DBG_SYS_MALLOC , and doing the dump of the pkg mem status when the kamailio was restarted and no traffic was processed, and also when it had traffic processed. in this last case, we removed traffic, waited for 20min and made the dump i attach you both files, we have the debug logs, but they are very big to be passed here, and we dont know what to look for :) could you please give us some guidance or clue to at least start to searching for something that can cause this issue? thanks alot and regards david

3 6

parse_addr_spec(): unexpected char [^M] in status 6:
by Abdoul Osséni 21 Feb '17

21 Feb '17

Hello, Could you help us to fix this error? <core> [parser/parse_addr_spec.c:662]: parse_addr_spec(): unexpected char [^M] in status 6: Regards Abdoul.

3 5

Transaction-persistent data and uac_req_send()
by Alex Balashov 21 Feb '17

21 Feb '17

Hello, I am using uac_req_send() to send INVITEs to a redirect server. However, upon receiving the redirect, I need access to some state information that was last available at the time of sending the request. Does uac_req_send() allow me to persist AVPs like a normal TM transaction? If not, what are my options, other than building my own state using htable or what have you? I would really rather not maintain—and be responsible for garbage-collecting—externally constructed state like that if possible. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 10

Kamailio 4.4 with 2 Asterisk 13
by Graham Brown 20 Feb '17

20 Feb '17

I'm trying to load balance 2 Asterisk servers with Kamailio offloading the registration to Kamailio using the Asterisk real-time database tables. I have a problem with the dispatcher module, if there is a t_on_failure route it always goes into that, if I remove that then I would expect every call to round robin but I'm only ever hitting the same server from the same extension (seems like sticky session based on source ip). 1. how do I fix my failure route so I can reboot server 1 and server 2 will be used 2. how to I change the sticky session behaviour. Another issue I am facing with extensions registering to Asterisk are only avaliable on the server they routed to My assumption was if using the Asterisk real-time database I would get the contact on both servers. Maybe my architecture is incorrect, I'm open to suggestions. Currently Kamailio is not storing locations (assuming that is the extension contact) maybe that is a better solution, but I dont know how to configure Asterisk to forward everything to Kamailio then or integrate via the database or something. Thanks in advance for any help Regards Graham

2 1

Cleaning wiki accounts
by Daniel-Constantin Mierla 20 Feb '17

20 Feb '17

Hello, I noticed a lot of random usernames for the accounts created to edit the wiki. Apart of several which did some which did some edits to add spam, most of them are inactive. Not sure what would be the real purpose behind them, but I am going to remove anything that doesn't look as a community user. Shall I remove an account with random id, but for a good user, let me know, it can be readded/account recreated. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

memory: Free fragment not found
by Dmitri Savolainen 17 Feb '17

17 Feb '17

Hi. My kamailio v4.4.5 installation ( -m 512 -M 64 ) stop sip processing with repeatable error: Feb 15 21:45:05 kamailio[14173] ERROR: <core> [mem/f_malloc.c:415]: > fm_search_defrag(): fm_search_defrag(0x7f39917fa010, 952); Free fragment > not found! > Feb 15 21:45:05 kamailio[14173] ERROR: <core> [mem/f_malloc.c:498]: > fm_malloc(): fm_malloc(0x7f39917fa010, 952) called from core: rvalue.c: > rval_new_empty(224), module: > core; Free fragment not foun while it was enough free shared memory according to kamcmd> core.shmmem m > { > total: 512 > free: 472 > used: 27 > real_used: 39 > max_used: 51m > fragments: 37035 > } whether it was about package memory or smth else? Thanks! -- Savolainen Dmitri

2 4

MSILO offline messages are coming in reverse order.
by Venkatesh Macha 17 Feb '17

17 Feb '17

Hi all, I am using MSILO module for the offline messaging feature. It is working good. But If I send more than one message to the offline user. Those messages are delivering in the reverse order. Please consider the following scenario. A sending B ( B is offline, so messages are stored in DB) hi hello how are you Now B comes online and registers. Kamailio sending the messages but App is getting them in reverse order. how are you hello hi Is there any possibility to tell Kamailio to send them in order. ? Thank you in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/MSILO-offline-messages-are-coming-i… Sent from the Users mailing list archive at Nabble.com.

2 2

Problem with dispatcher on Kamailio 4.2 and Debian 8
by przeqpiciel 17 Feb '17

17 Feb '17

Hi, I'm trying to setup dispatcher module to working with mysql database but when i put to kamailio.cfg below lines then kamailio wont starts. loadmodule "dispatcher.so" modparam("dispatcher", "db_url", "mysql://kamailioro:kamailioro@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") i tried connect to mysql with credentials which you can see in mod_param line and i could query all kamailio database. im really new in this and i dont know how to debug this, is anybody could help me with it ? Best regards, Tomasz

2 4

Kamailio 4.2 on Debian 8 and dispatcher problem
by przeqpiciel 16 Feb '17

16 Feb '17

Hi, I'm trying to setup dispatcher module to working with mysql database but when i put to kamailio.cfg below lines then kamailio wont starts. loadmodule "dispatcher.so" modparam("dispatcher", "db_url", "mysql://kamailioro:kamailioro@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") i tried connect to mysql with credentials which you can see in mod_param line and i could query all kamailio database. im really new in this and i dont know how to debug this, is anybody could help me with it ? Best regards, Tomasz

2 1

Re: [SR-Users] Timejump causes kamailio to hang/stall
by erik 16 Feb '17

16 Feb '17

Thanks Daniel, I will look into that. Indeed I found the logic that corrects the time ticks for any drifting, using the time of day. I guess that causes the registrations to expire. I am out of the office for a couple of days, but will get back to you. Best Regards, Erik Hello, On 15/02/2017 14:47, Erik Visser wrote: > After I have kamailio up and running, with UAs properly registrered and sessions routed successfully, my platform faces a timejump. > The platform starts up with date&time set to 1-1-2000 and after kamailio is up and running, the date&time jumps to the current date&time, causing two effects: > - I loose all my registrations (since they expired in 3600 seconds) > - kamailio does not respond to any SIP messages for a couple of minutes > > Is there a way to have kamailio use timeticks rather than relative time for the registrations? > Does anybody recognize the hanging behavior? Any suggestions? > kamailio is using unixtime for registrations, not relative time. But the timer callbacks are using relative intervals to know when to be executed (e.g., timer callbacks are defined to be executed like very 1 minute, or every 90 seconds). It may be that some of the timers try to catch up on missed intervals. It is strange that you say no sip traffic is processed. Can you simulate again the situation and get the backtrace with gdb from few sip workers? If your traffic is over udp, then do 'kamctl ps', get the pid of processes with index 1, 2 and 3, then use gdb to attach to those PIDs and grab the output of: bt full info locals list Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

remove offline message prefix of Offline message of MSILO module
by Venkatesh Macha 16 Feb '17

16 Feb '17

Hi all, I am implementing Offline messages functionality using the MSILO module. Everything went well. I am able to store and send those offline messages when the user comes online. But my present problem is, I don't want the offline message prefix. Please look at the following example. This is what Kamailio sending. [Offline message - Wed Feb 15 05:43:38 2017] Test2 I want only Test2 Is there any possibility to remove that offline message prefix ? Thank you in advance ! -- View this message in context: http://sip-router.1086192.n5.nabble.com/remove-offline-message-prefix-of-Of… Sent from the Users mailing list archive at Nabble.com.

2 2

Timejump causes kamailio to hang/stall
by Erik Visser 16 Feb '17

16 Feb '17

After I have kamailio up and running, with UAs properly registrered and sessions routed successfully, my platform faces a timejump. The platform starts up with date&time set to 1-1-2000 and after kamailio is up and running, the date&time jumps to the current date&time, causing two effects: - I loose all my registrations (since they expired in 3600 seconds) - kamailio does not respond to any SIP messages for a couple of minutes Is there a way to have kamailio use timeticks rather than relative time for the registrations? Does anybody recognize the hanging behavior? Any suggestions? Thanks! Erik ____________________ Erik Visser erik(a)vissershut.nl

2 1

IPv6 addresses in dispatcher.list file
by Joel Serrano | VOZELIA 16 Feb '17

16 Feb '17

Hi, What is the correct format for the dispatcher.list file using IPv6? I have the following file: $ cat /etc/kamailio/dispatcher.list # $Id$ # dispatcher destination sets # # line format # setit(int) destination(sip uri) flags(int,opt) priority(int,opt) attributes(str,opt) # proxies 4 sip:XX.XX.XX.XX:5060 4 sip:ZZ.ZZ.ZZ.ZZ:5060 6 sip:[X:X:X:X::X:1]:5060 6 sip:[Z:Z:Z:Z::Z:2]:5060 $ For some reason, Kamailio doesn't recognize the v6 endpoints: kamcmd> dispatcher.list { NRSETS: 1 RECORDS: { SET: { ID: 4 TARGETS: { DEST: { URI: sip:XX.XX.XX.XX:5060 FLAGS: AP PRIORITY: 0 } DEST: { URI: sip:ZZ.ZZ.ZZ.ZZ:5060 FLAGS: AP PRIORITY: 0 } } } } } kamcmd> I have tried multiple combinations for the v6 format but I cannot get one to work. What am I doing wrong? Thanks, Joel. Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado.Les informamos que los datos contenidos en el presente mail, han sido facilitados por usted o recogidos de Fuentes Accesibles al Público. Dicha información personal, será incluida en un fichero bajo la responsabilidad de VOZELIA TELECOM S.L.Todos los ficheros que trata nuestra empresa se encuentran debidamente inscritos ante la Agencia Española de Protección de Datos.Vd. Como propio interesado podrá ejercitar sus derechos de Acceso, Cancelación, Rectificación u Oposición, en el mail: administracion(a)vozelia.com o en la dirección Avenida de Manoteras 8 escalera 3 2ºB, 28050 Madrid.Tal y como establece la Ley Orgánica de Protección de Datos 15/1999 de 13 de Diciembre.Para ejercer sus derechos desde Panama podrá hacerlo en la dirección Vía Argentina, Edificio Mediteranean Loft. Panama.En caso de no ser el destinatario y haya recibido este mensaje por error, agradeceremos que nos lo comunique inmediatamente al remitente sin difundir, almacenar o copiar su contenido.La divulgación o el suministro, en todo o en parte, a cualquier tercero, no podrá ser realizada sin el previo, expreso y escrito consentimiento de VOZELIA TELECOM S.L. .

2 1

record_route() function
by Andy Chen 15 Feb '17

15 Feb '17

Hi all, I need some advise on how to use the record_route() function. So I make that call when the first method comes in and there are two Record-Route headers created. However, one of of the headers I want to add transport=tcp. What is the best way to do this? I've tried adding that string per documentation within the () and added it to both. Thanks. -- Andy Chen

3 10

Branch 5.0 created
by Daniel-Constantin Mierla 15 Feb '17

15 Feb '17

Hello, the GIT branch 5.0 has just been created, it will host the release series 5.0.x. To get this branch from GIT, you can use: git clone https://github.com/kamailio/kamailio.git kamailio cd kamailio git checkout -b 5.0 origin/5.0 Hopefully before the end of the month the full release of 5.0.0 will be out. >From now on, any corresponding fix has to be pushed first to master branch and then cherry-picked to branch 5.0. No new features can get in branch 5.0. Enhancements to documentation or helping tools are still allowed. If you are not sure about doing or not a backport, ask on sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

4 9

Dispatcher and "src based" routing algorithm
by Olli Attila 15 Feb '17

15 Feb '17

Hello, I am using kamailio 4.4.3 (x86_64/linux) 5a2195 and Dispatcher module to route calls towards PSTN (4 pstn gateways). The Dispatcher module is using round robin mechanism to select the pstn gateway. I have built a routing logic in a way that when call arrives to Kamailio from one of the PSTN gateways, and fails to route towards an external sip target, the call is marked as CFU call (a diverting call) and routed back to PSTN with different B number, but the actual challenge is that I would like the target PSTN gateway _not_ to be the same PSTN gw from where the call originally came in to Kamailio. This feature is requested because of the technical limitations of the pstn gateway software. So is there any mechanism in Dispatcher or elsewhere in Kamailio to do this? Basically I would need a flag or something to mark the pstn gateway when the call comes in and assure that when routing back to pstn, the same gateway would not be used at all, or it should be used as last target from the four gateways in Dispatcher. Cheers, -- Olli

1 0

Refreshing: info about private emails
by Daniel-Constantin Mierla 15 Feb '17

15 Feb '17

As I still receive a lot, I am re-posting the notification about private emails, either initiating a topic or sent directly after a discussion started on public mailing lists (subject with mailing lists prefix, but the sender header is not the mailing list daemon). I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is highly not recommended. The rule is also suggested in the mailing lists presentation page: - http://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed in decent time and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. Therefore if you wrote me and expect a response, re-send the message to the appropriate public mailing list. Checking and answering the messages on Kamailio mailing lists are among the top of priorities on my daily activity, so it is high chance you get the response faster when you write there. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. There are several common sense as well as technical reasons for keeping the discussion on public mailing lists. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine. Also, other people subscribed to the list can learn from conversation or contribute to the discussion, resulting in better outcome - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. They are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else if you don't get this instantly, or this application is missing what so ever tutorial giving exactly the solution you need, or simply asking for full config of complex requirements, is not helping at all, but actually making people to ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, have appropriate expectation and behavior. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

TOPOH MODULE
by Annus Fictus 14 Feb '17

14 Feb '17

Hello, I'm using TOPOH module but I have a problem with Call-ID. If I use: modparam("topoh", "mask_callid", 1) and send the INVITE to a media server, when media server send BYE to KAMAILIO, Kamailio answer like transaction not exist. In the parameter description, I read i have to use "lreq_callee_headers" dialog parameter to resolve this kind of issues, but I can't find this parameter on the module description. I'm using Kamailio 4.4.5 Thank's Regards

3 3

Remove
by Ryan Holbein 13 Feb '17

13 Feb '17

Can you please remove me

2 1

SIP-T conversion to SIP
by Martin Quevedo 13 Feb '17

13 Feb '17

hi all I need to interact with an MSC that talks SIP-T that is the MSC embeds ISUP in the SIP messages IAM, ANM, ACM and REL messages specifically but I have a SIP client which does not speak SIP-T is it possible that kamailio parses invites with IAM ISUP parameters like forwarding number and forwarding reason for incoming messages and embeds ACM, ANM and REL isup messages in OK, Session Progress and BYE regards, Martin ?

1 0

VUC in 2 hours - Allison Smith and Kamailio World
by Daniel-Constantin Mierla 10 Feb '17

10 Feb '17

Hello, short note for the people that might be interested on the topic - Allison Smith (aka, the voice of Asterisk) will be on VoIP Users Conference later today, in about 2 hours, among the topics of discussion being the Kamailio World 2017. I will also join, trying to squeeze some updates about upcoming Kamailio 5.0. More details about the online audio/video conference can be found at: - https://www.vuc.me/2017/vuc632-get-ready-for-kamailio-world-2017-with-allis… The session will start in about 2 hours (17:00UTC), anyone can connect via webrtc capable browser, SIP or PSTN dial in. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

python_exec and parameters
by gmele 10 Feb '17

10 Feb '17

Hello, in our Kamailio, we are using python_exec to execute script used to send push notifications to Google or Apple. Currenty, the method we call only accepts the message as parameter. We then extract from the message the Call Id, to URI and from URI. Here is the python method signature: *def SendPushNotification(self, msg)* To extract the toUri and fromUri (and remove the sip:), we use python regex : *uriPattern=re.compile(".*sips?:(.+@[^>;]+)")* This is working well 99% of the time, except sometimes (we don't know why) where we get the error: *logger_LM_ERR(): Exception occurred: Traceback (most recent call last): File "mypythonExample.py", line XXX, in SendPushNotification requestUri = self.ExtractURIFromHeader(msg.getHeader("to")) File "mypythonExample.py", line XXX, in ExtractURIFromHeader m= self.uriPattern.match(header); TypeError: an integer is required* We tried to analyze the cause of this error, but we didn't find any problem with the message itself (the message is complete, with the *to *and *from *correctly filled). So, to avoid using the python regex, we have decided to pass parameters to the python script (callId, from and to URI): *def SendPushRequest(self, msg, fromUri, toUri, callid)* But it seems python_exec doesn't support more than 2 parameters ($mb and another one). Is this exact? If this is the case, how can we pass several parameters from kamailio.cfg? Thx for your help. Regards Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/python-exec-and-parameters-tp155803… Sent from the Users mailing list archive at Nabble.com.

3 8

Video and Slides: SIP and WebRTC Load Balancing and High Availability for FreeSWITCH, FOSDEM
by Giovanni Maruzzelli 10 Feb '17

10 Feb '17

Hello, here you can find both video and slides of my presentation at FOSDEM 2017, Bruxelles: https://fosdem.org/2017/schedule/event/freeswitch/ Enjoy! -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT Cell : +39-347-2665618

1 0

MSRP with Kamailio
by Olli Attila 10 Feb '17

10 Feb '17

Hello, I am running kamailio version 4.4.2. I just configured MSRP module to kamailio with example config taken from "http://www.kamailio.org/docs/modules/4.4.x/modules/msrp.html#msrp.usage" (Example 1.16. Event Route (using htable for MSRP connection tracking). After configuring msrp relay, the kamailio service has been restarted succesfully and no errors occurred. Blink is configured to use my server name with tls and port 5061 for MSRP. I get this error on Blink when i try to send files: https://p.dnaip.fi/X4vwoy40 Any suggestion what is going wrong? Cheers, -- Olli

3 12

Replace old registration record with new one while keeping single AOR per contact
by Aqs Younas 09 Feb '17

09 Feb '17

Greetings list, I want to keep single AOR per contact. If new regitration comes i want to replace the old record with new. Suppose User Alice registers from device A and after certain time it try to register from device B. I want Registration from device B to replace A. Any pointer is mush appreciated. Cheers, Aqs

3 6

Fwd: [sr-dev] git:master:c32a5f64: sipcapture: added $hep(src_ip) variable
by David Villasmil 09 Feb '17

09 Feb '17

Hello guys, i just installed kamailio from deb http://deb.kamailio.org/kamailio50-nightly jessie main deb-src http://deb.kamailio.org/kamailio50-nightly jessie main And i'm trying to use the newly exposed $hep(src_ip) by: if ( ds_is_from_list("10", "2", "sip:$hep(src_ip)" ) ) but i'm getting: ERROR: sipcapture [sipcapture.c:2763]: pv_parse_hep_name(): REQUEST, PRE, src_ip ERROR: sipcapture [sipcapture.c:2793]: pv_parse_hep_name(): unknown hep name src_ip ERROR: <core> [core/pvapi.c:877]: pv_parse_spec2(): pvar "hep" has an invalid name param [src_ip] ERROR: <core> [core/pvapi.c:1032]: pv_parse_spec2(): wrong char [)/41] in [$hep(src_ip)] at [11 (5)] ERROR: <core> [core/sr_module.c:1251]: fix_param(): bad PVE format: "sip:$hep(src_ip)" ERROR: <core> [core/mod_fix.c:539]: fixup_spve_null(): Cannot convert function parameter 1 to spve Any ideas? Thanks Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ---------- Forwarded message ---------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Fri, Dec 9, 2016 at 10:17 AM Subject: [sr-dev] git:master:c32a5f64: sipcapture: added $hep(src_ip) variable To: sr-dev(a)lists.sip-router.org Module: kamailio Branch: master Commit: c32a5f64de4ff731488d9f6bf64651117d26d46e URL: https://github.com/kamailio/kamailio/commit/ c32a5f64de4ff731488d9f6bf64651117d26d46e Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: 2016-12-09T10:16:31+01:00 sipcapture: added $hep(src_ip) variable - reworked patch from GH #879 --- Modified: src/modules/sipcapture/sipcapture.c --- Diff: https://github.com/kamailio/kamailio/commit/ c32a5f64de4ff731488d9f6bf64651117d26d46e.diff Patch: https://github.com/kamailio/kamailio/commit/ c32a5f64de4ff731488d9f6bf64651117d26d46e.patch --- diff --git a/src/modules/sipcapture/sipcapture.c b/src/modules/sipcapture/ sipcapture.c index 117db74..b5bbe7b 100644 --- a/src/modules/sipcapture/sipcapture.c +++ b/src/modules/sipcapture/sipcapture.c @@ -2835,6 +2835,11 @@ static int pv_parse_hep_name (pv_spec_p sp, str *in) else goto error; } break; + case 6: + { + if(!strncmp(in->s, "src_ip", 6)) sp->pvp.pvn.u.isname.name.n = 2; + else goto error; + } case 7: { if(!strncmp(in->s, "version", 7)) sp->pvp.pvn.u.isname.name.n = 0; @@ -2857,17 +2862,24 @@ static int pv_parse_hep_name (pv_spec_p sp, str *in) static int pv_get_hep(struct sip_msg *msg, pv_param_t *param, pv_value_t *res) { + static char sc_buf_ip[IP_ADDR_MAX_STR_SIZE+12]; + int sc_buf_ip_len; + if(param==NULL) return -1; switch(param->pvn.u.isname.name.n) { case 0: - return pv_get_uintval(msg, param, res, hep_version(msg)); - case 1: - return pv_get_uintval(msg, param, res, hep_version(msg)); + return pv_get_uintval(msg, param, res, hep_version(msg)); + case 1: + return pv_get_uintval(msg, param, res, hep_version(msg)); + case 2: + sc_buf_ip_len=ip_addr2sbuf(&msg->rcv.src_ip, sc_buf_ip, + sizeof(sc_buf_ip)-1); + sc_buf_ip[sc_buf_ip_len]=0; + return pv_get_strlval(msg, param, res, sc_buf_ip, sc_buf_ip_len); default: return hepv3_get_chunk(msg, msg->buf, msg->len, param->pvn.u.isname.name.n, param, res); } return 0; } - _______________________________________________ sr-dev mailing list sr-dev(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev ᐧ

1 0

rtpengine key file
by Kjeld Flarup 09 Feb '17

09 Feb '17

Hello I have build rtpengine on a Raspberry PI 3 running Jessie, but it fails on startup pi@raspberrypi:/usr/local/src/rtpengine $ rtpengine --log-level=7 --log-stderr [1486630112.767604] CRIT: Fatal error: Bad command line: Key file does not start with a group Most likely I'm missing something which is found on regular PC Linux distributions. But what? Googling the error message reveals almost nothing and nothing related to rtpengine Any hints will be appreciated. -- Med venlig hilsen / Best regards Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef Viptel ApS, Hammershusvej 16C, DK-7400 Herning Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk

2 2

LCR module HowTo (Spanish)
by Campus VoToVoice 09 Feb '17

09 Feb '17

Hello, My small contribution to Kamailio's documentation: https://www.voztovoice.org/?q=node/1491 If you find any error, please write me. Regards

2 1

Branching v5.0
by Daniel-Constantin Mierla 09 Feb '17

09 Feb '17

Hello, following discussion on mailing lists and at Fosdem over the last weekend, I am considering to create the git branch 5.0 later this week, on Thursday, Feb 9, 2017. Afterwards, the master branch can take commits with new features. The branch 5.0 will be tested for 1-2 more weeks and if all is fine, the v5.0.0 can be released. Should anyone have other suggestions, jump into discussion. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

nathelper module fix_nated_register(); question
by JR Richardson 09 Feb '17

09 Feb '17

Hi All, Just confirming when using nathelper, the fix_nated_register function is only used to write corrected contact into to $avp then to database, this function does not actually change the contact in the SIP message when passing through the proxy, correct? fix_nated_contact is used to actually change the SIP message contact when passing through the proxy. I'm using kamailio 4.2. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

3 2

dlg_set_timeout bye to websockets client
by Switch168 09 Feb '17

09 Feb '17

Hello, Just wondering if it's possible to intercept the bye message that dlg_set_timeout sends out to do some nat_helper function to fix it so the bye can be sent to random.invalid uris. I know there is the edge proxy outbound module but wondering if I can do it without. Regular bye between 2 clients i was able intercept and fix with nat helper but I dont know how to fix the byes that is sent out from dlg_set_timeout function. Thanks in advance! Andrew

3 7

Re: [SR-Users] kamailio with cassandra
by Никитенко Виталий 09 Feb '17

09 Feb '17

Hi. >> Can you provide more details for what you need it,maybe we can provide some hints? I want to store the data of registered users (location) in the database cassandra (it would be great also dialog stored in cassandra). >>Can you provide access to a test system/vm where to try to fix it Yes, I can it, will prepare today the VM for access to public network and send to email data access. Thanks for the help. > Hello, > > unfortunately the module was not maintained for a while. Can you provide > access to a test system/vm where to try to fix it, because I don't use > it and have no platform where to test? > > There is another module, ndb_cassandra, depending of what you want to > do, it might help. Can you provide more details for what you need it, > maybe we can provide some hints? > > Cheers, > Daniel > > On 07/02/2017 10:55, Никитенко Виталий wrote: >> Hi! >> I want start kamailio with db_cassandra. do as it written in https://www.kamailio.org/docs/modules/4.2.x/modules/db_cassandra.html. Then trying kamctl start, failed with error "table version does not exist!". >> From CQLSH table is view >> cqlsh:kamailio> Describe tables; >> version location >> >> cqlsh:kamailio> select * from version; >> table_name | table_version >> ------------+--------------- >> version | 1 >> >> Tell me please wherein may be trouble? Full log below >> >> ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/location >> ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/version >> INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob >> INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available >> INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 >> INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled >> INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.allow >> INFO: permissions [permissions.c:608]: mod_init(): default allow file (/usr/local/etc/kamailio//permissions.allow) not found => empty rule set >> INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.deny >> INFO: permissions [permissions.c:617]: mod_init(): default deny file (/usr/local/etc/kamailio//permissions.deny) not found => empty rule set >> INFO: misc_radius [misc_radius.c:164]: mod_init(): initializing... >> ERROR: db_cassandra [dbcassa_base.cpp:449]: cassa_translate_query(): table version does not exist! >> ERROR: db_cassandra [dbcassa_base.cpp:962]: db_cassa_query(): Failed to query Cassandra cluster >> ERROR: <core> [db.c:401]: db_table_version(): error in db_query >> ERROR: <core> [db.c:440]: db_check_table_version(): querying version for table location >> ERROR: usrloc [dlist.c:640]: register_udomain(): error during table version check. >> ERROR: registrar [reg_mod.c:554]: domain_fixup(): failed to register domain >> ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:982 >> ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:987 >> ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:992 >> >> _______________________________________________ >> 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 > > -- > Daniel-Constantin Mierla > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > ------------------------------

1 0

Change Priority of Dispatcher
by Rodrigo Moreira 08 Feb '17

08 Feb '17

Hello, Help me with a question. Is it possible to change the prohirity of a Kamailio Dispatcher target at run time, without changing the destination entry in the database? If so, which module should I use? Have you some link for support? Thank you very much. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

1 0

Cannot hear voice with symmetric NAT and STUN
by Khoa Pham 08 Feb '17

08 Feb '17

Hi, I have Kamailio as SIP server and RTP server. Client is PJSIP. I read that STUN is for non-symmetric NAT, and RTP server is for symmetric NAT. Supposed A calls B. If A, B both use symmetric NAT and STUN, they cannot hear each other If A or B use non-symmetric NAT and NOT using STUN, they cannot hear each other. Why is that? I read http://tools.ietf.org/id/draft-takeda-symmetric-nat-traversal-00.txt for Prediction Failure, is that related to this problem ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

8 18

Diversion Header - relay messed up
by Roman Dissauer 08 Feb '17

08 Feb '17

When I get an INVITE with Diversion Header the Request is forwarded without Diversion Header and the Request User is taken from Diversion User. Problem is that on the Destination Host I cannot get original Request User what is the intended destination! Is this intended behaviour? How can I change this behaviour? I already tried to delete the Diversion Header on request_route but this didn’t change the behaviour. Thanks in andvance for your help! Roman Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as Carrier IP Incoming INVITE: INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 19. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; reason=unconditional; counter=1. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 260. Content-Disposition: session; handling=required. Content-Type: application/sdp. ... Relayed INVITE: INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582.4fc0216dbecafde29127db502993222c.0. Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 18. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 274. Content-Disposition: session; handling=required. ...

3 3

Second coredump in a month
by Jon Bonilla (Manwe) 08 Feb '17

08 Feb '17

Hi all A server running version 4.4.4+0~20161223011227 has segfaulted for the second time this month. I'm attaching a bt in case someone can explain it to me. cheers, Jon

2 12

kamailio with cassandra
by Никитенко Виталий 08 Feb '17

08 Feb '17

Hi! I want start kamailio with db_cassandra. do as it written in https://www.kamailio.org/docs/modules/4.2.x/modules/db_cassandra.html. Then trying kamctl start, failed with error "table version does not exist!". >From CQLSH table is view cqlsh:kamailio> Describe tables; version location cqlsh:kamailio> select * from version; table_name | table_version ------------+--------------- version | 1 Tell me please wherein may be trouble? Full log below ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/location ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/version INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.allow INFO: permissions [permissions.c:608]: mod_init(): default allow file (/usr/local/etc/kamailio//permissions.allow) not found => empty rule set INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.deny INFO: permissions [permissions.c:617]: mod_init(): default deny file (/usr/local/etc/kamailio//permissions.deny) not found => empty rule set INFO: misc_radius [misc_radius.c:164]: mod_init(): initializing... ERROR: db_cassandra [dbcassa_base.cpp:449]: cassa_translate_query(): table version does not exist! ERROR: db_cassandra [dbcassa_base.cpp:962]: db_cassa_query(): Failed to query Cassandra cluster ERROR: <core> [db.c:401]: db_table_version(): error in db_query ERROR: <core> [db.c:440]: db_check_table_version(): querying version for table location ERROR: usrloc [dlist.c:640]: register_udomain(): error during table version check. ERROR: registrar [reg_mod.c:554]: domain_fixup(): failed to register domain ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:982 ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:987 ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:992

2 1

Kamailio + Asterisk with multidomain
by Tomas Zanet 07 Feb '17

07 Feb '17

Hello, thanks to this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I successfully installed Asterisk 11.6 and Kamailio 4.4.4 on the same machine. Everything works fine. Right now I would like to extend this scenario adding MULTIDOMAIN support, which is not enabled by default Do you see any problems / technical limitations to add multidomain support in this scenario? Before enhancing Kamailio cfg file and Asterisk configuration I would like to know: is this possible? As far as I know Kamailio works fine with multidomain support but I don’t know if Asterisk or both processes can support it. Thanks in advance Regards, T. Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

2 1

DMQ Cluster - hosts loose network connectivity
by José Seabra 07 Feb '17

07 Feb '17

Hello there, My DMQ cluster has 4 nodes and by some reason 2 of them lost the network connectivity for long time(~ 4 hours), after the network of these 2 nodes come back they didn't get connected on DMQ cluster automatically, i had to restart kamailio to get them again on DMQ list. My doubts here are: - Is this an expected behavior of DMQ module? - Is there any way of put them again on DMQ bus without need restart kamailio? Thank you Best Regards José Seabra

3 5

Kamailio parse failed message
by Hai Bui Duc Ha 07 Feb '17

07 Feb '17

Hi all, I have problem when make call with my Android mobile use PJSIP library. Scenario: my client -> Kamailio -> Freeswitch (media server) -> another client (soft phone on Windows) my client: + use Bluestack + Capture via Wireshark + use Wifi Issue: The call will be drop after ~ 30 second. I see the error on Kamailio: *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 13)* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 13)* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<p:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012ACK sip:buiduchahai@125.212.212.36:11000;transport=tcp SIP/2.0#015#012Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias#015#012Max-Forwards: 70#015#012From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Ahuynhngocphap(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO#015#012To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Abuiduchahai(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=2SF4D790Zy6Kj#015#012Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB#015#012CSeq: 29055 ACK#015#012Route: <sip:125.212.212.40;transport=tcp;lr;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>#015#012Content-Length: 0#015#012#015#012>* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [receive.c:129]: receive_msg(): core parsing of SIP message failed (49.156.54.54:50785/2 <http://49.156.54.54:50785/2>)* Seem to the server error when parse (on INVITE SDP) *a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16* (on new ACK message) *ACK sip:buiduchahai@125.212.212.36:11000;transport=tcp SIP/2.0* *Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias* *Max-Forwards: 70* *From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Ahuynhngocphap(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO* *To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Abuiduchahai(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=2SF4D790Zy6Kj* *Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB* *CSeq: 29055 ACK* *Route: <sip:125.212.212.40;transport=tcp;lr;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>* *Content-Length: 0* I think the SIP message is fragmented but when resume package is not correct. Do you have any advice ? Thank you for watching ! Regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

2 11

FOSDEM 2017
by Daniel-Constantin Mierla 07 Feb '17

07 Feb '17

Hello, another edition of FOSDEM is approaching, about 2 weeks left: - https://fosdem.org/2017/ Couple of devs and many Kamailio friends will be around. Like past year, there is a Realtime Communications devroom, this edition on Sunday. Very interesting presentation for our ecosystem: Olle giving a presentation about IPv4/IPv6, Inaki introducing its SFU media server project, Jose talking about JsSIP, Daniel debating about fundraising FreeRTC, Lorenzo with Janus SIP-WebRTC gateway, the other Lorenzo with Homer Sipcature, Saul with Jitsi, Dan with cgrates, Matt with Asterisk, Giovanni with FreeSwitch... Schedule at: - https://fosdem.org/2017/schedule/track/real_time_communications/ On Sunday morning, part of Lua devroom, I will present about using Lua for building RTC services with Kamailio: - https://fosdem.org/2017/schedule/track/lua/ - https://fosdem.org/2017/schedule/event/luartcserviceskamailio/ During the past editions we organized a dinner on Saturday evening. Shall we attempt to do it again in advance this year? Or should we do it on spot based on the weather and mood at the end of Saturday? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

7 13

kamailio proxy behind firewall
by JBF 07 Feb '17

07 Feb '17

Hello, we have the following Configuration for our kamailio installation (we are using TLS and not udp) (1) F5 Firewall (configured as message fowarding), opening a TLS server on the outside (2) SIP proxy, with a TLS server accessed by the F5 . The SIP proxy doesnt see the F5 TLS server (3) SIP registrar REGISTER works find We have the following issue on INVITE: A sends an INVITE to B. The Registrar patches the R-URI with the content of location, which contains the publi ip of the Device (because the device used stun) we force the routing from registrar to proxy by using t_relay (SIP_PROXY_IP) /The proxy tries to route to this R-URI, which is not visible/ I am not sure how to fix that: Record Route is for a true sip proxy, but the Firewall does not have an server facing the SIP proxy: the sip proxy needs to find the proper client socket opened at register to route the INVITE We have arranged for the Firewall to add its own Via, but if i understand correctly, this is used for replies, and here we are dealing with a request forwarding, and t_relay uses the r-ruri to route requests. IT might be why REGISTER works correctly (ie the 200 OK is routed correctly from proxy to firewall) I could arrange for the location table to contain the private ip and port of the firewall connection (through the use of the received/rport info inserted in the Via by the proxy ) That would mean, however that the contact of the user will contain the private interface of the F5 which i found weird. How do you think i should proceed ? any advices are welcome Thank you -- View this message in context: http://sip-router.1086192.n5.nabble.com/kamailio-proxy-behind-firewall-tp15… Sent from the Users mailing list archive at Nabble.com.

2 3

RTPEngine disable/enable crash kamailio
by Sergey Basov 07 Feb '17

07 Feb '17

Hi All. I ave just try to temporary disable 1 of 2 RTPEngines. It was disabled successfully all media dialogs was finished. But when I have try to enable it I have kamailio crashed with error: CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 121 Does anyone have such issue? kamailio 4.4.5 on RHEL 7.2 x86_64 rtpengine 4.4.0 on RHEL 7.2 x86_64 # kamctl fifo nh_show_rtpp all which: no gdb in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/) udp:10.1.23.24:2223:: set=0 index:: 0 disabled:: 0 weight:: 1 recheck_ticks:: 0 udp:10.1.23.25:2223:: set=0 index:: 1 disabled:: 0 weight:: 1 recheck_ticks:: 0 after # kamctl fifo nh_enable_rtpp udp:10.1.23.25:2223 0 it was successfully disabled with media going on after # kamctl fifo nh_enable_rtpp udp:10.1.23.25:2223 1 I got kamailio crash with error CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 121 sorry I does not have coredump, it is production system. my config regarding rtpengine: modparam("rtpengine", "rtpengine_allow_op", 1) modparam("rtpengine", "rtpengine_sock", "udp:10.1.23.24:2223 udp:10.1.23.25:2223") Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

5 12

execute config route by ctp command?
by Juha Heinanen 07 Feb '17

07 Feb '17

Is there a way to execute a named config route, such as an event_route, by an rpc command? -- Juha

3 4

Re: [SR-Users] Spurious dialog timeouts
by Pranathi Venkatayogi 07 Feb '17

07 Feb '17

Any clue on this? Or to work around this, can someone shed light on how to differentiate when “Dialog:end” is called due to actual call drop - Bye vs some timeout. I need to do some special logic only on call drop. Relevant snippet from the logs are: 7719 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7720 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7721 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7722 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7723 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7724 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent2@translation.sms-test.cyracom.com. Marking free 7725 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7726 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7727 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '1478097bad2847cfaa60a768d76044b0' and tags '57315e78d6514251876bd5a9e32347d2' '00a1eac150f04e2592527372412f059e' ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ -----Original Message----- From: Pranathi Venkatayogi Sent: Friday, February 03, 2017 2:29 PM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Subject: Spurious dialog timeouts Hi, I have two kamailio proxies between the customer and agent. Cust -> proxy1 -> proxy2 -> agent. The dialogs are successfully established and remain undeleted within the customer / agent client. However in Proxy2 I am seeing spurious dialog timeout. I am attaching kamailio log trace below. Also attached is the siptrace on the agent/customer side. Question: How is this dlg_ontimeout() generated despite dialog being successfully established and undelete? How to avoid it. Kamailio (proxy2) log: 7719 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7720 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7721 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7722 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7723 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7724 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent2@translation.sms-test.cyracom.com. Marking free 7725 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7726 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7727 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '1478097bad2847cfaa60a768d76044b0' and tags '57315e78d6514251876bd5a9e32347d2' '00a1eac150f04e2592527372412f059e' 7728 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7729 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7730 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7731 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7732 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7733 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent5@translation.sms-test.cyracom.com. Marking free 7734 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7735 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7736 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '04d924dc160d4e0c9366d4699f579793' and tags '9b107772c82148c0a09182dc86804c51' '44b0a200438a42f8a6ac931ed72f795a' Thanks for your input, Pranathi

1 0

SIP message order
by Andy Chen 06 Feb '17

06 Feb '17

Hello sr-users, First of all, I'm new to this forum so any help would be greatly appreciated as my Kamailio knowledge is somewhat limited. Thanks for your patience. So to the question at hand. If the originator of the SIP method sends SIP messages out of order, does the Kamailio put them back in order before relaying it to the destination device? I'm seeing where the 200 OK was sent first before the SIP UPDATE within the same CSeq number. The reason I'm asking is because I'm seeing this behavior now with UDP transport protocol and I'm trying to justify why we don't need to switch to TCP to fix this issue. Thanks. --Andy

2 1

Kamailio 4.2.3-3.1 Segfaulting - Need debuginfo RPM for kamailio-4.2.3-3.1.x86_64
by Peter DiSanto 06 Feb '17

06 Feb '17

Our Kamailio 4.2.3 recently began seg-faulting in production. I ran a bt full, but without the debuginfo I presume it isn't much use. We cannot locate the original RPMs for this version to be able to obtain the debuginfo RPM. We are running the kamailio-4.2.3-3.1.x86_64 RPM for CentOS6. Anybody have it or can help us find the debug, or offer other advice?

2 1

loose_route ACK and BYE
by Ben Hood 06 Feb '17

06 Feb '17

Hi, I've fixed a loose_route problem with a broken gateway. Essentially gateway sends ACKs and BYEs with an unroutable contact header (i.e. it sticks the address of the proxy into the header, rather than the destination UA address). This is all on the same network, so there is not NAT. So by caching the contact URI to for replies back to the gateway, I can populate $ru to forward the ACK or BYE back to the correct UA. However, what I've noticed is that for BYEs, I also need to set the $du variable in addition to $ru. But this is required for ACKs. This is not a problem per se, but I'm interesting to know why - could anybody explain? Cheers, Ben

2 1

Os on virtual machine
by Diego Nadares 04 Feb '17

04 Feb '17

Hi guys, We are using vmware to run kamailio. The thing is that they give us support only if we install Red Hat. Did you test kamailio on RH? Any issues or things to have in mind? What linux dist do you recommend? Thanks! Diego.

4 4

Fosdem 2017 dinner
by Daniel-Constantin Mierla 04 Feb '17

04 Feb '17

Hello, the dinner at Fosdem 2017 is planned to start at 20:00, on Saturday, Feb 4, 2017. The address is: Le Pickwick Avenue Adolphe Buyl 79 1050 Ixelles, Belgium * https://en.yelp.be/biz/le-pickwick-ixelles The place is nearby Fosdem location. If you announced the intention to participate at the dinner before Feb 1, 2017, your seat is reserved. If you did it afterwards, please contact me during RTC Devroom to confirm the availability of the seats. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Spurious dialog timeouts
by Pranathi Venkatayogi 03 Feb '17

03 Feb '17

Hi, I have two kamailio proxies between the customer and agent. Cust -> proxy1 -> proxy2 -> agent. The dialogs are successfully established and remain undeleted within the customer / agent client. However in Proxy2 I am seeing spurious dialog timeout. I am attaching kamailio log trace below. Also attached is the siptrace on the agent/customer side. Question: How is this dlg_ontimeout() generated despite dialog being successfully established and undelete? How to avoid it. Kamailio (proxy2) log: 7719 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7720 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7721 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7722 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7723 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7724 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent2@translation.sms-test.cyracom.com. Marking free 7725 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7726 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7727 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '1478097bad2847cfaa60a768d76044b0' and tags '57315e78d6514251876bd5a9e32347d2' '00a1eac150f04e2592527372412f059e' 7728 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=490 a=16 n=if 7729 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=481 a=24 n=is_known_dlg 7730 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=482 a=63 n=assign 7731 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=489 a=16 n=if 7732 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=485 a=26 n=xlog 7733 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: INFO: <script>: Translation: Dialog ended for sip:agent5@translation.sms-test.cyracom.com. Marking free 7734 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=486 a=63 n=assign 7735 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: exec: *** cfgtrace:local_route=[dialog:end] c=[//etc/kamailio//routing.cfg] l=487 a=63 n=assign 7736 Feb 3 21:41:20 labsip00 /usr/sbin/kamailio[11323]: WARNING: dialog [dlg_handlers.c:1515]: dlg_ontimeout(): timeout for dlg with CallID '04d924dc160d4e0c9366d4699f579793' and tags '9b107772c82148c0a09182dc86804c51' '44b0a200438a42f8a6ac931ed72f795a' Thanks for your input, Pranathi

1 0

FreeSWITCH at FOSDEM, SIP and WebRTC Load Balancing and High Availability
by Giovanni Maruzzelli 03 Feb '17

03 Feb '17

See you all, in person or in the intertube, at: https://fosdem.org/2017/schedule/event/freeswitch/ sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT

1 0

Websocket TLS Issue
by Jade SZ 03 Feb '17

03 Feb '17

Hi Guys, I am trying to setup the following flow: Browser >> WSS >> HA Proxy >>> WSS >> Kamailio But getting TLS errors in Kamailio logs: *[29634]: ERROR: <core> [tcp_read.c:1321]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f68ebe872b0 r: 0x7f68ebe87330* *[29631]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F10B:SSL routines:SSL3_GET_RECORD:wrong version number* Browser <-----wss---->Kamailio works fine with same certs. Both HA Proxy and Kamilio are installed on separate servers, hosting on same port with different domain. Kamailio tls.conf has method = TLSv1 *@HA Proxy:* openssl s_client -connect HA-PROXY-DOMAIN:*10443* SSL-Session: Protocol : TLSv1.2 *@Kamailio :* openssl s_client -connect KAMAILIO-DOMAIN:*10443* SSL-Session: Protocol : TLSv1 So I made HA Proxy to be on TLSv1 "ssl-default-bind-options force-tlsv10" But still I get the same TLS error in Kamailio. *HA Proxy config looks like:* *frontend public* * bind *:10443 ssl crt /etc/haproxy/certs/cert.pem* * acl is_websocket hdr_end(host) -i m1.some-domain.com <http://m1.some-domain.com>* * use_backend wss if is_websocket* * default_backend wss* *backend wss* * timeout server 600s* * server ws1 k1.some-domain.com:10443 <http://k1.some-domain.com:10443>* * server ws1 k2.some-domain.com:10443 <http://k2.some-domain.com:10443>* Need some direction, thanks in advance. Regards, Jade

4 4

id column i tables
by Kristian F. Høgh 02 Feb '17

02 Feb '17

Hi list, We provision our kamailio database from an external system. In the future we would like to provision from two different systems. To prevent the two systems from touching each others data, we would like to create views from the two tables. provision system 1 would write to table1 provision system 2 would write to table2 kamailio will read a view over the two tables. The tables we will create views over are: - address - aliases - dispatcher - uri - usr_preferences Now to my question :-) Does kamailio use the id field in any of the tables above? The id's are auto-incremented, and we might get conflicts between table1 and table2 *Kristian Høgh* Telefon: 4422 8822 support(a)uni-tel.dk[1] Gydevang 19 | 3450 Allerød www.uni-tel.dk [2] -------- [1] mailto:support@uni-tel.dk [2] http://www.uni-tel.dk

2 2

Disabling DNS SRV records checks
by Ahmed Fouad 01 Feb '17

01 Feb '17

Hi, I'm running kamailio version 4.1 and having the following DNS related configuration use_dns_failover = off dns_try_naptr = no dns_udp_pref = -1 dns_try_ipv6 = off dns_retr_time=1 dns_retr_no=1 dns_servers_no=1 dns_use_search_list=no But still am seeing SRV record checks for my domains which I like to completely disable these checks and only use A records only. x.x.x.x.57828 > 8.8.8.8.53: 7417+ SRV? _sip._udp.subdomain.com 8.8.8.8.53 > x.x.x.x.57828: 7417 NXDomain 0/1/0 (129) I can't find a DNS option to disable SRV checks. Can you please advise ?

2 3

could not make ul_rm_contact work
by Aqs Younas 01 Feb '17

01 Feb '17

Greetings list, I am trying to remove a specific contact from memory but always resulted in vain. root@debian:/usr/local/kamailio/sbin# ./kamctl ul show Domain:: location table=1024 records=1 max_slot=1 AOR:: 1040(a)192.168.0.101 Contact:: sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp Q= Expires:: 2416 Callid:: _3595KDc1PinhUGFpSf4Dg.. Cseq:: 2 User-agent:: Z 3.9.32144 r32121 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:192.168.0.101:5060 Methods:: 5087 Ruid:: uloc-58911db0-f45-1 Reg-Id:: 0 Last-Keepalive:: 1485905402 Last-Modified:: 1485905402 . /kamctl fifo ul_rm_contact location 1040(a)191.168.0.101 sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp 404 AOR not found Tried another command root@debian:/usr/local/kamailio/sbin# ./kamctl fifo ul_rm location 1040(a)192.168.0.101 root@debian:/usr/local/kamailio/sbin# ./kamctl ul show Domain:: location table=1024 records=1 max_slot=1 AOR:: 1040(a)192.168.0.101 Contact:: sip:1040@192.168.0.102:36619;rinstance=1dfaed088d873a18;transport=tcp Q= Expires:: deleted Callid:: _3595KDc1PinhUGFpSf4Dg.. Cseq:: 2 User-agent:: Z 3.9.32144 r32121 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:192.168.0.101:5060 Methods:: 5087 Ruid:: uloc-58911db0-f45-1 Reg-Id:: 0 Last-Keepalive:: 1485905402 Last-Modified:: 1485905402 And this. . /kamctl ul rm 1040(a)192.168.0.1 sip:1040@192.168.0.102:36619 ;rinstance=1dfaed088d873a18;transport=tcp 404 AOR not found Can someone help me know to remove a contact from memory? Any pointer is much appreciated. Best Regards.

2 3

sdp_with_transport
by M. Salman 01 Feb '17

01 Feb '17

Hi Guys, Just wanted to clarify the following case: what should be result of sdp_with_transport("RTP/SAVPF") on line: m=audio 10231 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 I am having a weird behavior for two different versions of Kamailio. I hope I am doing something wrong. kamailio 4.3.2 => sdp_with_transport("RTP/SAVPF") = true kamailio 4.4.5 => sdp_with_transport("RTP/SAVPF") = false -- Regards M. Salman VoIP Professional

3 2

Re: [SR-Users] sr-users Digest, Vol 140, Issue 30
by Andrew Miller 01 Feb '17

01 Feb '17

Is it too late to request a place for the meal at FOSDEM? I would love to come. Andy Miller On 30 Jan 2017 11:00, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Diversion Header - relay messed up (Roman Dissauer) > 2. Re: dlg_set_timeout bye to websockets client (Switch168) > 3. Presence Module do not load db_url (Vu. Minh Cao) > 4. Re: FOSDEM 2017 (Giacomo Vacca) > 5. Re: Replace old registration record with new one while > keeping single AOR per contact (Federico Cabiddu) > 6. Re: Replace old registration record with new one while > keeping single AOR per contact (Aqs Younas) > 7. Re: mhomed opened sockets (Diego Nadares) > 8. Re: [sr-dev] FOSDEM 2017 (Daniel-Constantin Mierla) > 9. Re: FOSDEM 2017 (Ivan Todorov) > 10. Re: [Kamailio-Business] [sr-dev] FOSDEM 2017 (Alexandr Dubovikov) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 29 Jan 2017 18:42:23 +0100 > From: Roman Dissauer <roman(a)dissauer.net> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] Diversion Header - relay messed up > Message-ID: <DB3FC45E-ACCE-426E-B662-031909F6BA40(a)dissauer.net> > Content-Type: text/plain; charset=utf-8 > > When I get an INVITE with Diversion Header the Request is forwarded > without Diversion Header and the Request User is taken from Diversion User. > Problem is that on the Destination Host I cannot get original Request User > what is the intended destination! > Is this intended behaviour? How can I change this behaviour? > > I already tried to delete the Diversion Header on request_route but this > didn’t change the behaviour. > > Thanks in andvance for your help! > Roman > > Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as > Carrier IP > > Incoming INVITE: > > INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. > Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. > From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. > To: <sip:436761234567@x.x.x.x>. > Call-ID: 906143813_44460603(a)y.y.y.y. > CSeq: 28100 INVITE. > Max-Forwards: 19. > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, > NOTIFY,PRACK,UPDATE,OPTIONS. > Accept: application/sdp, application/isup, application/dtmf, > application/dtmf-relay, multipart/mixed. > Contact: <sip:+43987654321@y.y.y.y:5060>. > P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. > Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; > reason=unconditional; counter=1. > Supported: timer,100rel. > Session-Expires: 1800. > Min-SE: 90. > Content-Length: 260. > Content-Disposition: session; handling=required. > Content-Type: application/sdp. > ... > > > Relayed INVITE: > > INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. > Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. > Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. > Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582. > 4fc0216dbecafde29127db502993222c.0. > Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch= > z9hG4bK02B7871714d9345f843. > From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. > To: <sip:436761234567@x.x.x.x>. > Call-ID: 906143813_44460603(a)y.y.y.y. > CSeq: 28100 INVITE. > Max-Forwards: 18. > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, > NOTIFY,PRACK,UPDATE,OPTIONS. > Accept: application/sdp, application/isup, application/dtmf, > application/dtmf-relay, multipart/mixed. > Contact: <sip:+43987654321@y.y.y.y:5060>. > P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. > Supported: timer,100rel. > Session-Expires: 1800. > Min-SE: 90. > Content-Length: 274. > Content-Disposition: session; handling=required. > ... > > > > > > > > ------------------------------ > > Message: 2 > Date: Sun, 29 Jan 2017 18:21:37 +0000 > From: Switch168 <team(a)switch168.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org>, miconda(a)gmail.com > Subject: Re: [SR-Users] dlg_set_timeout bye to websockets client > Message-ID: > <CAJ4KJRjeZT++vYWkfnY10WnkWztpVtpSV700TW0ebh > 7z+jRAEg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi Daniel, > > > For anyone else who might run into this issue I found > https://github.com/kamailio/kamailio/issues/85 to be related to mine. And > its a good a starting point. > > Cheers > Andrew > > > > > On Fri, Jan 27, 2017 at 10:47 AM Switch168 <team(a)switch168.com> wrote: > > > HI Daniel, > > > > modparam("dialog", "dlg_flag", 4) modparam("dialog", "send_bye", 1) > > modparam("dialog", "timeout_noreset", 1) > > > > So regular bye's by useragent i can succesfully relay to this > > <sip:6gjlmali@ec2c66jsa0ei.invalid;transport=ws> > > style of uri by using this snippet below > > > > # Handle requests within SIP dialogs > > route[WITHINDLG] { > > if(is_method("BYE")) { > > xlog("DEBUG: Received BYE"); > > route(NATDETECT); > > loose_route(); > > dlg_manage(); > > t_check_trans(); > > handle_ruri_alias(); > > rtpengine_delete(); > > route(RELAY); > > exit; > > } > > ... > > > > But the byes that are sent out by dlg_set_timeout("$var(timer)") gives > out > > error cannot resolve > > the random.uri and I don't know how to fix the bye before it get sent > out. > > > > Thanks > > Andrew > > > > > > > > > > > > > > > > > > On Fri, Jan 27, 2017 at 12:15 AM, Daniel-Constantin Mierla < > > miconda(a)gmail.com> wrote: > > > > Hello, > > > > isn't the dialog module setting the right value there? What function are > > you using to update the contact? > > > > Cheers, > > Daniel > > > > On 27/01/2017 06:01, Andrew Tan wrote: > > > > Hello, > > > > Just wondering if it's possible to intercept the bye message that > > dlg_set_timeout sends out to do some nat_helper function to fix it so the > > bye can be sent to random.invalid uris. > > > > I know there is the edge proxy outbound module but wondering if I can do > > it without. > > > > Regular bye between 2 clients i was able intercept and fix with nat > helper > > but I dont know how to fix the byes that is sent out from dlg_set_timeout > > function. > > > > Thanks in advance! > > Andrew > > > > > > > > _______________________________________________ > > 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 Mierlawww.twitter.com/miconda -- > www.linkedin.com/in/miconda > > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - > www.asipto.com > > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > > > > _______________________________________________ > > 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

Re: [SR-Users] FOSDEM 2017
by Hugh Waite 01 Feb '17

01 Feb '17

Hi all, Please count me and one other in for the meal on Saturday. See you in Brussels! Hugh Waite On 30 Jan 2017 11:00 a.m., <sr-users-request(a)lists.sip-router.org> wrote: Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Diversion Header - relay messed up (Roman Dissauer) 2. Re: dlg_set_timeout bye to websockets client (Switch168) 3. Presence Module do not load db_url (Vu. Minh Cao) 4. Re: FOSDEM 2017 (Giacomo Vacca) 5. Re: Replace old registration record with new one while keeping single AOR per contact (Federico Cabiddu) 6. Re: Replace old registration record with new one while keeping single AOR per contact (Aqs Younas) 7. Re: mhomed opened sockets (Diego Nadares) 8. Re: [sr-dev] FOSDEM 2017 (Daniel-Constantin Mierla) 9. Re: FOSDEM 2017 (Ivan Todorov) 10. Re: [Kamailio-Business] [sr-dev] FOSDEM 2017 (Alexandr Dubovikov) ---------------------------------------------------------------------- Message: 1 Date: Sun, 29 Jan 2017 18:42:23 +0100 From: Roman Dissauer <roman(a)dissauer.net> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Diversion Header - relay messed up Message-ID: <DB3FC45E-ACCE-426E-B662-031909F6BA40(a)dissauer.net> Content-Type: text/plain; charset=utf-8 When I get an INVITE with Diversion Header the Request is forwarded without Diversion Header and the Request User is taken from Diversion User. Problem is that on the Destination Host I cannot get original Request User what is the intended destination! Is this intended behaviour? How can I change this behaviour? I already tried to delete the Diversion Header on request_route but this didn’t change the behaviour. Thanks in andvance for your help! Roman Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as Carrier IP Incoming INVITE: INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 19. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; reason=unconditional; counter=1. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 260. Content-Disposition: session; handling=required. Content-Type: application/sdp. ... Relayed INVITE: INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582. 4fc0216dbecafde29127db502993222c.0. Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 18. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE, NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 274. Content-Disposition: session; handling=required. ... ------------------------------ Message: 2 Date: Sun, 29 Jan 2017 18:21:37 +0000 From: Switch168 <team(a)switch168.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>, miconda(a)gmail.com Subject: Re: [SR-Users] dlg_set_timeout bye to websockets client Message-ID: <CAJ4KJRjeZT++vYWkfnY10WnkWztpVtpSV700TW0ebh7z+jRAEg(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi Daniel, For anyone else who might run into this issue I found https://github.com/kamailio/kamailio/issues/85 to be related to mine. And its a good a starting point. Cheers Andrew On Fri, Jan 27, 2017 at 10:47 AM Switch168 <team(a)switch168.com> wrote: > HI Daniel, > > modparam("dialog", "dlg_flag", 4) modparam("dialog", "send_bye", 1) > modparam("dialog", "timeout_noreset", 1) > > So regular bye's by useragent i can succesfully relay to this > <sip:6gjlmali@ec2c66jsa0ei.invalid;transport=ws> > style of uri by using this snippet below > > # Handle requests within SIP dialogs > route[WITHINDLG] { > if(is_method("BYE")) { > xlog("DEBUG: Received BYE"); > route(NATDETECT); > loose_route(); > dlg_manage(); > t_check_trans(); > handle_ruri_alias(); > rtpengine_delete(); > route(RELAY); > exit; > } > ... > > But the byes that are sent out by dlg_set_timeout("$var(timer)") gives out > error cannot resolve > the random.uri and I don't know how to fix the bye before it get sent out. > > Thanks > Andrew > > > > > > > > > On Fri, Jan 27, 2017 at 12:15 AM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > > Hello, > > isn't the dialog module setting the right value there? What function are > you using to update the contact? > > Cheers, > Daniel > > On 27/01/2017 06:01, Andrew Tan wrote: > > Hello, > > Just wondering if it's possible to intercept the bye message that > dlg_set_timeout sends out to do some nat_helper function to fix it so the > bye can be sent to random.invalid uris. > > I know there is the edge proxy outbound module but wondering if I can do > it without. > > Regular bye between 2 clients i was able intercept and fix with nat helper > but I dont know how to fix the byes that is sent out from dlg_set_timeout > function. > > Thanks in advance! > Andrew > > > > _______________________________________________ > 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 Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > _______________________________________________ > 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

any public databases or schemas for dial plans?
by Daniel Pocock 31 Jan '17

31 Jan '17

Hi, Are there any public databases of dial plan information? Are there any schemas (e.g. XML / XSD) useful for describing arbitrary dial plans from different carriers around the world? The type of things that I'm interested in: - being able to translate any local number to E.164 (if possible) - knowing the expected length (or range of lengths) for different numbers - being able to recognize numbers that can't be translated to E.164 (e.g. the emergency numbers) - being able to classify specific types of number that usually have a carrier-specific or country-specific meaning (e.g. classifying numbers as directory assistance, customer service, emergency) I'm familiar with Google's libphonenumber, although it is not a database, rather, it is a library that has dial plan logic in Java. It supports many of the things I want for numbers that can be translated to E.164 and country-level dial plans, but not local dial plans (e.g. dialing a London number without the 020 area code). Regards, Daniel

2 2

Using DNS for Service Discovery with Kamailio
by Casco, Alessio 31 Jan '17

31 Jan '17

Hello Guys! I'd like to be able to use SRV records inside dispatcher groups, so I can automatise the discovery of new backends on my network using DNS I've tried with the following formats into the dispatcher.list file: $group _sip._udp.sip-voice_backends $group sip-voice_backends and none looks to be working, running the kamailio_ctl dispatcher.list won't show me the group set with SRV records. ( I also try NAPTR records, with no luck as well.) * Does the dispatcher module support SRV or NAPTR records? (maybe my format is not correct) * If not, do you have any other idea on what I could do to achieve what I'd like to do? Thanks Alessio

3 7

RFC: Kamailio tutorials/cookbooks/docs - wiki or markdown
by Daniel-Constantin Mierla 31 Jan '17

31 Jan '17

Hello, I want to start a discussion about some of the kamailio tutorials from the wiki, mainly those related to core cookbook, variables and transformations. They are now in dokuwiki at: - https://www.kamailio.org/wiki/#cookbooks The wiki served us pretty well so far, however the contributions there are not that active. From time to time we still get spammers, even we added restriction that only registered and authenticated users can write. I am thinking that maybe people don't like creating yet another account just add some example or rephrase for clarity. On the other hand, there are over 400 registered users, many of them I expect to be either made because if misunderstanding the purpose or by spammers that guessed the (sip) captcha (plenty of random usernames). Besides that I think the wiki adds overhead when releasing a new version as each time we have to copy and paste old version content to a new set of pages and update the version number. With the above in mind, I wonder if won't be better to store those tutorials in markdown format and use mkdocs.org or gitbook (https://github.com/GitbookIO/gitbook) tools to generate the html and host it in kamailio.org. The wiki will still be used to index them. Among the benefits I see: * store in github.com (a new repo, like: kamailio-docs) and all devs can contribute directly, other users can make pull requests * create branches for each major version of kamailio and backport across them with git whenever is applicable * the edit can be done directly via github.com website (similar to the wiki right now) and even read it directly from github in html transformed from markdown by github * one can get them offline by cloning the git repo, use them or enhance when offline Many if the tutorials will stay in the wiki, but I think those that are related to documenting per kamailio version should be migrated to gihub storage + markdown. If there is a strong pro opinion for this change, I can try to convert one of the cookbooks very soon so people can feel better the difference. If we get to a consensus, then I think we can have those tutorials in the new format for v5.0. Should anyone have comments, other suggestions or improvements, let's put them on debate. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

2 7

serial forking calls using TSILO
by Tomas Zanet 30 Jan '17

30 Jan '17

Hello, I'm using Kamailio 4.4.4 with TSILO module in order to support Push Notifications used by our voip app on Apple ios10. So far, everything works fine: Kamailio can get an incoming call, suspend it, then send push notification with an external script, receive a new app registration and then call it after resuming the invite. To do so, thanks to Mr, Cabiddu, I used all the functions described here http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-… Right now, I would like to support serial forking calls using push notification because it was already supported with legacy voip. To do so, I used failure_route function where, after getting the call destinations and setting them into an avp, I set the new SIP request uri $ru. Here is my failure_route: failure_route[MANAGE_FAILURE] route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) t_reply("404","Not found"); exit; #!endif if (is_avp_set("$avp(group_members_db)")) $ru ="sip:"+$avp(group_members_db)+"@"+$fd; $avp(group_members_db) = $null; route(LOCATION); exit; I am troubleshooting this scenario and this is what I see: 1) Kamailio receives incoming call 2) Suspend it 3) Send Push notification to account1 4) Kamailio receives account1 registration -> INVRESUME route is now called 5) Then call account1 6) Nobody answers the call 7) failure_route[MANAGE_FAILURE] is now called -> set $ru with next sipaccount: account2 8) suspend invite 9) send push notification to account2 10) account2 sends its sip registration -> INVRESUME route is now called 11) Kamailio calls account1 instead of account2 -> this is my issue On step 10: despite account2 is registered, I checked it using "kamctl ul show", the second call is forwarded all'account1 and not to account2. I also checked the TSILO logs and everything seems ok. Here is the logs: First call: suspended transaction [53945:1648394094] asterisk => account1 htable key value [53945:1648394094] resuming trasaction [53945:1648394094] account1 53945:1648394094) second call: suspended transaction [53945:1648394094] asterisk => account2 htable key value [53945:1648394094] resuming trasaction [53945:1648394094] account2 53945:1648394094) In order to find out the issue I put some xlogs, printing $ru value: what I see is the $ru value is set correctly on failure_route but as soon as the t_continue is called, the ru overwritten back to account1 Can anyone address me to find out the solution? Thanks in advance T. Tomas Zanet Software Design Department tzanet(a)came.com

3 7

Re: [SR-Users] FOSDEM 2017
by DanB 30 Jan '17

30 Jan '17

Hey Daniel, if anything to be reserved, please count us also in. Will come back regarding number of seats but just to be aware that we are interested to join the dinner. Thanks! DanB

8 7

mhomed opened sockets
by Diego Nadares 30 Jan '17

30 Jan '17

Hi Guys, We have two network interfaces. One public and one private. We had enabled mhomed and works great BUT if we do a netstat -naup | grep 5060 we see that some sockets keeps opened also when the call is finished. The source port in netstat it's not 5060 so I suppose that kamailio use another one for something else. Is this the right behavior? How long does this sockets remain opened? Are we missing something in our cfg? 6 2017-01-25 15:52:04.986651 172.16.213.38 172.16.200.159 SIP/SDP 146 Request: INVITE sip:xxx@172.16.200.159:5060 34 2017-01-25 15:52:19.636895 172.16.213.38 172.16.200.159 SIP 716 Request: ACK sip:xxx@172.16.200.159:5060 | 37 2017-01-25 15:54:38.947831 172.16.208.111 172.16.213.38 SIP 736 Request: BYE sip:127.0.0.8:5060;line=sr-987jhlk* | 40 2017-01-25 15:54:38.950899 172.16.213.38 172.16.200.159 SIP 660 Request: BYE sip:172.16.208.111:5060 | [root@dwrfsd01 kamailio]# netstat -naup | grep 5060 udp 0 0 172.16.213.38:*56086* <http://172.16.213.38:56086> 172.16.200.159:5060 ESTABLISHED 17015/kamailio udp 0 0 172.16.213.38:5060 0.0.0.0:* 17006/kamailio udp 0 0 172.16.213.38:*41971* <http://172.16.213.38:41971> 172.16.200.159:5060 ESTABLISHED 17051/kamailio udp 0 0 172.16.213.38:*54651* <http://172.16.213.38:54651> 172.16.200.159:5060 ESTABLISHED 17008/kamailio Thanks in advance! Diego.

2 3

Presence Module do not load db_url
by Vu. Minh Cao 29 Jan '17

29 Jan '17

Hi, I got error when loading presense module in Kamailio 4.4: [Configure] loadmodule "presence.so" modparam("presence", "db_url", "mysql://kamailio:kamailiorw@127.0.0.1/kamailio") [Error] ERROR: presence [presence.c:608]: fixup_presence(): Bad config - you can not call 'handle_publish' function (db_url not set) Which is really strange since according to the source code this error just occurs if “db_url” is not set. I already checked my database also, I can access mysql with about username/password. Yours sincerely, — Vu

1 0

rtpengine - Key file error
by Serhat Guler 29 Jan '17

29 Jan '17

Hello, I am trying to setup my Kamailio environment on a new Debian system. Everything went well, except I started facing a problem with the rtpengine. It did install it fine and I can view the program parameters via the help menu; however, when I run it with this command "rtpengine --interface 192.168.0.66 --listen-ng=127.0.0.1:22222 -m 30000 -M 35000 -L 2" or any other commands, it gives an error as "CRIT: Fatal error: Bad command line: Key file does not have group 'rtpengine'". Where is this key file it is complaining about ? I couldn't find anything neither in the doc nor anywhere else. Cheers, Serhat

2 2

missing memory after tls cleanup
by Andy 28 Jan '17

28 Jan '17

Hi, the tls total memory before and after a tls connection is cleaned up does not match, during test there were 10 tls connection created for which I see tls_h_close and tls_h_tcpconn_clean were called exactly 10 times for the same connection ids, bit before and after the tls shm memory does not match, using command: kamcmd mod.stats tls shm could there be a leak? please suggest thanks! -- View this message in context: http://sip-router.1086192.n5.nabble.com/missing-memory-after-tls-cleanup-tp… Sent from the Users mailing list archive at Nabble.com.

2 4

Ansible Kamailio role
by Alberto Llamas 27 Jan '17

27 Jan '17

Hello Kamailians, I just wanted to share an Ansible role for the installation and compilation of a Kamailio server. It may required improvements and I will be more than happy to heard them. *https://galaxy.ansible.com/albertollamaso/Ansible-kamailio-role/ <https://galaxy.ansible.com/albertollamaso/Ansible-kamailio-role/>* Cheers, -- Alberto Llamas Telecommunications Engineer dCAA|dCAP|KPAC|SSCA

4 4

Re: [SR-Users] FOSDEM 2017
by DanB 26 Jan '17

26 Jan '17

Hey Daniel, Would be there 2 persons for sure, so please count us also in. Thanks, DanB

2 1

parallel media forking during ringing phase
by Tomas Zanet 26 Jan '17

26 Jan '17

Hi all, I'm writing to the community because I need a hint from Kamailio and voip gurus: the goal is adding parallel forking call with early-media to our PABX where there's a Kamailio instance running. I mean media streams should be sent to all call destinations before answering the call order, for instance, to see the incoming video. As far as I know, from signaling side should not be a problem because, using Kamailio, there’s append_branch function which forks many branches to all users. The problem is related to media streams because there’s no way to fork media to all users after 180 + SDP or 183, right? Obviously all users must negotiate the same media because media transcoding is basically impossible, especially for video streams. I think there are two ways to do that: 1) using Kamailio and Asterisk. Asterisk has Queue function with ringall strategy, but I’m not sure if Asterisk supports multi domain scenarios. Have ever used Asterisk with Kamailio in multi domain scenarios? 2) enhancing rttpproxy-ng. I don't really know the effort to achieve that. In your opinion, what’s the easiest way to achieve that? Thanks, Best Regards. Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

2 2

How does Kamailio decide which protocol to use when fwding to another proxy?
by Pranathi Venkatayogi 25 Jan '17

25 Jan '17

Hi, I have two instances of Kamailio acting as edge proxies. One on the customer side and one on the agent side. Like: customer -> proxy1 -> proxy2 -> agent. Both customer and agent are registered to proxy1/proxy2 via TLS. However when proxy1 forwards to proxy2, it is using UDP. How can I force it to use TLS? Attached is the result of nslookup on the domain: translation.sms-test.cyracom.com. Thanks Pranathi [cid:image001.jpg@01D27631.42773770] From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Ryan Wagoner Sent: Tuesday, January 24, 2017 8:26 AM To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Asterisk Proxy Multiple Devices / BLF Issues I'm following the latest Kamailio and Asterisk Realtime guide to offload registrations from my FreePBX / Asterisk setup and possibly load balance down the road. I'm running Kamailio 4.4.5 and Asterisk 11.6-cert15. I realize FreePBX isn't realtime and will work around that with a database view, etc. I was excited to see Kamailio will handle multiple devices registering to the same device/extension and placing / receiving calls works. I did run into an issue when any device unregisters Kamailio always forwards the register with expires 0 to Asterisk. To workaround this I modified the route[REGFWD] and added the if($hdr(Expires)==$null) chunk of code. I wanted to use caller->count, but ran into stale contact records with expires set to deleted. I then tried enumerating the contacts, but don't understand why ulc(caller->expires) is 10 when kamctl ul show shows expires deleted. The code below works, but I was hoping for an explanation of the expires = 10 or if there was a better way to handle this scenario. Additionally I enabled presence (WITH_PRESENCE) but Kamailio responds 489 bad event for subscribe requests from devices registered to it. I was hoping it would proxy these to Asterisk for BLF support. If somebody could point me in the right direction it would be appreciated. # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } if($hdr(Expires)==$null) { reg_fetch_contacts("location", "$sel(contact.uri)", "caller"); $var(i) = 0; $var(j) = 0; while($var(i) < $(ulc(caller=>count))) { if($(ulc(caller=>expires)[$var(i)])!=10) { $var(j) = $var(j) + 1; } $var(i) = $var(i) + 1; } if($var(j)>=1) { return; } } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } Thanks, Ryan

2 3

kamctl cr reload got 500 failed to re-built tree
by Igor Potjevlesch 25 Jan '17

25 Jan '17

Hello, With more than 173k entriers into carrierroute, now I'm doing "kamctl cr reload", I got the following reply: 500 failed to re-built tree, see log What could going wrong? Regards, Igor. --- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

2 4

Issue with From address being modified in ACK when UAC module used version 4.1
by Jonathan Hunter 25 Jan '17

25 Jan '17

Hi Guys, running kamailio 4.1.4 and using uac_replace_from, I am seeing a strange issue with the proxying of an ACK message back from a carrier to freeswitch on the ingress path into a network. So its just a normal call inbound, where on outbound leg we modify the From address, on the inbound leg all remains the same. Now after the ingress side receives the 200ok, it sends an ACK as below; ACK sip:+441624111111@192.168.24.8:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 4.4.4.4:5060;branch=z9hG4bK+7f5c8c756ae3b26a956b33b88c77c29f1+sip+3+aa6d1466 Call-ID: 8791dbd3855eeafd484f397de6e2f76e(a)carrier.peering.telecom.im From: "+44792498881474" <sip:+44792498881474@192.168.24.8:5080;user=phone>;tag=carrier.peering.telecom.im+3+863d20a3+1b9801d2 To: "+441624111111" <sip:+441624111111@8.8.8.8:5080;user=phone>;tag=6rrtgNFQDNrFF CSeq: 1 ACK Contact: <sip:4.4.4.4:5060> Route: <sip:192.168.24.8;lr=on;ftag=carrier.peering.telecom.im+3+863d20a3+1b9801d2;vsf=AAAAAAAAAAYNBgAPAAgLAgZ3UTMACgBXFUsVFwwcDkAKTwMZGh0YAA8KDkEEQ1NYC0NDXgdOFRpSHg1vbmU-> Content-Length: 0 Max-Forwards: 68 However kamailio changes the From address; ACK sip:+441624111111@192.168.24.8:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 109.73.69.165:5060;branch=z9hG4bKc1ce.47974fc3da2b669a78f2dcc9a057a127.0 Via: SIP/2.0/UDP 4.4.4.4:5060;rport=5060;branch=z9hG4bK+7f5c8c756ae3b26a956b33b88c77c29f1+sip+3+aa6d1466 Call-ID: 8791dbd3855eeafd484f397de6e2f76e(a)carrier.peering.telecom.im From: "+44792498881474" <sip:+441444680332@es132y$}-9>.8n?~9,*%(;zyk393;7e&C^NRone>;tag=carrier.peering.telecom.im+3+863d20a3+1b9801d2 To: "+441624111111" <sip:+441624111111@8.8.8.8:5080;user=phone>;tag=6rrtgNFQDNrFF CSeq: 1 ACK Contact: <sip:4.4.4.4:5060> Content-Length: 0 Max-Forwards: 67 Causing FreeSWITCH to not recognise the request, and therefore not send an ACK. There are no rules set against the ACK processing. Has anyone seen this before? We dont know when it started happening which doesnt help, I will look to setup debug on test environment but just wondered if this is an issue thats been seen before? Many thanks in advance. Jon

2 5

Event when acc is written?
by Tobias 25 Jan '17

25 Jan '17

Hi! I'm wondering if there is a way for the acc-module to notify/send an event to Kamailio (so I can pick it up from the config, similar to event_route) once the acc-module has performed a write? As an example I'm using acc and db-mysql to store acc data in MySQL, but instead of having a service polling the MySQL for new acc's written I'd like to make Kamailio send an event once an acc-row has been written. I realize this could most likely be done by using MySQL triggers "ON INSERT", but I'm wondering if there is another (reliable) way to do this directly from within Kamailio? I haven't tested it, but I would guess that if I tried to use for example event_route[dialog:end] there could be times where the event would hit Kamailio before the acc-module have written to missed_calls? Thanks! /Tobias

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