- sr-users - kamailio.org

Subscribe Reinvites totag
by Soltanici Ilie 27 Sep '17

27 Sep '17

Hi All, I have system setup where there are few Asterisk servers behind Kamailio. Everything is working properly, the issue only is with the Incoming Subscribe's with to-tag from endpoints. Below you can see the invite: ---------------------------------------------------------------------------------------------------------- 2017/09/27 19:40:29.995598 192.168.2.33:5060 -> 10.25.23.1:5060 SUBSCRIBE sip:10000211@10.25.23.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.33:5060;branch=z9hG4bK-fc65f67d From: "Cisco" <sip:211@10.25.23.1>;tag=258a856c69c288 To: <sip:10000211@10.25.23.1>; tag=02cca2b6-1d9b-4edc-b1de-fdd78f8f9ab0 Call-ID: 4ce04a70-6128897b(a)192.168.2.33 CSeq: 1003 SUBSCRIBE Max-Forwards: 70 Route: <sip:10.25.23.1;lr;r2=on;ftag=258a856c69c288;nat=yes>, <sip:10.1.1.101;lr;r2=on;ftag=258a856c69c288;nat=yes> Proxy-Authorization: Digest username="211",realm="10.25.23.1",nonce="Wcvq0VnL6aUbQ3c+nZa5xbh8KMVoOwAq",uri="sip:10000211@10.25.23.1",algorithm=MD5,response="155f68d66b3c6ad035f7db1ec5732020" Contact: "Cisco" <sip:211@192.168.2.33:5060> Accept: application/dialog-info+xml Expires: 1800 Event: dialog User-Agent: Cisco/SPA504G-7.6.2 Content-Length: 0 ---------------------------------------------------------------------------------------------------------- Receiving such Invite Kamailio are routing to RELAY route (from WITHINDLG) and showing such error: ERROR: rr [loose.c:107]: find_first_route(): failed to parse Route HF It seems that Kamailio cannot find such Dialog, but why is happening? where is it destroyed? How can i solve this? Route (WITHINDLG) is as default, below you can see it as well. ----------------------------------------------------------------------------------------------------------- route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { if ( is_method("ACK") ) { route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { exit; } } sl_send_reply("404","Not here"); } exit; } ----------------------------------------------------------------------------------------------------------- Thank You.

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Next IRC devel meeting
by Daniel-Constantin Mierla 27 Sep '17

27 Sep '17

Hello, to sync properly for the next major release (v5.1.0) and ongoing development, I propose an IRC devel meeting for next week, on Wednesday, Sep 27, 2017. An alternative would be the following day, Sep 28, or if there are many devs that want to attend and cannot do it these days, we can look at another date. Just propose new day and time. I created a wiki page to collect the topics that are wanted to be discussed: - https://www.kamailio.org/wiki/devel/irc-meetings/2017b Fell free to add there or reply to the mailing list with what you think it is relevant to discuss. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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dialog:active_dialogs counter is steadily increasing
by Andrew Pogrebennyk 27 Sep '17

27 Sep '17

Hi, using kamailio 4.4.5, we have noticed that the active_dialogs counter is not accurate and there seems to be some kind of "leak" present at least on higher loaded systems. For example, here active_dialogs was reporting just a bit over 1000 dialogs, but the numbers of dialogs confirmed and not just do not sum up together to that value: # grep -c 'state:: 1' dialogs_20170919_1712.txt 5 # grep -c 'state:: 2' dialogs_20170919_1712.txt 62 # grep -c 'state:: 3' dialogs_20170919_1712.txt 0 # grep -c 'state:: 4' dialogs_20170919_1712.txt 531 # grep -c 'state:: 5' dialogs_20170919_1712.txt 27 What could be the possible reasons or issues that lead to that? We are thinking now about switching statistics collection to: kamctl fifo dlg_list | grep -c 'state:: 4' or reading the count directly from the database, as it seems to only write the dialogs with state 4 to mysql and those numbers seem OKish. Any ideas? Regards, Andrew

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How to do msg_apply_changes with Kemi python version 5.0.x
by Muhammad Zaka 27 Sep '17

27 Sep '17

Hi We are experiencing issue with python script KSR. textopsx.msg_apply_changes using branch 5.0.2: - Example Code: KSR.textopsx.msg_apply_changes() Error: Sep 22 15:25:19 sbc99-kam-dev /usr/sbin/kamailio[23141]: PYT {INVITE}: ERROR: app_python [python_support.c:142]: python_handle_exception(): python_exec2: Unhandled exception in the Python code:#012Traceback (most recent call last):#012 File "/etc/kamailio/kamailio-kemi-python.py", line 64, in ksr_request_route#012 if self.ksr_route_reqinit(msg)==-255 :#012 File "/etc/kamailio/kamailio-kemi-python.py", line 173, in ksr_route_reqinit#012 KSR.textopsx.msg_apply_changes()#012AttributeError: 'module' object has no attribute 'textopsx' Is anybody have any idea how to workaround and apply message changes in python? Many Thanks Regards Muhammad Zaka

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topos and record-route headers
by Giuseppe Pandolfi 27 Sep '17

27 Sep '17

Hi, I'm trying to use topos module but I have a problem with record-route header in re-invite for on hold. Uac is a subscriber of sipwise platfrom (vm open source version); my kamailio version is: /# kamailio -v// //version: kamailio 5.1.0-dev7 (ppc/linux) // //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_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES// //ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB/ and uas is a subscriber of kamailio. The call flow is: UAC Sipwise Kamailio UAS | | (topos active) | |-------- Invite -------->|---------- (1)Invite ----->|------- (2)Invite ----->| |<------- 180/200 ------|<--- (4)180/(4)200 ----|<------- 180/200 ------| |---------- Ack --------->|----------- (5)Ack ------>|-------- (6)Ack ------->| |--- Invite (hold) ----->|-- (7)ReInvite(hold)-->|-- (8)Invite (hold)--->| |<--------- 200 ----------|<--------- (9)200 -------|<--------- 200 ----------| | corrupt | corrupt | | |---- No Ack Sent-------|---- No Ack Sent-------|---- No Ack Sent-------| *The problem:* the 200 ok of re-invite is corrupt by topos, it hasn't via header and contact isn't modified, then the uac can't send the ack. In the kamailio trace I see the ERROR below: *E!: U 22/09/2017 10:24:19:880 proxy0: ERROR: topos [tps_storage.c:1254]: tps_db_update_dialog(): no valid dlg uuid* *The questions are:* Can I use topos for this scenario? Is it possible that topos have bug when receive a re-invite with record-route headers? Infact, in simple scenario without record-route (call and hold between tow subscriber) everything works! *My opinion is*: when topos receive a request with record-route generates from record-route the new aleg and bleg tags (atpsh-xxxxx, btpsh-yyyyyy) but don't save their in internal database if the request is internal at dialog. Then a reintive whit record-route headers, generate new aleg/bleg tags but they aren't in database, then the response 200 ok it isn't elaborate. below messages details, attached kamailio log and pcap trace. tanks in advance for support, Giuseppe* * * * *(1)Invite:* INVITE sip:100@192.168.110.79:5060;transport=udp SIP/2.0 Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.6c8d17d29887a0078c2cb6aca8684a27.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKPa3L8art;rport=5080 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79> CSeq: 10 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri P-Asserted-Identity: <sip:390735594284@192.168.110.132> Content-Type: application/sdp Content-Length: 276 Contact: <sip:ngcp-lb@192.168.110.132:5060;ngcpct=7369703a3132372e302e302e313a35303830> [sdp] *(2)Invite:* INVITE sip:100@192.168.1.73:5080;line=4d3148ee1d889aa SIP/2.0 Via: SIP/2.0/UDP 192.168.110.79;branch=z9hG4bK1a43.c9b996baf80be68d288d2de06cd9b212.0 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79> CSeq: 10 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri P-Asserted-Identity: <sip:390735594284@192.168.110.132> Content-Type: application/sdp Content-Length: 276 Contact: <sip:btpsh-59c55b4a-4fa6-1@192.168.110.79> [sdp] *(3) 180 Ringing* SIP/2.0 180 Ringing From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 CSeq: 10 INVITE User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.6c8d17d29887a0078c2cb6aca8684a27.0,SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKPa3L8art;rport=5080 Contact: <sip:atpsh-59c55b4a-4fa6-1@192.168.110.79> Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes>,<sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> *(4) 200 OK* SIP/2.0 200 OK From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 CSeq: 10 INVITE Content-Type: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 181 Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.6c8d17d29887a0078c2cb6aca8684a27.0,SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKPa3L8art;rport=5080 Contact: <sip:atpsh-59c55b4a-4fa6-1@192.168.110.79> Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes>,<sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> [sdp] *(5) ACK* ACK sip:atpsh-59c55b4a-4fa6-1@192.168.110.79 SIP/2.0 Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK1a43.4446035c867218358dbd848f773b5d09.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKspF2zas~;rport=5080 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 10 ACK Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Content-Length: 0 Contact: <sip:ngcp-lb@192.168.110.132:5060;ngcpct=7369703a3132372e302e302e313a35303830> *(6) ACK* ACK sip:100@192.168.1.73:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.110.79;branch=z9hG4bK1a43.b1cce71dc1b004e442faca53cd6d8994.0 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 10 ACK Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Content-Length: 0 Contact: <sip:atpsh-59c55b4a-4fa6-2@127.0.0.1> *(7) Re-Invite * INVITE sip:atpsh-59c55b4a-4fa6-1@192.168.110.79 SIP/2.0 Record-Route: <sip:192.168.110.132;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=6D91D5F4-59C53F38000E3F0A-D551C700;ngcplb=yes> Via: SIP/2.0/UDP 192.168.110.132;branch=z9hG4bK2a43.3723a1a3f343f1f1974a66f4a62f41a9.0 Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKf6peiap7;rport=5080 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 11 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Content-Type: application/sdp Content-Length: 276 Contact: <sip:ngcp-lb@192.168.110.132:5060;ngcpct=7369703a3132372e302e302e313a35303830> [sdp] *(8) Re-Invite* INVITE sip:100@192.168.1.73:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.110.79;branch=z9hG4bK2a43.a4e5f5baf85414d4471a1e817bdc1e00.0 From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 CSeq: 11 INVITE Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Content-Type: application/sdp Content-Length: 276 Contact: <sip:atpsh-59c55b4a-4fa6-3@127.0.0.1> [sdp] * **(9) 200 OK* SIP/2.0 200 OK From: <sip:390735594284@192.168.110.132>;tag=6D91D5F4-59C53F38000E3F0A-D551C700 To: <sip:100@192.168.110.79>;tag=856281851 Call-ID: OTkwMTdlNjEyYTQ2NWFiNTJkN2U1YTA2YWY1ZGI5YmE._b2b-1 CSeq: 11 INVITE Contact: <sip:100@192.168.1.73:5080> Content-Type: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 211 [sdp] ------------------------------------------------------------------------

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UACREG - answering to 401 on invite
by Jean Cérien 26 Sep '17

26 Sep '17

Hello I'm using Kamailio 5.0.1 With the UACREG module, I am registering to a remote provider. Register goes out, 401 back, Register goes out with nonce & co, OK Later, when I send an invite, the provider issues an 401 Unauthorized. I guess it expects me to resubmit an INVITE with the authentication data, but I dont see how to do that. Any help would be appreciated ! J.

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How to change SDP in 200 Ok reply
by Jurijs Ivolga 26 Sep '17

26 Sep '17

Hi, I have difficult times to edit SDP in 200 ok replies, I need to change RTP/SAVPF to UDP/TLS/RTP/SAVPF in all 200 Ok. Here is code snippet: onreply_route { if ( $rm == "INVITE" && status=="200") { if (search_body("a=fingerprint")) { if (search_body(" RTP/SAVPF ")) { xlog("L_INFO","200 OK DTLS call.\n"); replace_body_all(" RTP/SAVPF "," UDP/TLS/RTP/SAVPF "); } } } } What I'm doing wrong? I think my code is executed against SDP of Invite, but not 200 OK. Jurijs

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Best variant to share variables between two dialogs?
by Kristijan Vrban 26 Sep '17

26 Sep '17

Hi, based on Daniels example from: http://blog.miconda.eu/2010/01/best-of-new-in-kamailio-300-19-htable.html ... modparam("htable", "htable", "acalls=>size=8;") ... route { ... if(is_method("INVITE") && !has_totag()) { # a new call if($shtcv(ht=>^$fU$)>=3) { send_reply("403", "limit exceeded"); exit; } $sht(acalls=>$ci) = $fU; t_on_failure("NEW_INVITE"); } if(is_method("BYE")) { $sht(acalls=>$ci) = $null; } ... } failure_route[NEW_INVITE] { $sht(acalls=>$ci) = $null; } I use this htable variant to share data between two dialogs (store Contact from one dialog, and re-write other R-URI with it). Referenced by call-id: ... $avp(referenceID) = $ci + "-" + $avp(direction); $sht(a=>$avp(referenceID)) = $ct; ... ... $avp(referenceID) = $ci + "-" + $avp(direction); avp(newru) = $sht(a=>$avp(referenceID)); avp_subst("$avp(newru)", "/(<)(.*)(>)/\2/"); xlog("L_NOTICE", ">>>>>>>> ReWrite ru with:$avp(newru)\n"); $ru = $avp(newru); ... Is this the best method to share data between two dialogs? Because you have to take care yourself with failure_route and so on, to free (set back to $null) them. Or is there a more elegant way for that task? Best Kristijan

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Send 407 for Invite
by Logeshwaran G 26 Sep '17

26 Sep '17

Hi, How to send the 407 proxy authentication for Invite message? Input Please Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

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Memory allocation issue
by Logeshwaran G 26 Sep '17

26 Sep '17

Hi, I am having the below Error while executing the kamailio cfg file. Before an hour it executed successfully, There is no changes in cfg. loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(27575) : <core> [core/mem/shm.c:105]: shm_getmem(): could not attach shared memory segment: Cannot allocate memory 0(27575) : <core> [core/mem/q_malloc.c:1203]: qm_malloc_init_shm_manager(): could not initialize qm shm memory pool Too much qm shm memory demanded: 67108864 bytes 0(27575) CRITICAL: <core> [core/cfg.y:3414]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/config/kamailio_final.cfg, line 302, column 39: Can't initialize shared memory ERROR: bad config file (1 errors) Input please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

1 0

Docker image
by Sergey Safarov 25 Sep '17

25 Sep '17

Hello guys Yes i packaged kamailio to docker container <https://hub.docker.com/r/safarov/kamailio3/>. This container is ready to run kamailio in docker environment. Could you make review of this. Dockerfiles you can review at https://github.com/kamailio/kamailio/tree/master/pkg/kamailio/alpine_docker Build process logs you can review at https://hub.docker.com/r/safarov/kamailio3/builds/bl9rv6ygeubwnydh4an2tuc/ Sergey

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Astricon 2017
by Daniel-Constantin Mierla 25 Sep '17

25 Sep '17

Hello, Kamailio will be present at Astricon 2017 having a stand in the expo area (Orlando, FL, during October 3-5 -- more details at www.astricon.net). Should you participate at the event and have flyers/other materials with products/solutions/services that use Kamailio or giveaways, you are welcome to join the booth or just drop them there, we are more than happy to promote and spread them. Also, you are very welcome to just stop by and have a chat. Besides the activity in the exhibition area, I and Fred Posner will have presentations about using Kamailio together with Asterisk or other VoIP applications. Be sure you do not miss them, the agenda is available at: - http://www.asterisk.org/community/astricon-user-conference/schedule Expect a consistent group from Kamailio community, it's a great event to be at! Looking forward to meeting many of you next week in Orlando! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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Re: [SR-Users] [sr-dev] SIP ping, sockets and multi homed
by Patrick Wakano 25 Sep '17

25 Sep '17

Hello list, After deep investigation I found out that the issue is happening because of the mechanism Kamailio uses to find the most appropriate socket combined with the virtual IP usage.... For socket selection, Kamailio first asks Linux which would be the routing for a certain destination, then matches this outcome with the list of listened sockets. By default Linux will always return the local IP instead of the virtual one. So if Kamailio only binds to the virtual IP, the Linux's returned IP won't match our socket list and then the socket selection fails.... So currently Kamailio active/stand-by HA setup with virtual IP in a multi-homed environment does not work properly... We can overcome this by listening to all addresses, but then Kamailio will decide to use the local IP instead of the virtual one, and then the HA idea end up broken.... We can also force socket for everything. For regular call routing, this options actually works fine if the script always force some socket, but it seems to be not always possible for local generated messages. By disabling mhomed and using the dispatcher socket attr to send the OPTIONS message it works for example. But for the LCR module we have no option because apparently the LCR module has only one socket option, and so I am not able have a different socket per gateway. We can also explicitly set the src address for a routing rule to be the virtual IP instead of the local one. This makes Linux to return the virtual IP instead of the local one. This solution seems fine, but since it is done at linux level, every other service on the box would be affected and every route on the box would needed to be changed.. Another problem I found out is that I am not able to force the socket for local generated messages. It actually crashes Kamailio, and I always get a "core was not generated" msg.... anyone aware of that? The actual fix for this, would be having Kamailio to search the most appropriate socket in a different manner. It could loop the sockets and see which one reaches the destination instead of asking Linux to do it, or it could have some routing directives which could be consulted before selecting a socket.... (just ideas, not sure how feasible they are...) I am considering using the Linux routing change option, but has anyone faced such situation before? Regards, Patrick Wakano On 24 August 2017 at 09:18, Patrick Wakano <pwakano(a)gmail.com> wrote: > Hello Daniel, > I still couldn't make this SIP probing work.... So currently my setup > cannot actively check for GW availability... > Any idea about what may be causing it? > > Thanks for your time, > Patrick Wakano > > On 16 August 2017 at 13:01, Patrick Wakano <pwakano(a)gmail.com> wrote: > >> Forgot to mention it is a centos7 box... >> >> On 16 Aug. 2017 11:02 am, "Patrick Wakano" <pwakano(a)gmail.com> wrote: >> >>> Hi Daniel! >>> Very grateful for your attention! >>> >>> Apologies for lacking important info, check below the version and >>> sockets. >>> It may be worth mentioning these IPs are virtual ones and I am using net.ipv4.ip_nonlocal_bind >>> = 1(in this case the virtual IP is assigned to the box) >>> Interesting that it has a "mhomed: no" for all sockets but I do have >>> mhomed=1 in my cfg file.... could that be the issue? >>> Also there is no message before this one: /usr/sbin/kamailio[6164]: >>> ERROR: <core> [core/forward.c:181]: get_out_socket(): no socket found >>> This is the first one that pops when dispatcher tries to ping.... >>> >>> Thanks, >>> Patrick >>> >>> kamailio -v >>> version: kamailio 5.0.2 (x86_64/linux) >>> 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_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, >>> USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, >>> HAVE_RESOLV_RES >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB >>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>> id: unknown >>> compiled on 12:03:49 Jun 26 2017 with gcc 4.8.5 >>> >>> kamcmd> core.sockets_list >>> { >>> socket: { >>> proto: udp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: udp >>> address: 192.168.33.100 >>> ipaddress: 192.168.33.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 192.168.33.100 >>> ipaddress: 192.168.33.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 8080 >>> mcast: no >>> mhomed: no >>> } >>> } >>> >>> >>> On 15 August 2017 at 17:24, Daniel-Constantin Mierla <miconda(a)gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> what are the sockets kamailio is listening on? See `kamcmd help` for >>>> the command that lists the sockets. >>>> >>>> Also, what is the log messages before the first one you pasted, there >>>> should be another one printed from get_out_socket(). >>>> It is always important to provide kamailio version (kamailio -v) and >>>> the operating system you are using. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 14.08.17 14:41, Patrick Wakano wrote: >>>> >>>> Hello Kamailio list, >>>> >>>> Hope you all doing well! >>>> >>>> I am stuck with one problem that should be incredibly easy to be done, >>>> but apparently it is not.... I want my dispatcher, LCR and nathelper >>>> modules to always SIP ping my SIP clients. I have a Kamailio based SBC that >>>> is separating two networks, and so I am using the 'mhomed' flag, but none >>>> of these SIP pings are working.... for now I will only ask about the >>>> dispatcher.... >>>> My dispatcher module has to talk with SIP servers in both networks, the >>>> thing is when I enable SIP ping for it (modparam("dispatcher", >>>> "ds_probing_mode", 1)), it gives me this type of error: >>>> /usr/sbin/kamailio[6164]: ERROR: <core> [core/forward.c:181]: >>>> get_out_socket(): no socket found >>>> /usr/sbin/kamailio[6164]: ERROR: <core> [core/forward.c:183]: >>>> get_out_socket(): no corresponding socket found for(udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[6164]: ERROR: tm [ut.h:317]: uri2dst2(): no >>>> corresponding socket found for "192.168.33.110" af 2 (udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[6164]: ERROR: tm [uac.c:443]: t_uac_prepare(): no >>>> socket found >>>> /usr/sbin/kamailio[6164]: ERROR: dispatcher [dispatch.c:2652]: >>>> ds_ping_set(): unable to ping [sip:192.168.33.110:6060] >>>> >>>> After research it seems I need to manually set a "socket" attribute for >>>> each of my GWs in dispatcher list. After putting the socket parameter for >>>> the GW, I got another error: >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/socket_info.c:2046]: >>>> parse_protohostport(): bad port number in udp:192.168.33.100:5060 >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/forward.c:181]: >>>> get_out_socket(): no socket found >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/forward.c:183]: >>>> get_out_socket(): no corresponding socket found for(udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[14190]: ERROR: tm [ut.h:317]: uri2dst2(): no >>>> corresponding socket found for "192.168.33.110" af 2 (udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[14190]: ERROR: tm [uac.c:443]: t_uac_prepare(): no >>>> socket found >>>> /usr/sbin/kamailio[14190]: ERROR: dispatcher [dispatch.c:2652]: >>>> ds_ping_set(): unable to ping [sip:192.168.33.110:6060] >>>> >>>> What can be wrong in my port info in the socket config? (I tried the >>>> socket with and without quotes but had no luck...) >>>> select * from dispatcher where id=2; >>>> id | setid | destination | flags | priority >>>> | attrs | description >>>> ----+-------+-------------------------+-------+----------+-- >>>> -------------------------------------------+--------------------- >>>> 2 | 2 | sip:192.168.33.110:6060 | 0 | 0 | socket="udp: >>>> 192.168.33.100:5060";rweight=50 | >>>> (1 row) >>>> >>>> Anyhow, I was expecting the 'mhomed' parameter to do the socket >>>> selection for me. Regular call routing either by the dispatcher or by LCR, >>>> also seems to require me to explicitly set the socket otherwise Kamailio >>>> (or the OS) may end up using the wrong socket. So, another question is why >>>> 'mhomed' does not dictate the whole socket selection Kamailio has to do >>>> regardless the module it is using to send SIP requests? >>>> >>>> Sorry for the long e-mail.... >>>> Best regards, >>>> Patrick Wakano >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Development Mailing Listsr-dev@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Kamailio Advanced Training - www.asipto.com >>>> Kamailio World Conference - www.kamailioworld.com >>>> >>>> >>> >

2 2

LCR Ping GWs
by Patrick Wakano 25 Sep '17

25 Sep '17

Hello list, I was trying to activate continuous SIP ping from the LCR module, which apparently was available in the past, but currently it is not possible anymore. As I could check It can only ping inactive GWs. Question is, why such change was promoted? We have to send a call to the GW to detect it has died instead of actively monitoring them and so avoid this useless call attempt. It is not that big deal, but it does gives me a better monitoring of my providers (specially the ones not used so frequently) health. Regards, Patrick

1 0

Enabling Redirection in Kamailio SIP Server
by Anas Rasheed 23 Sep '17

23 Sep '17

How can we enable redirection in Kamilio SIP Server using Siremis User Interface for it? Can anyone provide us with the tutorial of it?

1 0

Memory configuration
by Joe Baran 23 Sep '17

23 Sep '17

I feel fairly confident I’m running into a memory tuning issue, but I’m struggling to determine the next step. Can someone give me some guidance on how to adjust settings to allow the system to stop crashing kamailio? Sep 14 18:25:15 server1 kamailio[29468]: ERROR: <core> [core/mem/q_malloc.c:292]: qm_find_free(): qm_find_free(0x7fa9eeeea000, 144); Free fragment not found! Sep 14 18:25:15 server1 kamailio[29468]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fa9eeeea000, 144) called from kazoo: kz_amqp.c: kz_amqp_async_query(1357), module: kazoo; Free fragment not found! Sep 14 18:25:15 server1 kamailio[29468]: ERROR: kazoo [kz_amqp.c:1359]: kz_amqp_async_query(): failed to allocate kz_amqp_cmd in process 29468 Sep 14 18:25:15 server1 kamailio[29468]: INFO: <script>: kGddp3V0bA6nitnk25dB4w..|log|failed to send Kazoo query for authentication xxxxxxxxxxxxxxxxxx Sep 14 18:25:15 server1 kamailio[29468]: ERROR: tm [t_reply.c:494]: _reply_light(): cannot allocate shmem buffer Sep 14 18:25:15 server1 kamailio[29468]: ERROR: sl [sl.c:271]: send_reply(): failed to reply stateful (tm) Sep 14 18:25:16 server1 kamailio[29483]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 108 Sep 14 18:25:16 server1 kamailio[29360]: ALERT: <core> [main.c:743]: handle_sigs(): child process 29438 exited by a signal 11 Sep 14 18:25:16 server1 kamailio[29360]: ALERT: <core> [main.c:746]: handle_sigs(): core was not generated Thanks, Joe Baran

3 6

Shared htable between servers
by Iskren Hadzhinedev 22 Sep '17

22 Sep '17

Hi list, Is it possible to have a htable shared between servers on different machines? I have two kamailio proxies on different servers for resilience and I'm storing call information in a hash table. Then I can route calls for the same user to the respective PBX. The reason for this is to have attended transfers working reliably (attended transfer between PBXes are a mess). Thank you! Kind regards, -- /Iskren Hadzhinedev/

2 1

Call transfer between pbx servers behind kamailio proxy
by Iskren Hadzhinedev 22 Sep '17

22 Sep '17

Hi everyone, I'm having a hard time transferring calls when the users are on different FreeSWITCH servers behind kamailio. The users are distributed as follows: user1(a)sip.test.com -> Kamailio - ds_select_dst(1,1) -> FreeSWITCH1 user2(a)sip.test.com-> Kamailio - ds_select_dst(1,1) -> FreeSWITCH1 user3(a)sip.test.com -> Kamailio - ds_select_dst(1,1) -> FreeSWITCH2 When user1 calls user2 and user2 does an attended transfer to user3, the following happens: 285 67.220453 USER2_IP -> KAMAILIO_IP SIP 726 Request: REFER sip:mod_sofia@FreeSWITCH1_IP:5060, in-dialog | 286 67.220674 KAMAILIO_IP -> FreeSWITCH1_IP SIP 877 Request: REFER sip:mod_sofia@FreeSWITCH1_IP:5060, in-dialog | 287 67.223651 FreeSWITCH1_IP -> KAMAILIO_IP SIP 925 Status: 202 Accepted | 288 67.223753 KAMAILIO_IP -> USER2_IP SIP 838 Status: 202 Accepted | 289 67.225502 FreeSWITCH1_IP -> KAMAILIO_IP SIP/SDP 1225 Request: INVITE sip:user3@sip.test.com | 290 67.387323 KAMAILIO_IP -> 69.172.200.109 SIP/SDP 1420 Request: INVITE sip:user3@sip.test.com | Turns out 69.172.200.109 is the actual IP address of sip.test.com (I've probably should have gone with sip.example.com or something else). What is the proper way to route that INVITE back to the other FreeSWITCH server so that the transfer would work? Thanks! Kind regards, -- /Iskren Hadzhinedev/

2 2

how to play ring tune when callee decline s
by 赵国杰 22 Sep '17

22 Sep '17

Hi guys, I want the caller to play a short audio(like "the number your are calling is busy") when the callee declines the call. How can i do that? Best Regards Jesse

4 21

Typos in Kamctl Stats results
by Noah Mehl 21 Sep '17

21 Sep '17

Hey, I’m noticing that the results from the websockets module when running # kamctl stats there are some odd results that may just be typos? "websocket:ws_msrp_max_concurrent_connections = 0", "websocket:ws_msrp_max_concurrent_connectons = 0”, "websocket:ws_sip_max_concurrent_connections = 0", "websocket:ws_sip_max_concurrent_connectons = 0", Thanks! ~Noah

3 2

Encryption of voice stream
by Vasan VS 21 Sep '17

21 Sep '17

Hello all, Kindly let us know if we can have a mechanism to have the voice stream in using the SIP server completely encrypted as well as prove that it is encrypted using any speific mechanism. Thanks in advance Vasan

3 2

COALESCE Transformation?
by Brandon Armstead 20 Sep '17

20 Sep '17

Hello all, I've checked the docs and did some brief searching, curious if this transformation or method exists currently? Such as: https://dev.mysql.com/doc/refman/5.7/en/comparison-operators.html#function_… Thanks! -- Sincerely, Brandon Armstead CTO / CRYY.com

1 0

private memory leak - but where?
by Sebastian Damm 20 Sep '17

20 Sep '17

Hi, I'm trying to debug a Kamailio which runs out of memory after a while. But following the hints at https://www.kamailio.org/wiki/tutorials/troubleshooting/memory doesn't help. I know it is the private memory that I'm missing. I already increased the pkg size to 16M, but somehow that seems to be not enough. The Kamailio receives HEP messages, sends them to a lua script where it is processed. Overall there are not more than 10 custom private variables used. In the log I'm seeing these messages: Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: app_lua [app_lua_api.c:726]: app_lua_run_ex(): error executing: handle_packet (err: 2) Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <script>: SCRIPT: failed to execute lua function! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/mem/q_malloc.c:292]: qm_find_free(): qm_find_free(0x7f9744eea010, 232); Free fragment not found! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f9744eea010, 232) called from core: core/parser/msg_parser.c: get_ hdr_field(116), module: core; Free fragment not found! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/parser/msg_parser.c:118]: get_hdr_field(): out of memory Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/parser/msg_parser.c:330]: parse_headers(): bad header field [Via: SIP/2.0/WSS o6m] Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: pv [pv_core.c:647]: pv_get_callid(): cannot parse Call-Id header kamcmd pkg.stats for this process gives me this output. { entry: 61 pid: 19134 rank: 61 used: 8748080 free: 4384 real_used: 16772832 total_size: 16777216 total_frags: 30 } Now I tried to get a dump of what is in use, following the guidelines on the wiki page mentioned above, but without success. I can easily start and stop the traffic coming in to Kamailio (by starting or stopping the captagent sending it there), but I can't get any output. Here's what I did: captagent stop kamcmd cfg.set_now_int core memlog 1 kamcmd cfg.set_now_int core mem_dump_pkg 19134 captagent start ; sleep 1 ; captagent stop (I started captagent so that the process actually had something to do because that's what the wiki page says it needs for printing memory information.) During that second, there were quite a lot of packets running through this process, but I did not get a memory dump. A shm_summary dump however works as expected. What am I missing? How do I get more information about what this process is using its memory for? TIA Sebastian

2 6

sipp audio file
by Diego Nadares 19 Sep '17

19 Sep '17

Hi guys, Any body know how to create pcap (with audio) file to use with sipp?  <nop> <action> <exec play_pcap_audio="../scenarios/pcap/demo.cap"/> </action> </nop> I tried with wireshark capturing rtp but I'm doing something wrong i think. Thanks! Diego

4 4

jsonrpcc problem
by Adam Kuśmirek 19 Sep '17

19 Sep '17

Hello, I have a problem with jsonrpcc module. OS is Debian Jessie. I tried with packeged kamailio 4.4 and compiled kamailio 5.0. Kamailio starts properly. Routing block is very simple: route { jsonrpc_request("get_source", "{'ipv4': '$si', 'port': $sp}", "RESPONSE", "ERROR", "$var(result)"); exit; } route[RESPONSE] { xlog("Result received: $var(result)"); } route[ERROR] { xlog("Error received: $var(result)"); } If any message arrives, kamailio sends it to my rpc_server (netstring encoded): 106:{'id': 2, 'jsonrpc': '2.0', 'method': 'get_source', 'params': {'ipv4': '10.20.121.231', 'port': 5060}}, My server decodes the message and immediately answers with netstring encoded response: 47:{"jsonrpc": "2.0", "result": "200 OK", "id": 2}, At this moment kamailio logs errors and disconnects from rpc server: root@kamailio-test:/etc/kamailio# 4(925) DEBUG: <core> [parser/msg_parser.c:595]: parse_msg(): SIP Request: 4(925) DEBUG: <core> [parser/msg_parser.c:597]: parse_msg(): method: <REGISTER> 4(925) DEBUG: <core> [parser/msg_parser.c:599]: parse_msg(): uri: <sip:kamailio.conpeek.com> 4(925) DEBUG: <core> [parser/msg_parser.c:601]: parse_msg(): version: <SIP/2.0> 4(925) DEBUG: <core> [parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1033201528>; state=16 4(925) DEBUG: <core> [parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 4(925) DEBUG: <core> [parser/msg_parser.c:486]: parse_headers(): Via found, flags=2 4(925) DEBUG: <core> [parser/msg_parser.c:488]: parse_headers(): this is the first via 4(925) DEBUG: <core> [receive.c:178]: receive_msg(): After parse_msg... 4(925) DEBUG: <core> [receive.c:221]: receive_msg(): preparing to run routing scripts... 4(925) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff 4(925) DEBUG: <core> [parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 4(925) DEBUG: <core> [parser/msg_parser.c:171]: get_hdr_field(): <To> [40]; uri=[sip:1000@kamailio.conpeek.com] 4(925) DEBUG: <core> [parser/msg_parser.c:172]: get_hdr_field(): to body ["1000" <sip:1000@kamailio.conpeek.com> ] 4(925) DEBUG: <core> [parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <REGISTER> 4(925) DEBUG: <core> [parser/msg_parser.c:183]: get_hdr_field(): content_length=0 4(925) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 4(925) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=43020, isACK=0 4(925) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 4(925) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 4(925) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): MD5 calculated: 920bfe28c581e1bf18554e3a9b234ceb 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 4(925) DEBUG: <core> [receive.c:318]: receive_msg(): cleaning up *0(921) ALERT: <core> [main.c:740]: handle_sigs(): child process 929 exited by a signal 11* * 0(921) ALERT: <core> [main.c:743]: handle_sigs(): core was generated* 0(921) INFO: <core> [main.c:755]: handle_sigs(): terminating due to SIGCHLD 0(921) DEBUG: <core> [main.c:757]: handle_sigs(): terminating due to SIGCHLD 7(928) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 6(927) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 5(926) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 4(925) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 3(924) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 2(923) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 1(922) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 0(921) DEBUG: tm [t_funcs.c:86]: tm_shutdown(): DEBUG: tm_shutdown : start 0(921) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): DEBUG: tm_shutdown : emptying hash table 0(921) DEBUG: tm [h_table.c:127]: free_cell_helper(): freeing transaction 0x7fb99a822350 from h_table.c:449 0(921) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): DEBUG: tm_shutdown : removing semaphores 0(921) DEBUG: tm [t_funcs.c:93]: tm_shutdown(): DEBUG: tm_shutdown : destroying tmcb lists 0(921) DEBUG: tm [t_funcs.c:96]: tm_shutdown(): DEBUG: tm_shutdown : done 0(921) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 0(921) DEBUG: <core> [mem/shm.c:174]: shm_core_lock_destroy(): destroying the shared memory lock 0(921) DEBUG: <core> [mem/shm.c:266]: shm_destroy_manager(): destroying memory manager: f_malloc 0(921) DEBUG: <core> [mem/pkg.c:89]: pkg_destroy_manager(): destroying memory manager: f_malloc The problem occures also if rpc server does not reply, i think when trying to start ERROR route. If I try with jsonrpc_notification, everything is OK. Any help ? Regards Adam

2 2

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