- sr-users - kamailio.org

question about replace_body_all
by Wilkins, Steve 07 Dec '18

07 Dec '18

Hello, I am attempting to change the Message Body of an SIP message using replace_body_all; I want to replace all attributes that start with rtcp. Here is my code => replace_body_all("rtcp:{1}[0-9]{5,}", newstring), and I call msg_apply_changes() after this call. Note that I verified that the regex is working using search. The problem is that nothing gets changed. However, what I do get is a new string at the bottom of the Message Body with my newstring concatenated three times, which happens to be the number of times that rtcp occurs in the Message Body. Note, I am doing this so that I can add the IP address to the rtcp attribute because this is required by a Provider. Any ideas? Thank you, -Steve

3 2

Dialog getting disconnected as soon as call is ended
by Subham Singhal 07 Dec '18

07 Dec '18

Hello, I am new to kamailio. I have been trying to integrate kamailio with asterisk. I am able to send INVITE and get RTP packets as well, but I am unable to receive BYE from asterisk. Call is getting terminated through following websocket error. "WebSocket connection to 'wss://172.16.16.169:5061/' failed: One or more reserved bits are on: reserved1 = 1, reserved2 = 0, reserved3 = 0." -- Thanks and Regards Subham Singhal

2 1

Auth_xkeys module - freeing already freed pointer
by José Seabra 06 Dec '18

06 Dec '18

Hello there, I'm constantly getting the following message when i'm using module auth_xkeys. CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7f6310987400), called from core: core/data_lump.c: free_lump(466), first free core: core/data_lump.c: free_lump(466) - ignoring Apparently kamailio only throw this message but i would like to know if its a normal protection from kamailio or it should be fixed? Thank you for the support Regards -- José Seabra

3 5

Assistance with re.subst & newline
by Mark Hall 06 Dec '18

06 Dec '18

Hi, I am using rtpengine to proxy audio to/from media servers. The outbound calls are being rejected by a couple of my carriers (although accepted by others), and the closest that I have been able to find for the reason is the inclusion of "a=rtcp:xxxxx" in the SDP generated by rtpengine. To test further, I am trying to remove this line. Rtpengine itself doesn't have any options to remove it, so I am left with manipulating the generated SDP in Kamailio. A sample of generated SDP is: a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-16 a=sendrecv a=rtcp:30939 s=My Media Server I am using rtpengine_manage("RTP/AVP ICE=remove"); to generate the SDP to a variable ($avp(sdp)) Now, I am trying to just remove the whole a=rtcp line with re.subst $avp(sdp)=$(avp(sdp){re.subst,|a=rtcp:[0-9]*\R||s}); *** Tried / as delimiter also, but | looks cleaner *** This should match the line and any CR/LF characters - BUT in results, nothing is changed (various online Regex testers show this match to be correct - I tried \r\n also). If I do the same but without the \R, then the line is removed but not the CR/LF, leaving a gap in the SDP - which is invalid. $avp(sdp)=$(avp(sdp){re.subst,|a=rtcp:[0-9]*||s}); Gives: a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-16 a=sendrecv s=My Media Server Can anyone see where I am going wrong, or suggest an alternative? Kind Regards, Mark Hall

3 3

New features: dialog - list dialogs via rpc using matching expressions
by Daniel-Constantin Mierla 06 Dec '18

06 Dec '18

Hello, being discussed and requested couple of times via lists, irc channel and discussions at events world wide, I just want to send a short note to announce that yesterday I pushed a new rpc command to dialog module that returns dialog records that match on conditions applied to request uri, from uri, to uri or callid fields. The filtering can be done with string comparison, regular expression or starts-with (prefix) matching. See more details at: - https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.list_… It should enable listing dialogs only for specific users, caller or callee, without dumping all dialogs. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 0

set_advertised_address
by Kjeld Flarup 06 Dec '18

06 Dec '18

Hello I have a PBX behind NAT. Thus I advertise the public IP, and forwards the port to my PBX. listen=LOCALIP:5070 advertise EXTERNALIP:5070 Now clients can connect to the PBX from the Internet. And also inside the LAN, because I have enabled NAT loopback. However some customers sysadmins complains that NAT loopback is a security risk. I have not been able to find any exploits of this, but the sales and support people asks if it is possible to remove this NAT loopback requirement. I could look at $rd and if it is local, then I could advertise LOCALIP. I found set_advertised_address("LOCALIP"); set_advertised_address however only seems to modify the latest Via header, not the Record-route, and audio neither works. Could I do something to make this work, or is it a dead end? -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

4 11

Dispatcher clear dst_avp
by Grant Bagdasarian 05 Dec '18

05 Dec '18

Hello, I'm trying to use multiple destination sets which may include the same destination. For instance: SET ID: 1 sip:10.0.0.1:5060 sip:10.0.0.2:5060 SET ID: 2 sip:10.0.0.1:5060 Whenever a new SIP INVITE is sent to the Kamailio instance it will use SET ID 1. If then for some reason this call attempt fails, failure_route will trigger and SET ID 2 will be used. At first glance this might not make any sense, but there is some business logic behind this, but it's not relevant for this case. For some reason, the dst_avp variable still contains the destinations of SET ID 1 in failure_route. Is there a built-in way to reset this variable for every call to ds_select_domain(), such that only the destinations of passed SET ID will be loaded into the avp? I tried setting the avp to $null before calling ds_select_domain, but that had some undesirable effects. The reason I'm asking this is the destinations the call is being sent to are Freeswitch instances, which will return a 482 Request Merged when the same INVITE is sent to the same Freeswitch within 4 seconds (default T4 timer). I'm trying to match the previously selected destination in failure_route with the next destination and skip if they're the same, or stop processing when they are the same and no more destinations are available. I know I can mess with the T4 timer in Freeswitch and perhaps even sleep in failure route to deal with the 482 failure, but I rather fix this in a normal way. Regards, Grant Bagdasarian Senior Developer

2 2

xlog issue
by Soltanici Ilie 04 Dec '18

04 Dec '18

Hello, I have a strange situation by using xlog module. I don't know for what reason I'm not receiving all logs generated by kamailio. This is the configuration which I'm using in kamailio.cfg: log_facility=LOG_LOCAL0 log_name="kamailio" log_prefix="{$rm ($mt) | Seq=$cs | Source IP=$si ($proto) | Call ID=$ci} " # ----- xlog ----- modparam("xlog", "buf_size", 8192) modparam("xlog", "long_format", 0) modparam("xlog", "force_color", 1) modparam("xlog", "log_colors", "L_ERR=cr;L_WARN=px,L_INFO=px") This is rsyslog configuration file: local0.* /var/log/kamailio/kamailio.log Kamailio is running under kamailio user, permission for the log file are as shown below: -rwxrwxr-x 1 kamailio kamailio 252885132 Dec 2 14:51 /var/log/kamailio/kamailio.log In request_route block this is the first line: xlog("L_INFO","[START ROUTING]-(Source IP=$si:$sp/Destination IP=$Ri:$Rp)\n"); The problem is that not every request is logged in the log-file. For example, some "INVITES" requests I can find in the log file, but some of them - I cannot, even that in sngrep I see the request and the call is successfully processed by kamailio. Also, for some requests i can see only partial data, not full call-flow as it supposed to be (for ex. i see only BYE requests, or ACK response instead of full call flow). Does someone have the same issue? If you don't - how are you dealing with kamailio log files? I'm thinking to send them to the central ELK stack, but if I have such problems by saving them locally - I don't see any reason to send them elsewhere. I may think that the problem could be in rsyslog itself, but how can I troubleshoot that? The traffic on the server is not very high - 30-50 concurrent calls. As a storage i'm using an SSD disk and xfs filesystem. Load on the disk - according to iostat/iotop - is minimum. OS: CentOS Linux 7 (Core), Kamailio: 5.2.0 (x86_64/linux) 535e13 Thank You.

3 9

presence/mwi issue
by Andreas Granig 04 Dec '18

04 Dec '18

Hi, Currently playing with presence_mwi in kamailio 5.1, and it appears the issue reported by Juha two years ago at https://lists.kamailio.org//pipermail/sr-dev/2016-September/036609.html seems to have appeared again somewhere along the way. I’m sending a PUBLISH with message-summary event type, however I’m getting this error: ERROR: presence [presentity.c:283]: check_if_dialog(): failed to parse xml document As said in the old thread, it’s not an XML document, so it shouldn’t try to treat it as such. Any ideas? Best, Andreas

2 8

Dispatcher with a naptr address as destination
by José Seabra 04 Dec '18

04 Dec '18

Hello all, I'm trying to use a naptr address as a dispatcher destination and i'm facing some issues here. When i configure a naptr address in dispatcher table, like: *1:1:sip\:proxy-registrar-ci1.fuze.domain:0:0:'':''* I see kamailio making a dns query of type A, but since that this name doesn't have any A record configured in dns, kamailio didn't get any positive reply from it. Should't kamailio make dns query by this order? 1. NAPTR 2. SRV 3. A I noticed also that if the name configured in dispatcher table is resolved successfully with A record from dns, then kamailio tries to make a query to NAPTR and then SRV services for the exactly same name used for query previously done(type A), but only if it gets a positive answer from dns to the A type query. I'm using kamailio version 5.1.0 with the following dns parameters configured: ####### DNS Parameters ######### dns = no rev_dns = no dns_try_ipv6 = no dns_retr_time = 1 dns_retr_no = 1 dns_cache_del_nonexp = no dns_cache_flags = 1 dns_cache_gc_interval = 120 dns_cache_init = 1 dns_cache_mem = 1000 dns_cache_negative_ttl = 60 dns_try_naptr = yes use_dns_cache = on use_dns_failover = on dns_srv_lb = yes Is there something wrongly configured here that can make kamailio behaves like that? Thank you. Regards -- Cumprimentos José Seabra

4 8

Issue with ACK's
by Dan Quinney 30 Nov '18

30 Nov '18

Hi all, I’m having some issues with ACK’s. Specifically, when an ACK reaches Kamailio from either Asterisk or an endpoint, Kamailio is adding a second Via header - which isn’t valid as per RFC 3261 (17.1.1.3); “The ACK MUST contain a single Via header field, and this MUST be equal to the top Via header field of the original request." Endpoints are registered with Kamailio but INVITES are routed via Asterisk for dial plan and media reasons. I’m running fairly bog-standard Kamailio and Asterisk installs. I’ve attached a sample PCAP for if anyone has any suggestions. Thanks. Dan

2 2

Issue with ACK's
by Dan Quinney 30 Nov '18

30 Nov '18

Hi all, I’m having some issues with ACK’s. Specifically, when an ACK reaches Kamailio from either Asterisk or an endpoint, Kamailio is adding a second Via header - which isn’t valid as per RFC 3261 (17.1.1.3); “The ACK MUST contain a single Via header field, and this MUST be equal to the top Via header field of the original request." Endpoints are registered with Kamailio but INVITES are routed via Asterisk for dial plan and media reasons. I’m running fairly bog-standard Kamailio and Asterisk installs. I’ve attached a sample PCAP for if anyone has any suggestions. Thanks. Dan

1 0

Kamailio v5.2.0 Released - new major version is out
by Daniel-Constantin Mierla 30 Nov '18

30 Nov '18

Kamailio v5.2.0 is out – it comes with 6 new modules and a consistent group of improvements touching more than 100 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-2-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.2.0 brings more flexibility and optimizations for KEMI interpreters (including two new interpreters for Python3 and Ruby), major enhancements to load balancer and tls implementations, new variables and lots of other new features. Enjoy Kamailio v5.2.0! 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 2019 - http://www.kamailioworld.com

2 2

Response VIA Header
by Soltanici Ilie 29 Nov '18

29 Nov '18

Hi, We have a Kamailio Instance running on Public IP Address, one of our ISP cannot send ACK back to us because we are sending 100 Trying - without "received" parameter. Is there any way in Kamailio to force a "received" parameter in the via header for 100 Trying Message? SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK8tgchsfjaKJdf.1;rport=5060 Thank you.

2 1

Forward() not handles Route headers
by Yuriy Gorlichenko 29 Nov '18

29 Nov '18

We are building some stateless solution and found that in case of using forward() kamailio sends traffic only via URI or $du variable. It never removes itself from route headers. Is this is the truth? Or I need to add some other module to handle it? (rr already added)

1 1

KAMAILIO 4.2 - RTPPROXY config confirmation - error message from systemctl status kamailio.service -l (ERROR: <script>: NATMANAGE coee)
by yboujraf＠by-research.be 28 Nov '18

28 Nov '18

Dear All, I am still using Kamailio 4.2 with Debian Jessie without any issues. RTPPROXY - CONFIG > # Defaults for rtpproxy > # The control socket. > #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" > # To listen on an UDP socket, uncomment this line: > CONTROL_SOCK=udp:127.0.0.1:7722 > # Additional options that are passed to the daemon. > > # log in /var/log/kamailio.log > EXTRA_OPTS="-l ip_public/192.168.40.101 -m 11000 -M 12000 -d DBUG:LOG_LOCAL0" WHEN I MAKE AN INVITE FROM 801 TO 803, I HAVE THIS ERROR MESSAGE : KAMAILIO SERVICE (ERROR: <SCRIPT>: NATMANAGE COEE) root@sipsecure:~# systemctl status kamailio.service -l ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio) Active: active (running) since Tue 2018-11-20 22:34:31 CET; 40s ago Process: 1820 ExecStop=/etc/init.d/kamailio stop (code=exited, status=0/SUCCESS) Process: 1830 ExecStart=/etc/init.d/kamailio start (code=exited, status=0/SUCCESS) CGroup: /system.slice/kamailio.service ├─1846 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1848 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1849 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1850 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1851 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1852 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1853 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1854 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1855 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1856 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1858 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1860 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1863 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1865 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1867 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1869 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1871 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1873 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio └─1875 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1853]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1854]: ERROR: <script>: NATMANAGE M=INVITE OU=<null> RURI=<null> RD=<null> F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=<null> IP=192.168.0.207 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1854]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1855]: ERROR: <script>: NATMANAGE M=INVITE OU=<null> RURI=<null> RD=<null> F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=<null> IP=192.168.0.207 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1855]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1852]: ERROR: <script>: NATMANAGE M=INVITE OU=<null> RURI=<null> RD=<null> F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=<null> IP=192.168.0.207 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1852]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1852]: NOTICE: acc [acc.c:317]: acc_log_request(): ACC: transaction answered: timestamp=1542749703;method=INVITE;from_tag=4d488551;to_tag=2006d156;call_id=fjpUwRGm0u--3RvvB4fZ7Q..;code=200;reason=OK;src_user=801;src_domain=192.168.40.101;src_ip=192.168.0.229;dst_ouser=803;dst_user=803;dst_domain=192.168.0.207;user_agent=Z 5.2.19 rv2.8.99 Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1853]: ERROR: <script>: NATMANAGE M=ACK OU=sip:803@192.168.0.207:5060;alias=192.168.0.207:5060~1 RURI=sip:803@192.168.0.207:5060 RD=192.168.0.207 F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=192.168.0.207 IP=192.168.0.229 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1853]: ERROR: <script>: NATMANAGE coee Don't understand why the error??? RTPPROXY SERVICE : (RTPPROXY IS DOING HIS JOB PERFECTLY AND GET AUDIO ON BOTH DIRECTION) root@sipsecure:~# systemctl status rtpproxy.service -l ● rtpproxy.service - LSB: RTP Proxy Loaded: loaded (/etc/init.d/rtpproxy) Active: active (running) since Tue 2018-11-20 22:13:58 CET; 22min ago Process: 443 ExecStart=/etc/init.d/rtpproxy start (code=exited, status=0/SUCCESS) CGroup: /system.slice/rtpproxy.service └─1137 /usr/bin/rtpproxy -s udp:127.0.0.1 7722 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 212.76.231.102 192.168.40.101 -m 11000 -M 12000 -d DBUG LOG_LOCAL0 Nov 20 22:34:31 sipsecure rtpproxy[1137]: DBUG:doreply: sending reply "1875_11 1 " Nov 20 22:35:00 sipsecure rtpproxy[1137]: DBUG:handle_command: received command "1853_8 USEEc106,9,3,111,0,8,97,110,112,98,101,100,99,102 fjpUwRGm0u--3RvvB4fZ7Q.. 192.168.0.229 11000 4d488551;1" Nov 20 22:35:00 sipsecure rtpproxy[1137]: INFO:handle_command: new session fjpUwRGm0u--3RvvB4fZ7Q.., tag 4d488551;1 requested, type strong Nov 20 22:35:00 sipsecure rtpproxy[1137]: INFO:handle_command: new session on a port 11496 created, tag 4d488551;1 Nov 20 22:35:00 sipsecure rtpproxy[1137]: INFO:handle_command: pre-filling caller's address with 192.168.0.229:11000 Nov 20 22:35:00 sipsecure rtpproxy[1137]: DBUG:doreply: sending reply "1853_8 11496 192.168.40.101 " Nov 20 22:35:03 sipsecure rtpproxy[1137]: DBUG:handle_command: received command "1852_8 LSEEc106,9,3,111,0,8,97,110,112,102,98,101,100,99 fjpUwRGm0u--3RvvB4fZ7Q.. 192.168.0.207 11112 4d488551;1 2006d156;1" Nov 20 22:35:03 sipsecure rtpproxy[1137]: INFO:handle_command: lookup on ports 11496/11402, session timer restarted Nov 20 22:35:03 sipsecure rtpproxy[1137]: INFO:handle_command: pre-filling callee's address with 192.168.0.207:11112 Nov 20 22:35:03 sipsecure rtpproxy[1137]: DBUG:doreply: sending reply "1852_8 11402 192.168.40.101 " ??? Question : for rtpproxy config, do I need to add the natted ip address (NAT 1:1) or the real public address (where ip phone needs to assign as domain)? I am confused. Thanks a lot in advance Best Regards, Youssef

3 6

Issue sending SIP OPTIONS request to multiple contacts
by Dan Quinney 28 Nov '18

28 Nov '18

Hi, When using Kamailio v5.2.0-rc2 (x86_64/linux) 9e729b and setting the below nathelper module parameters it appears only the first contact is sent a SIP OPTIONS request. modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@mydomain <sip:ping@mydomain>.com") modparam("nathelper", "keepalive_timeout", 90) Reading the release notes for v5.2 what I’m doing *should* be possible, but as the functionality has been recently added I’m assuming that it might be a bug. Any assistance on this would be great. Thanks, Dan

2 1

Releasing Kamailio v5.2.0
by Daniel-Constantin Mierla 28 Nov '18

28 Nov '18

Hello, I am consider to release v5.2.0 (the first stable version out of branch 5.2) next week, likely on Wednesday, Nov 28, 2018. It still allows a bit more than a week of testing as well as well time to prepare the online resources for it (documentation, wiki pages, upgrade guidelines, etc...). If there is any issue you are of and not yet reported to github bug tracker, do it as soon as possible to give it a chance to be fixed in time for the next major release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 2

Error opening Kamailio's FIFO
by Markus 28 Nov '18

28 Nov '18

Hi list, when I run "/usr/sbin/kamctl address reload" I get: ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. In /etc/kamailio/kamailio.cfg I have: loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") But there's no fifo in /tmp. What am I doing wrong? kamailio-4.4.4-1.1.x86_64 on CentOS Linux release 7.2.1511 (Core). Thank you! Markus

3 8

[CFP] FOSDEM 2019, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 28 Nov '18

28 Nov '18

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2019 takes place 2-3 February 2019 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time Communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Sunday, 3rd of February 2019. The lounge will be present for both days. To discuss the dev-room and lounge, please join the Free RTC mailing list: http://lists.freertc.org/mailman/listinfo/discuss To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 2nd of December. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". https://penta.fosdem.org/submission/FOSDEM19/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2019/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 3rd of November. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators based on the number of received proposals. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 2nd of February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 1st of February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: http://lists.freertc.org/mailman/listinfo/discuss Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: http://lists.freertc.org/mailman/listinfo/discuss The dev-room administration team: Saúl Ibarra Corretgé <s(a)saghul.net> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro> Guus der Kinderen <guus.der.kinderen(a)gmail.com>

2 1

Dialiog and dispatcher
by David Villasmil 27 Nov '18

27 Nov '18

Hello guys, I'm testing the dispatcher with the dialog module. After making several test calls, which are properly routed, the number of dialogs stay there for a while, preventing new calls from going to those destinations: "SET": { "ID": 101, "TARGETS": [{ "DEST": { "URI": "sip:1.2.3.4:5080", "FLAGS": "AP", "PRIORITY": 0, "ATTRS": { "BODY": "weight=100;maxload=10;duid=bedd39c7-6c73-4477-ae7d-77f8cdc96797", "DUID": "bedd39c7-6c73-4477-ae7d-77f8cdc96797", "MAXLOAD": 10, "WEIGHT": 100, "RWEIGHT": 0, "SOCKET": "" }, "RUNTIME": { "DLGLOAD": 10 } } }] The calls have been down for several seconds, but the "DLGLOAD" parameter doesn't go down, even though /usr/local/kamailio5/sbin/kamctl', 'dialog', 'show Output the correct amount of dialogs. I set the dialog handler just before relaying out: t_on_failure("RTF_DISPATCH"); set_dlg_profile("gateways","$du"); route(RELAY); And the module is supposed to remove them automatically. Am i doing something wrong? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

dispatcher: Get gateway current calls
by David Villasmil 27 Nov '18

27 Nov '18

Hello guys, is there any way of printing out the selected gw current calls? i.e.: if(ds_select_dst("1", "4")) { xlog("L_ERR", "[DISPATCH] $ci: ds_select_routes was succesful (du: <$du>)\n"); ****** PRINT GWs CALLS ******* } else { xlog("L_ERR", "[DISPATCH] $ci: did NOT find a gateway to use!\n"); send_reply("404", "No destination"); exit; } xlog("L_DBG", "[DISPATCH] $ci: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

Re: [SR-Users] "kamdbctl create" failing with MySQL 8
by Jonathan Tyler 26 Nov '18

26 Nov '18

I just had the same thing happen to me. I installed Kamailio via Apt, and I'm using a dedicated MySQL server, with non-root admin credentials. Previously today I installed it using a local MySQL 5.7 database with no issues. Kamailio Host: Kamailio 5.1.2 (Ubuntu 18.04) MySQL Host: MySQL 8.0.13 (Windows Server 2012 R2) It creates the database, then fails at granting permissions. Accounts were created and privileges flushed before executing. > root@kamailio:~# kamdbctl create > INFO: creating database kamailio ... > mysql: [Warning] Using a password on the command line interface can be insecure. > INFO: granting privileges to database kamailio ... > mysql: [Warning] Using a password on the command line interface can be insecure. > ERROR 1064 (42000) at line 1: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'IDENTIFIED BY '<password>'' at line 1 I turned on query logging on the database host but didn't see any errors there, or in the error log itself. > 2018-11-21T01:58:15.045586Z 8 Connect <user>@kamailio on using SSL/TLS > 2018-11-21T01:58:15.046239Z 8 Query select @@version_comment limit 1 > 2018-11-21T01:58:15.046764Z 8 Query CREATE DATABASE kamailio CHARACTER SET utf8mb4 > 2018-11-21T01:58:15.189109Z 8 Quit > 2018-11-21T01:58:15.208476Z 9 Connect <user>@kamailio on using SSL/TLS > 2018-11-21T01:58:15.209019Z 9 Query select @@version_comment limit 1 > 2018-11-21T01:58:15.210114Z 9 Quit I've tried with the default MySQL 5.7 Client on Ubuntu, and upgrading it to 8.0.13, but got the same result. Unfortunately I don't know how to see the output of the SQL Statement being sent (without installing 5.7), or I'd try to run it manually. - Jonathan

3 5

issue tracking dialogs with db_redis in kamailio 5.2
by David Escartín 26 Nov '18

26 Nov '18

hello all using version 5.2.0-rc0 of kamailio, we are trying to track the dialogs of dialog module using db_redis module with this setup #!define DBURL_DIALOG "redis://127.0.0.1:6390/7" # ----- dialog params ----- modparam("dialog", "db_url", DBURL_DIALOG) modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 16384) modparam("dialog", "dlg_flag", 31) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "timer_procs", 1) modparam("dialog", "db_update_period", 1) # ----- db_redis params ----- modparam("db_redis", "schema_path", "/usr/local/kamailio/share/kamailio/db_redis/kamailio") modparam("db_redis", "keys", "dialog=entry:hash_entry,hash_id,callid") modparam("db_redis", "keys", "dialog_vars=entry:hash_entry,hash_id,dialog_key,dialog_value") and we see it works and if a kamailio instance resets in the middle of a dialog it recovers all the dlg_vars and values because we have the db_skip_load to 0 in dialog module. But we have noticed that the redis is being written with the information of the dlg_var assigned until that moment when we get a provisional response with SDP, and we dom see a set on the redis with the dialog entry and id untill call is connected. Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: INFO: Call-ID 1-1511(a)9.7.8.1: Status 180 Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog' while performing update Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' on the redis (we only see hmset for dialog_vars key) 1542374575.721816 [7 127.0.0.1:56858] "HMSET" "dialog_vars:entry::13725:16494:_uac_funew:sip:1231215423@voda.interconnect.manxtelecom.im" "hash_entry" "13725" "hash_id" "16494" "dialog_key" "_uac_funew" "dialog_value" "sip:1231215423@voda.interconnect.manxtelecom.im" 1542374575.722001 [7 127.0.0.1:56858] "HMSET" "dialog_vars:entry::13725:16494:_uac_fu:sip:anonymous@voda.interconnect.manxtelecom.im" "hash_entry" "13725" "hash_id" "16494" "dialog_key" "_uac_fu" "dialog_value" "sip:anonymous@voda.interconnect.manxtelecom.im" .... Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: INFO: Call-ID 1-1511(a)9.7.8.1: Status 200 Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog_vars' while performing update Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 2 is 'dialog_key' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog' while performing update Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog' while performing update Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: INFO: Call-ID 1-1511(a)9.7.8.1: ACK received in A-Leg on the redis 1542374585.810520 [7 127.0.0.1:56882] "SCAN" "0" "MATCH" "dialog:entry::*" "COUNT" "1000" 1542374585.810672 [7 127.0.0.1:56882] "EXISTS" "dialog:entry::13725:16494:1-1511@9.7.8.1" 1542374585.810683 [7 127.0.0.1:56882] "HMGET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "hash_entry" "hash_id" 1542374585.810776 [7 127.0.0.1:56882] "HMSET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "state" "4" "timeout" "1542377585" "caller_cseq" "1" "callee_cseq" "0" "caller_contact" "sip:0000123456@9.7.8.1:5084" "callee_contact" "sip:50622959898@9.70.1.52:5060;transport=udp" 1542374585.810899 [7 127.0.0.1:56882] "SCAN" "0" "MATCH" "dialog:entry::*" "COUNT" "1000" 1542374585.810969 [7 127.0.0.1:56882] "EXISTS" "dialog:entry::13725:16494:1-1511@9.7.8.1" 1542374585.810977 [7 127.0.0.1:56882] "HMGET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "hash_entry" "hash_id" 1542374585.811059 [7 127.0.0.1:56882] "HMSET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "state" "4" "timeout" "1542377585" "caller_cseq" "1" "callee_cseq" "0" "caller_contact" "sip:0000123456@9.7.8.1:5084" "callee_contact" "sip:50622959898@9.70.1.52:5060;transport=udp" i don't see a redis set order to update the dialog entry key before the 200OK is received, to create and update state 1, etc. when created with dlg_manage command could you please check if we might be missing something? thanks a lot and regards david

2 3

Kamailio "Presence Server" configuration
by denislav.tsonev＠musala.com 26 Nov '18

26 Nov '18

Hello, I'm trying to configure Kamailio to be "Presence Server". Simple scheme: two phones (one "hard phone" Grandstream GXP1620 and "Soft phone" MicroSIP) and Kamailio server with default configuration. So when "MicroSIP" send SIP SUBSCRIBE to Kamailio, server response with SIP NOTIFY but without XML part. Then when hard phone change his status from "Ready" to "Ringing" and then "On-The-Phone" Kamailio is not sending any SIP NOTIFY messages. Is this normal behavior or I missed something? (In DB I have records in "active_watchers" and "watchers" tables but not in "presentity" table) ====================================================================== ====================== Module configuration: ============================= ====================================================================== #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:10.0.6.123:5060") # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif ====================================================================== ============================================================== ======================== Route: ================================ ============================================================== route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } ================================================================ Thanks in advance Denislav Raychev Tsonev | Integration and Infrastructure Engineer | Musala Soft JSC www.musala.com | t: +359 2 969 58 21 | m: +359 878 270 965 | f: +359 2 969 58 22 World Trade Center, 36 Dragan Tsankov blvd., Sofia 1057, Bulgaria

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