sr-users August 2014

sr-users@lists.kamailio.org

73 participants
130 discussions

by Yuriy Gorlichenko

Hello. I have Kamailio running behind NAT. It lesten eth0 with ip 192.168.0.3 and I have external IP that have domain name (for example sip.myserver.com). Register packets from clients comes from external IP. If I write at kamailio.cfg: alias=sip.myserver.com I see error at log - bad_uri sip.myserver.com I think it happens because kamailio does not know atything about external ip pecause kamamiliol working with server interfaces (eth0, eth1, etc.) So my question - how to listen external IP with domain name on kamailio that running behind NAT? P.S. I do not have any access to router, that present NAT for me.

9 years, 8 months

SIPp load testing

by bob sacamano

I am trying to follow the guide shown here:http://www.kamailio.org/docs/openser-performance-tests/ to load test my kamailio system.I am a little unclear as to how many instances of sipp are running. For the first part i see the command ./sipp -sf uac_msg.xml -rsa 192.168.2.102:5060 192.168.2.102:5070 -m 200000 -r 10000 -d 1 -l 70for generating the UAC part but is there another instance of sipp running on the kamailio computer acting as a UAS? I have seen come sites use for example: sipp 192.168.1.100:5060 -sn uas -p 5060 but is this not necessary? thanks for any help

9 years, 8 months

Sample Kamailio Config for Siremis

by Noah Mehl

I have managed to install Kamailio and Siremis 4.x, but I have noticed that neither Kamaliio or Siremis come with a sample config that makes sense for Siremis (as far as I can tell). For instance, if I want to use LCR with dispatchers, the sample Kamilio configs don’t have the dispatcher module loaded or configured. Am I missing something? Can someone please share theirs? Thanks! ~Noah

9 years, 8 months

ERROR: registrar [save.c:598]: test_max_contacts(): invalid cseq for aor <rene>

by Rene Montilva

Hi list i've been working with t_suspend, t_continue and async sleep, now the problem it's with register users, the softphones are frozen when try to establish a session with kamailio, now my users can't register and the only error i got is this: ERROR: registrar [save.c:598]: test_max_contacts(): invalid cseq for aor <rene> my kamailio config: # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 30) # max value for expires of registrations #modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "hash_size", 15) #!endif

9 years, 8 months

kamailio routes packets with invalid From/To headers with uac.restore_mode=auto when incoming packet does not use exact same replaced From/To header

by Alex Villacís Lasso

I am currently handling a system that runs kamailio and asterisk in the same machine. The kamailio instances are being used to emulate multiple SIP domains, by means of From/To mangling of incoming packets, which are then routed to Asterisk. The attached kamailio.cfg does this work. There is an problem when handling SUBSCRIBE requests (as required for BLF and voicemail indications). My configuration is written so that these SUBSCRIBE requests are not handled by kamailio, but instead routed to asterisk. There is a failure to check From/To headers to see whether NOTIFY packets generated as part of a subscription can be restored using the information in Record-Route. The end result is that kamailio ends up sending packets with garbled tags that are (rightly) rejected by the SIP endpoint. The following is an example that demonstrates the issue (using Jitsi as endpoint): After registration, Jitsi sends a SUBSCRIBE request: SUBSCRIBE sip:avillacisIM@pbx.villacis.com SIP/2.0 Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE From: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM@pbx.villacis.com> Max-Forwards: 70 Contact: "avillacisIM" <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com> User-Agent: Jitsi2.5.5255Linux Event: message-summary Accept: application/simple-message-summary Expires: 3600 Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Proxy-Authorization: Digest username="avillacisIM",realm="pbx.villacis.com",nonce="U9lZJlPZV/r06Xep/ukc1UzAIO0V3TbS",uri="sip:avillacisIM@pbx.villacis.com",response="0e18f4913c2693f6154c91f158fb17fe" Content-Length: 0 This packet is mangled by the configuration, and is sent to asterisk like this: SUBSCRIBE sip:avillacisIM@pbx.villacis.com SIP/2.0 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE From: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080> Max-Forwards: 69 Contact: "avillacisIM" <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com> User-Agent: Jitsi2.5.5255Linux Event: message-summary Accept: application/simple-message-summary Expires: 3600 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKd941.2ab9cf36e41dc48855ae2cbe9a309d0a.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Content-Length: 0 The asterisk response for the SUBSCRIBE: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKd941.2ab9cf36e41dc48855ae2cbe9a309d0a.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=as5562e95e Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:avillacisIM@127.0.0.1:5080>;expires=3600 Content-Length: 0 This is in turn transformed back by kamailio, and sent to Jitsi like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=bf427f4a To: "avillacisIM" <sip:avillacisIM@pbx.villacis.com>;tag=as5562e95e Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:avillacisIM@127.0.0.1:5080;alias=127.0.0.1~5080~1>;expires=3600 Content-Length: 0 Now asterisk wants to send a NOTIFY to the endpoint for the subscription. The NOTIFY looks like this: NOTIFY sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK658fa5fc;rport Max-Forwards: 70 Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes>,<sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as5562e95e To: <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.11.0 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) Here is where the bug appears. The autoprocessing does not recognize that the From header (From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as5562e95e) from the above request has nothing to do with the saved information (vsf parameter). Instead, it blindly mangles the From header, and does not even run a sanity check on the result before routing it. The end result is shown below. NOTIFY sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0 Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as5562e95e> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as5562e95e> Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bK8333.8bfe7bc2bd554a8631f0d00d463b28ee.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK658fa5fc;rport=5080 Max-Forwards: 69 From: "asterisk" <sip:asterisk@12(.0.0.1:5080.....@127.0.0.1:5080>;tag=as5562e95e To: <sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.11.0 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) From examination of the source code, the vsf and vst strings are base64 encodings of the result of XORing the byte strings of the old and new tags. For this to work, the headers of future packets should match. However, here kamailio does not realize that the header does not match (by the ftag), and also does not check that the resulting "restored" header is a valid header.

9 years, 8 months

SCA module - sca_call_info_update() function for 180 responses

by João Vitor Arruda

Hello all, I've had proposed a fix to SCA module in the tracker ( https://sip-router.org/tracker/index.php?do=details&task_id=446) but did´t hear anything about it until now. I would appreciate some comments on that to see if its feasible to be committed or even hear for anyone else using SCA module with Polycom phones to see if they have the same issue described. Thanks, João Vitor Arruda

9 years, 8 months

Kamailio V4.1.3 - Dialog timeout -ACC ERROR

by Yasin Caner

Hello; i am trying to limit call duration with dialog module. So i use *timeout_avp *for limiting it. there is no problem to close calls but when it happens i got some errors and kamailio can not write Bye result in DB. There is code sample and result in below. 2. question : is there a big difference between dialog_ng and dialog? Which one is more stable? 3. question : After restarting kamailio , Dialog hash vanish so timeout isn't working. Can DB mode solve this problem? PS: event_route[dialog:end] isn't working in Kamailio V 4.1.4. I got some error like t_w_relay. Thanks for helps. ------------------------------------------------------------------------ #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #endif ------------------------------------------------------------------------ route[SW]{ xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); $dlg_ctx(timeout_bye)=1; $avp(i:10)=15; $dlg_ctx(timeout_route)=33; dlg_manage(); set_dlg_profile("userid","$avp(userid)"); if(get_profile_size("userid","$avp(s_userid)")){ xlog("L_INFO","Userid_Size:$avp(s_userid)"); } xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); if(!t_relay()) { send_reply("408", "Servis Disi"); }else{ exit; } return; } event_route[dialog:start]{ xlog("L_ALERT","START:ci:$ci:method:$rm:start"); } event_route[dialog:end]{ xlog("L_ALERT","END"); xlog("L_ALERT","END:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_ALERT","END:$rm:$rr:$rs:"); } event_route[dialog:failed]{ xlog("L_INFO","FAILED:ci:$ci:method:$rm:end"); } route[33]{ xlog("L_INFO","route:33:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_INFO","route:33:$rm:$rr:$rs:"); } # 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()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } ------------------------------------------------------------------------ Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: ALERT: <script>: USERID:456123 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: ALERT: <script>: arayan:XXXX:aranan:XXXXX Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri XXXX was rewritten to sip:YYYYYYYYYYYYYYY@XXXXXXX5:5060, carrier 3, domain 1 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: vardesc: 40 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:<null>:h_entry:<null>:timeout:0:bye:0:<null> Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Userid_Size:1 Aug 1 16:51:52 host-91-93-189-136 /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:3953:h_entry:3170:timeout:14:bye:1:15 Aug 1 16:51:55 host-91-93-189-136 /usr/local/sbin/kamailio[6114]: ALERT: <script>: START:ci:2c5695c1644fa2b135f57ea72c590cc7@XXXXXXX:5060:method:INVITE:start Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:OPTIONS:<null>:<null>: Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:OPTIONS:<null>:<null>: Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6121]: WARNING: dialog [dlg_req_within.c:212]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7feb2115d918 [3170:3953] with clid '2c5695c1644fa2b135f57ea72c590cc7@95.0.154.92:5060' and tags 'as3c7c5aa4' 'as0b640624' Aug 1 16:52:11 host-91-93-189-136 /usr/local/sbin/kamailio[6121]: ERROR: acc [acc_cdr.c:574]: cdr_on_end(): invalid values#012!

9 years, 8 months

sip

by Nar

Hi, I am developing a social application where I want to enable voice & video calling function and still I don't know what to use in server side, do you have any product that I can use to develop the server side(for 10.000+ users)? Thank you in advance.

9 years, 8 months

[Kamailio]list members

by Djamel Bahamid

Hi, I would like to subscribe to the list. Best Regards, Djamel BAHAMID.

9 years, 8 months

SIGUSR1 for memory status not working as documented - only one process reports back

by Alex Villacís Lasso

I am trying to track down a memory leak that was triggered by a patch I wrote for my local copy of kamailio 4.1.4 . For this, I am following the documentation at http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory . This page claims that once memlog is set in the configuration file, a kamailio process will dump a report of the allocation map when shutting down, or when receiving a SIGUSR1. I have configured my kamailio.cfg with memlog=1 and no other change, and I see the memory report on shutdown for all processes. However, when I send a SIGUSR1 to a kamailio process, the process does absolutely nothing, with the exception of the first kamailio process (the one reported as Type=attendant by "kamctl ps"). All of the other processes just ignore SIGUSR1. What is going on? It is inconvenient to force a shutdown of all the kamailio processes just to get the memory report.

9 years, 8 months

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sr-users August 2014