- sr-users - kamailio.org

Info: Kamailio presentations later today at ClueCon TGI
by Daniel-Constantin Mierla 16 Feb '21

16 Feb '21

Hello, short note to inform that later today will be 3 sessions related to Kamailio at ClueCon TGI (online event): * a workshop about basics of SIP routing with Kamailio * a presentation about DID routing with Kamailio * a presentation about outsourcing services from FreeSwitch to Kamailio The first two are done by me, the last one by Fred Posner. The event is free to attend, by participating to a video conference or watch the youtube stream -- more details at: * https://www.cluecon.com/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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Re: [SR-Users] Advice on handling SIP trunk to a Perimeta SBC
by Björn Bylander 16 Feb '21

16 Feb '21

Hi again, I'm using Kamailio 5.3.2 by the way. Björn Bylander writes: > Hi, > > So I’m tasked with setting up a SIP trunk to a Perimeta SBC (https://www.metaswitch.com/products/perimeta-sbc). > > Apparently, > > Perimeta hides the topology of networks by rewriting or removing topology-sensitive information from SIP messages. By default, your Session Controller will make the following changes to SIP messages. > > > > The Session Controller generates new dialog identifiers (call-IDs, From tags and To tags) for each side of the call, so that it does not expose information about your core network to your access networks. > > The Session Controller strips Record-Route and Route information from the message. > > The Session Controller rewrites Contact and Via headers so that the source IP address is replaced with the local address of the outbound adjacency. > > The Session Controller replaces the IP addresses in c= lines in SDP with the addresses it has allocated for media forwarding. For more information on this, see Media addresses and gates. > > My problem is this: When the Perimeta SBC sends its first INVITE (I can't say anything about any succeeding INVITEs yet) it's added a Record-Route header with the "lr" parameter included which, as far as I can see makes Kamailio think it should use loose routing which is all well I think. But when the SBC sends an ACK for the "200 OK" from the Kamailio side it doesn't include any Route headers and I think that makes it hard for Kamailio (at least with the standard script which is basically what I'm using) to know where to relay the ACK, at least it makes the t_check_trans() call in WITHINDLG after 'is_method("ACK")' return false which makes the script ignore the ACK. > > Any thoughts or suggestions on how to handle this? > > Thanks in advance, > Björn Bylander

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Kamailio v5.4.4 Released
by Daniel-Constantin Mierla 15 Feb '21

15 Feb '21

Hello, Kamailio SIP Server v5.4.4 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.4. For more details about version 5.4.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/02/kamailio-v5-4-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Releasing Kamailio v5.4.4
by Daniel-Constantin Mierla 15 Feb '21

15 Feb '21

Hello, I am considering to release Kamailio v5.4.4 sometime next week, likely on Monday, February 15, 2020. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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IMS-PSTN
by eyas barhouk 13 Feb '21

13 Feb '21

Hello dears I'm trying to build an IMS testbed based on kamailio and integrate it with the PSTN network, in order to do that I'm looking for AGCF,MGCF open source solution. Am I able to do the functionality of those nodes based on Kamailio,Asterisk or any other open source? thanks and regards Eyas

1 0

IMS-PSTN
by eyas37＠hotmail.com 12 Feb '21

12 Feb '21

1 0

basic help, In over my head in getting a webrtc up.
by Daniel Siemens 12 Feb '21

12 Feb '21

I am trying to get a webrtc setup going. Here is what I have I have asterisk server at 10.123.244.18. The webrtc works internally from the freepbx UCP application As well as the Raspberry phone allocation. This server doesn't have any nat on it all devices are on local / reouted networks. I have a Proxy server at 10.123.245.30 address. This server is located in AWS and has an elastic IP. On this server I have ngiinx that will load the raspberry phone up. What configuration do I need in kamailio and rtpengine to get this working. If I forward port 8089 in nginx to the /ws side on my asterisk server I can get a call to bridge but with no audio and the call end at 30 seconds when remote. It works internally fine. Likely beccuae the web browser can get to https://10.123.244.18:8089/ws ports fine. Thanks. ------------------------------------------------------------------ This email is intended solely for the use of the addressee and may contain information that is confidential, proprietary, or both. If you receive this email in error please immediately notify the sender and delete the email. ------------------------------------------------------------------

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loose_route() does not find socket by name
by Juha Heinanen 12 Feb '21

12 Feb '21

I did more tests and found that loose_route() does not find socket by name given in ;sn param: Feb 12 10:40:59 lab /usr/bin/sip-proxy[16975]: WARNING: rr [loose.c:799]: rr_do_force_send_socket(): no socket found to match second RR (sip:tenantX.teams.tutpro.com:8007;transport=tls;r2=on;sn=ext_tls;lr;n1;savp=avp;pm=0) In config I have defined: listen=tls:192.x.x.x:8007 name "ext_tls" -- Juha

2 8

Kamailio Memory Leak
by Uttam Giri 11 Feb '21

11 Feb '21

Hi, We have observed increased memory utilization in Kamailio, It increases every day, even though traffic is not increased. We are using Kamailio+RTPEngine in our setup. Phone---->Kamailio+RTPEngine------>Asterisk /etc/kamailio# kamcmd mod.stats all shm Module: core { sip_msg_shm_clone(496): 542688 msg_lump_cloner(986): 178776 build_req_buf_from_sip_req(2218): 111968 xavp_new_value(116): 22240 counters_prefork_init(212): 39680 cfg_clone_str(130): 40 cfg_shmize(217): 712 main_loop(1313): 8 init_pt(107): 8 init_pt(106): 8 init_pt(105): 4592 register_timer(995): 192 init_tcp(4962): 8192 init_tcp(4956): 32768 init_tcp(4947): 8 init_tcp(4940): 8 init_tcp(4933): 8 init_tcp(4927): 8 init_tcp(4915): 8 init_avps(90): 8 init_avps(89): 8 timer_alloc(498): 48 init_dst_blacklist(438): 16384 init_dst_blacklist(430): 8 init_timer(267): 8 init_timer(266): 16384 init_timer(265): 8 init_timer(264): 8 init_timer(253): 8 init_timer(221): 8 init_timer(210): 278544 init_timer(209): 8 init_timer(197): 8 cfg_child_cb_new(829): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 qm_shm_lock_init(1202): 8 Total: 1253464 } Module: sl { init_sl_stats_child(125): 5952 sl_startup(90): 8 init_sl_stats(110): 8 Total: 5968 } Module: siptrace { mod_init(319): 8 Total: 8 } Module: permissions { Total: 0 } Module: xlog { Total: 0 } Module: rtpengine { rtpp_function_call(2627): 1076024 shm_str_dup(779): 553016 rtpengine_hash_table_init(111): 14336 rtpengine_hash_table_init(96): 2048 rtpengine_hash_table_init(85): 1024 rtpengine_hash_table_init(76): 2048 rtpengine_hash_table_init(67): 2048 rtpengine_hash_table_init(58): 32 add_rtpengine_socks(975): 24 add_rtpengine_socks(956): 64 get_rtpp_set(870): 8 get_rtpp_set(861): 56 mod_init(1596): 8 mod_init(1589): 24 mod_init(1576): 8 mod_init(1569): 8 mod_init(1555): 16 Total: 1650792 } Module: pv { add_shvar(177): 8 add_shvar(170): 64 lock_set_alloc(66): 80 Total: 152 } Module: phonenum { Total: 0 } Module: textops { Total: 0 } Module: rtimer { Total: 0 } Module: tm { _reply_light(496): 1640 build_cell(334): 973848 relay_reply(2058): 69376 init_tm_stats_child(56): 7936 init_tmcb_lists(72): 16 init_tmcb_lists(70): 16 init_hash_table(482): 2097152 Total: 3149984 } Module: kex { pkg_proc_stats_init(79): 1488 Total: 1488 } Module: tmx { tmx_init_pretran_table(90): 256 Total: 256 } Module: utils { conf_init(552): 24 init_shmlock(134): 8 Total: 32 } Module: xhttp_prom { prom_metric_init(134): 8 Total: 8 } Module: dispatcher { ds_ht_init(89): 12288 reindex_dests(723): 768 mod_init(484): 80 ds_avl_insert(3621): 1712 pack_dest(411): 72 ds_ht_init(79): 32 ds_init_data(262): 16 ds_init_data(254): 16 mod_init(339): 8 mod_init(337): 8 ds_ping_active_init(138): 8 Total: 15008 } /etc/kamailio# kamcmd mod.stats all pkg Module: core { init_io_wait(469): 6192 init_modules(885): 16 rpc_hash_add(151): 1024 cnt_hash_add(394): 2048 add_callback(59): 168 register_select_table(458): 48 sr_wtimer_init(351): 136 cfg_new_group(79): 624 fix_sock_str(517): 1024 fix_sock_str(500): 240 add_alias(103): 96 add_alias(99): 96 init_dst_set(89): 32296 grp_hash_add(241): 512 fix_hostname(1457): 56 fix_socket_list(1666): 24 route_new_list(199): 184 fix_param(1114): 88 fix_param(1053): 312 fix_expr(554): 192 init_io_wait(516): 3224 subst_str(514): 720 mk_rval_expr2(2681): 17360 mk_elem(90): 232 mk_new_net(52): 48 tr_new(1730): 240 mk_rval_expr_v(2553): 216 yyparse(2852): 736 mk_rval_expr1(2621): 19096 mk_rval_expr_v(2540): 109152 pv_cache_add(349): 5616 pv_parse_format(1173): 4984 fix_param(1019): 2568 mk_action(118): 44416 cfg_declare(50): 3136 set_mod_param_regex(145): 88 set_mod_param_regex(134): 104 tr_table_add(1959): 360 register_module(233): 28784 register_module(216): 2808 load_module(417): 1552 sr_push_yy_state(1627): 1008 new_sock_info(348): 16 new_sock_info(339): 32 new_sock_info(327): 32 new_sock_info(304): 48 new_sock_info(299): 1200 add_alias(103): 56 add_alias(99): 64 yyparse(677): 40 yyparse(2387): 128 yyparse(658): 144 yyparse(1878): 168 get_hdr_field(116): 232 parse_headers(325): 64 subst_parser(301): 120 subst_parser(291): 1440 subst_parser(274): 640 pp_subst_add(58): 240 sr_push_yy_state(1697): 24 addstr(1445): 69072 rpc_hash_add(105): 16280 str_hash_alloc(59): 512 pv_table_add(236): 17248 pv_init_buffer(2139): 327680 pv_init_buffer(2129): 320 init_nonsip_hooks(43): 24 init_rlist(146): 40 rval_get_str(1259): 3696 route_add(124): 968 str_hash_alloc(59): 768 sr_push_yy_state(1687): 464 pp_define(1819): 568 init_counters(125): 128 cnt_hash_add(341): 27096 str_hash_alloc(59): 1280 Total: 762656 } Module: sl { sl_register_callback(514): 64 Total: 64 } Module: siptrace { mod_init(340): 384 Total: 384 } Module: permissions { get_pathname(243): 64 Total: 64 } Module: xlog { mod_init(212): 4104 xdbg_fixup_helper(498): 616 xlog_fixup_helper(533): 3720 Total: 8440 } Module: rtpengine { child_init(1964): 240 build_rtpp_socks(1791): 8 Total: 248 } Module: pv { pv_parse_xavp_name(541): 112 pv_parse_xavp_name(515): 112 tr_parse_string(2639): 120 tr_parse_string(2628): 120 tr_parse_string(2519): 104 tr_parse_string(2509): 64 tr_parse_string(2484): 64 tr_parse_string(2473): 64 add_var(65): 176 add_var(58): 344 Total: 1280 } Module: phonenum { pv_parse_phonenum_name(131): 32 sr_phonenum_add_item(99): 8 sr_phonenum_add_item(93): 304 Total: 344 } Module: textops { fixup_method(3598): 128 hname_fixup(3528): 504 Total: 632 } Module: rtimer { stm_e_param(384): 112 stm_t_param(299): 48 Total: 160 } Module: tm { Total: 0 } Module: kex { Total: 0 } Module: tmx { Total: 0 } Module: utils { Total: 0 } Module: xhttp_prom { Total: 0 } Module: dispatcher { Total: 0 } /etc/kamailio# kamailio -v version: kamailio 5.4.3 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, 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 with gcc 6.3.0 Any idea which could cause a memory leak. Does any of the module cause an issue? Thanks in Advance. Uttam

3 2

loose_route() fails on domain name
by Juha Heinanen 11 Feb '21

11 Feb '21

Using a domain name in R-R header may not be such a good idea: Feb 11 12:12:06 lab /usr/bin/sip-proxy[735]: WARNING: rr [loose.c:799] rr_do_force_send_socket(): no socket found to match second RR (sip:tenantX.teams.tutpro.com:8007;transport=tls;r2=on;pm=0;savp=avp;lr) Or is there some means to tell K, which FQDN maps to which IP address, when there are many such FQDNs? -- Juha

4 9

ds_mark_dst( ) in DISPATCHER
by alirsm 11 Feb '21

11 Feb '21

Hi, In my failover setting using Dispatcher module, I mark the GW if fails to response. OPTIONS are being sent every 30 sec: failure_route[DISPATCH_FAILOVER]{ if (t_is_canceled()) exit; if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { ds_mark_dst("ip"); if (ds_next_dst()) { t_set_fr(0,1000); t_on_failure("DISPATCH_FAILOVER"); route(RELAY); exit; } } } However, Kamailio still recognize it as active gw ( A.A.A.A is down, no response from OPTIONS being sent) $ sudo sercmd dispatcher.list { SET_NO: 1 SET: { SET_ID: 1 DEST: { URI: sip:A.A.A.A:5060 FLAGS: AP PRIORITY: 0 ATTRS: } DEST: { URI: sip:B.B.B.B:5060 FLAGS: AP PRIORITY: 0 ATTRS: } } } Any help would be appreciated. AL

1 0

R-R to MS Teams Proxy
by Juha Heinanen 11 Feb '21

11 Feb '21

When request is sent from Kamailio to MS Teams SIP Proxy, the top R-R URI needs to contain FQDN of Kamailio SIP proxy instead of its IP address. Document https://skalatan.de/de/blog/kamailio-sbc-teams suggest to replace record_route(); call with record_route_preset("SBC-DNS-DOMAIN:5061;transport=tls", "SBC-IP-ADDR:5060"); That works only in a very simple case where the request came in over UDP or TCP and SIP Proxy has only one listening address, i.e., SBC-IP-ADDR. One way to solve the problem might be a new r_r function that would take FQDN of the top R-R URI as argument or introduction of a pv from where the current record_route() function would take the FQDN if it has been set. Any comments or other solutions? -- Juha

6 19

No ACK from MS Teams
by Phillman25 Kyriacou 11 Feb '21

11 Feb '21

Dear List Hope this email finds you all well. I have followed the below tutorial on how to integrate Kamailio with MS Teams https://skalatan.de/en/blog/kamailio-sbc-teams However i have been facing an issue with MS Teams with direct routing where MS Teams does not send back an ACK after a 200 OK. The current inbound call flow is the following: *MS Teams ==> Kamailio==>Asterisk* Once Asterisk answers with 200 OK we send this 200 OK back to MS Teams however, MS Teams just doesn't answer back with ACK and call drops. The below trace shows the INVITE coming from MS Teams and the 200 OK we send back. |||||||||||||||||||| ==================== tag: rcv pid: 26135 process: 23 time: 1613025911.983278 date: Thu Feb 11 08:45:11 2021 proto: tls ipv4 srcip: 52.114.132.46 srcport: 4352 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ INVITE sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone;transport=tls SIP/2.0 FROM: Phillip Kyriacou<sip:+357XXXXXXXX@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c3da9477e05d45fca31c24a155af3318 TO: <sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: 9cb9a2f3144e594c87bccda76791c28e MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK0cf2c45 RECORD-ROUTE: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> CONTACT: <sip:api-du-b-euwe.pstnhub.microsoft.com:443 ;x-i=6e5254d9-79c2-4657-b54b-68791aa8e81f;x-c=9cb9a2f3144e594c87bccda76791c28e/d/10/e21a 6ff7125e4cbd91e75ec687fd4c5d> CONTENT-LENGTH: 1097 MIN-SE: 300 SUPPORTED: timer USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.2.9.1 i.USEA.7 CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY SESSION-EXPIRES: 3600 v=0 o=- 170239 0 IN IP4 127.0.0.1 s=session c=IN IP4 52.113.40.94 b=CT:10000000 t=0 0 m=audio 51414 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118 c=IN IP4 52.113.40.94 a=rtcp:51415 a=ice-ufrag:BwFs a=ice-pwd:sFyvnfQ9iH101E80ZKjCxi9+ a=rtcp-mux a=candidate:1 1 UDP 2130706431 52.113.40.94 51414 typ srflx raddr 10.0.137.19 rport 51414 a=candidate:1 2 UDP 2130705918 52.113.40.94 51415 typ srflx raddr 10.0.137.19 rport 51415 a=candidate:2 1 tcp-act 2121006078 52.113.40.94 49152 typ srflx raddr 10.0.137.19 rport 49152 a=candidate:2 2 tcp-act 2121006078 52.113.40.94 49152 typ srflx raddr 10.0.137.19 rport 49152 a=label:main-audio a=mid:1 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Igc2LVM9FH5kQtkRpHw99H5SB7Rd7eKzJy4gdWJg|2^31 a=sendrecv a=rtpmap:104 SILK/16000 a=rtpmap:9 G722/8000 a=rtpmap:103 SILK/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=ptime:20 |||||||||||||||||||| ==================== tag: snd pid: 26086 process: 6 time: 1613025917.726664 date: Thu Feb 11 08:45:17 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.132.46 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 200 OK Via: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK0cf2c45 Record-Route: <sip:SBC-IP-ADDR:5060;ftag=c3da9477e05d45fca31c24a155af3318;lr=on> Record-Route: <sip:SBC-FQDN:5061;transport=tls;ftag=c3da9477e05d45fca31c24a155af3318;lr=on> Record-Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> From: Phillip Kyriacou<sip:+357XXXXXXX@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c3da9477e05d45fca31c24a155af3318 To: <sip:+357XXXXXXXX@SBC-FQDN:5061;user=phone>;tag=as659924a4 Call-ID: 9cb9a2f3144e594c87bccda76791c28e CSeq: 1 INVITE Server: MediaGW V1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:357XXXXXXXX@ASTERISK_PUBLIC_IP:5060> Content-Type: application/sdp Require: timer Content-Length: 339 v=0 o=root 411266324 411266324 IN IP4 ASTERISK_PUBLIC_IP s=ASTERISK c=IN IP4 ASTERISK_PUBLIC_IP t=0 0 m=audio 14416 RTP/SAVP 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9PcaAGjbemJKTERlFTcVwmLRQoDSEQPxX0L1a6RF a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=sendrecv |||||||||||||||||||| Am I doing anything wrong here that MS Teams wont respond to my 200 OK? The Kamailio SBC is showing active on MS Teams and Kamailio dispatcher shows all the MS Team hubs as AP. I also can't see any errors in the log file My TLS cfg has the following: ====================== *[server:default]method = TLSv1.2+verify_certificate = yesrequire_certificate = yesprivate_key = /etc/letsencrypt/live/sbc.intelligentsupport.eu/privkey.pem <http://sbc.intelligentsupport.eu/privkey.pem>certificate = /etc/letsencrypt/live/sbc.intelligentsupport.eu/fullchain.pem <http://sbc.intelligentsupport.eu/fullchain.pem>ca_list = /etc/ssl/certs/ca-certificates.crt* *[client:default]method = TLSv1.2+verify_certificate = yesrequire_certificate = yesprivate_key = /etc/letsencrypt/live/sbc.intelligentsupport.eu/privkey.pem <http://sbc.intelligentsupport.eu/privkey.pem>certificate = /etc/letsencrypt/live/sbc.intelligentsupport.eu/fullchain.pem <http://sbc.intelligentsupport.eu/fullchain.pem>ca_list = /etc/ssl/certs/ca-certificates.crt* Thanks very much in advance! Phillip

5 6

Time elapsed since previous message.
by David Villasmil 10 Feb '21

10 Feb '21

Hello all, Is it possible to know the elapsed time since the previously received message? On outgoing calls, I.e: when i get a 180, how long did the 100 arrived? Or the INVITE... Thanks David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 11

Kamailio Split Variable
by Mihai 10 Feb '21

10 Feb '21

Hi! Can I split the variable $fd ? the $fd has the value: asterisk-wind.voicevw.lan; and i want to take the the first part - asterisk-wind any hints ?

4 3

Health Check on Dispatcher Failover
by alirsm 10 Feb '21

10 Feb '21

Hi I have setup failover on Kamailio 5.1 using Dispatcher module as below. On a round robin algorithm, INVITEs are being sent to BOTH healthy and faulty GW. When an INVITE is being sent to the faulty GW, a re-transmission will be sent after "t_set_fr" time to the healthy GW and calls are completed successfully. I was wondering if I can setup the health check correctly so the faulty IP GW can be out of round robin till the health check pass it again, i.e. no INVITE is being sent to the faulty GW as long as the health check not pass it. Your help is greatly appreciated, AL # dispatcher.list 1 sip:X.X.X.X:5060 0 0 1 sip:Y.Y.Y.Y:5060 0 0 # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_threshold", 1) # Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "4")) { send_reply("503", "No destination"); exit; } t_on_failure("DISPATCH_FAILOVER"); t_set_fr(0,1000); route(RELAY); exit; } # Failure Route failure_route[DISPATCH_FAILOVER]{ if (t_is_canceled()) exit; if(!ds_next_dst()) { send_reply("503", "Out of gateways"); exit; } t_on_failure("DISPATCH_FAILOVER"); t_relay(); }

1 0

http_async_client expected result in $http_rr
by Anthony Joseph Messina 10 Feb '21

10 Feb '21

https://www.kamailio.org/docs/modules/stable/modules/http_async_client says $http_rr will be the HTTP reason phrase. I'm understanding this to mean: Code: Reason: 200 OK 404 Not Found In my current 5.4 build, $http_rr returns empty (not <null> or anything else) regardless of what the server sends in the headers. Am I misunderstanding this? I'm writing an API and was hoping to do send_reply("$http_rs", "$http_rr"); -- Anthony - https://messinet.com F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 2

authenticated INVITE badly managed when topos is activated
by frédéric Gaisnon 10 Feb '21

10 Feb '21

Hi, Here is some more information about my problem. I think that topos impacts challenge computing. Do you have the same behaviour I observed? Do you need more information? My tests were done with kamailio 5.4.3 on Centos7 Without topos activated (note that with topoh activated I have the same good behaviour): CPE - INVITE -> SBC CPE <- 407 ---- SBC CPE - INVITE ->SBC (with proxy-authorization header) -- INVITE --> PROXY (So in this case challenge is validated and INVITE is forwarded) With topos activated: CPE - INVITE -> SBC CPE <- 407 ---- SBC CPE - INVITE ->SBC (with proxy-authorization header) CPE <-407 -----SBC topos configuration: loadmodule "ndb_redis.so" loadmodule "topos.so" loadmodule "topos_redis.so" # ----- topos params ----- modparam("topos", "storage", "redis") modparam("topos", "dialog_expire", 15000) Code used: # IP authorization and user authentication route[AUTH] { xlog("L_DBG", "route[AUTH]\n"); #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif #!ifdef WITH_AUTH if ((is_method("REGISTER")) || ($avp(need_auth) == "1")) { ####need_auth is equal to 1 in this case # authenticate requests $var(key)=$fU + "@" + $fd; if($sht(auth_cache=>$var(key))!=$null) { if (!pv_auth_check("$fd", "$sht(auth_cache=>$var(key))", "0", "1")) { auth_challenge("$fd", “1”); #################### we always go here with INVITE with proxy-authorization header and the return code is always -5 (AUTH_NO_CREDENTIALS) exit; } } else { if (!auth_check("$fd", "subscriber", "1")) { if ($rc == -1) { append_to_reply("Retry-After: 10\r\n"); send_reply("503", "Authentication server error"); exit; } auth_challenge("$fd", "0"); exit; } $sht(auth_cache=>$var(key)) = $avp(password); } # user authenticated - remove auth header consume_credentials(); ######## without topos we go here with INVITE with proxy-authorization header } #!endif return; } Note that in this case (with topos) the return code of function pv_auth_check is always -5 (AUTH_NO_CREDENTIALS) CASE OK: Frame 3279: 545 bytes on wire (4360 bits), 545 bytes captured (4360 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.102, Dst: 192.168.1.11 Transmission Control Protocol, Src Port: 5060, Dst Port: 60796, Seq: 1, Ack: 953, Len: 477 Session Initiation Protocol (407) Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bK2df8e195D1847B94;rport=60796;received=192.168.1.11 From: "6200" <sip:6200@entreprise-108.fr>;tag=B583B663-FBFBFCAA To: <sip:0900000000@entreprise-108.fr ;user=phone>;tag=83518db21d5b2e9b777975024049f5a3.8f270000 CSeq: 1 INVITE Call-ID: 9378ee27e6b7aea384a881c938de8138 [Generated Call-ID: 9378ee27e6b7aea384a881c938de8138] Proxy-Authenticate: Digest realm="entreprise-108.fr", nonce="YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO" Authentication Scheme: Digest Realm: "entreprise-108.fr" Nonce Value: "YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO" Content-Length: 0 Frame 3285: 1259 bytes on wire (10072 bits), 1259 bytes captured (10072 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.102 Transmission Control Protocol, Src Port: 60796, Dst Port: 5060, Seq: 1578, Ack: 478, Len: 1191 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:0900000000@entreprise-108.fr;user=phone;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bK827c83577BAADACE From: "6200" <sip:6200@entreprise-108.fr>;tag=B583B663-FBFBFCAA SIP Display info: "6200" SIP from address: sip:6200@entreprise-108.fr SIP from tag: B583B663-FBFBFCAA To: <sip:0900000000@entreprise-108.fr;user=phone> SIP to address: sip:0900000000@entreprise-108.fr;user=phone CSeq: 2 INVITE Call-ID: 9378ee27e6b7aea384a881c938de8138 [Generated Call-ID: 9378ee27e6b7aea384a881c938de8138] Contact: <sip:6200@192.168.1.33;transport=tcp> Contact URI: sip:6200@192.168.1.33;transport=tcp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomVVX-VVX_500-UA/5.7.0.14430 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Supported: replaces,100rel Allow-Events: conference,talk,hold Proxy-Authorization: Digest username="6200", realm=" entreprise-108.fr", nonce="YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO", uri=" sip:0900000000@entreprise-108.fr;user=phone;transport=tcp", response="3e0013cc3dc3855602ce1939af7e6f40", algorithm=MD5 Authentication Scheme: Digest Username: "6200" Realm: "entreprise-108.fr" Nonce Value: "YCPgXmAj3zLDB3+utLVpmc+Y917i5qZO" Authentication URI: "sip:0900000000@entreprise-108.fr ;user=phone;transport=tcp" Digest Authentication Response: "3e0013cc3dc3855602ce1939af7e6f40" Algorithm: MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 270 Message Body Bad case (with topos activated): Frame 9071: 545 bytes on wire (4360 bits), 545 bytes captured (4360 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.102, Dst: 192.168.1.11 Transmission Control Protocol, Src Port: 5060, Dst Port: 43608, Seq: 1, Ack: 953, Len: 477 Session Initiation Protocol (407) Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bK5c0a58f3707458FA;rport=43608;received=192.168.1.11 From: "6200" <sip:6200@entreprise-108.fr>;tag=59191351-FD3B2D60 To: <sip:0900000000@entreprise-108.fr ;user=phone>;tag=83518db21d5b2e9b777975024049f5a3.8f270000 CSeq: 1 INVITE Call-ID: 727c871081e29672abcb8bd05dde8138 [Generated Call-ID: 727c871081e29672abcb8bd05dde8138] Proxy-Authenticate: Digest realm="entreprise-108.fr", nonce="YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/" Authentication Scheme: Digest Realm: "entreprise-108.fr" Nonce Value: "YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/" Content-Length: 0 Frame 9078: 1259 bytes on wire (10072 bits), 1259 bytes captured (10072 bits) Linux cooked capture Internet Protocol Version 4, Src: 192.168.1.11, Dst: 192.168.1.102 Transmission Control Protocol, Src Port: 43608, Dst Port: 5060, Seq: 1578, Ack: 478, Len: 1191 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:0900000000@entreprise-108.fr;user=phone;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 192.168.1.33;branch=z9hG4bKbca400a5DCDB8264 From: "6200" <sip:6200@entreprise-108.fr>;tag=59191351-FD3B2D60 SIP Display info: "6200" SIP from address: sip:6200@entreprise-108.fr SIP from tag: 59191351-FD3B2D60 To: <sip:0900000000@entreprise-108.fr;user=phone> SIP to address: sip:0900000000@entreprise-108.fr;user=phone CSeq: 2 INVITE Call-ID: 727c871081e29672abcb8bd05dde8138 [Generated Call-ID: 727c871081e29672abcb8bd05dde8138] Contact: <sip:6200@192.168.1.33;transport=tcp> Contact URI: sip:6200@192.168.1.33;transport=tcp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomVVX-VVX_500-UA/5.7.0.14430 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Supported: replaces,100rel Allow-Events: conference,talk,hold Proxy-Authorization: Digest username="6200", realm=" entreprise-108.fr", nonce="YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/", uri=" sip:0900000000@entreprise-108.fr;user=phone;transport=tcp", response="281d775e7166a96d5efe2e100df3df9a", algorithm=MD5 Authentication Scheme: Digest Username: "6200" Realm: "entreprise-108.fr" Nonce Value: "YCPlfGAj5FCsPHbzhSK1i2Oqt9APc1+/" Authentication URI: "sip:0900000000@entreprise-108.fr ;user=phone;transport=tcp" Digest Authentication Response: "281d775e7166a96d5efe2e100df3df9a" Algorithm: MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 270 Message Body Regards, Frederic

1 0

CANCEL transaction
by David Villasmil 10 Feb '21

10 Feb '21

Hello, When receiving a reply, there's a logic which if true, i must cancel the outgoing invite (can 480 the origination). I tried using something like t_set_fr(10,10); to have kamailio cancel it immediately, but it's not happening. Is there a way to cancel an INVITE/transaction? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 7

master daily build with disabled compiler optimisation
by Sergey Safarov 10 Feb '21

10 Feb '21

I want suggested to use "-ggdb3 -O0" gcc options when compiled daily master RPM files. Through?

1 0

Permissions module with Mongodb
by H Yavari 09 Feb '21

09 Feb '21

Hi to all, It's mentioned in the Permissions module doc, we can set DB driver:https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#i… So we can use MongoDB as a DB backend? All the modules that use DB, support Mongo? Thanks. Regards,Hossein

4 13

Kamailio as MS-Teams SBC
by Davide Zanichelli 09 Feb '21

09 Feb '21

Hello everybody! i am trying a setup of kamailio like sbc for ms teams, I currently have the following scenario: TLS Connection OK Out Call ok (except BYE 481 on reply) my problem is in incoming calls microsoft only supports srtp and my pbx (freeswitch based) only works with rtp, now i was reading that you need to use rtpengine to manage srtp to rtp, unfortunately it seems that rtpengine has become very difficult to build especially with recent os, I was wondering if it is possible to have an alternative to rtpengine for managing the srtp (rtpproxy does not seem to work) has anyone managed to get a working setup? in case it is possible to send a configuration template? Thanks You !!! Davide

8 26

run_trans_callbacks_internal crash
by Juha Heinanen 09 Feb '21

09 Feb '21

We got the crash below with 5.4.2. Before the crash there was db_mysql_async_exec_task(): failed to execute query related error messages in syslog. -- Juha Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f601d4b2f32 in run_trans_callbacks_internal (cb_lst=0x7f5ffb53e318, type=512, trans=0x7f5ffb53e2a0, params=0x7ffd1c349720) at t_hooks.c:254 254 t_hooks.c: No such file or directory. (gdb) bt full #0 0x00007f601d4b2f32 in run_trans_callbacks_internal (cb_lst=0x7f5ffb53e318, type=512, trans=0x7f5ffb53e2a0, params=0x7ffd1c349720) at t_hooks.c:254 cbp = 0x72743b6d6f632e61 backup_from = 0x560e8abbe2f0 <def_list+16> backup_to = 0x560e8abbe2f8 <def_list+24> backup_dom_from = 0x560e8abbe300 <def_list+32> backup_dom_to = 0x560e8abbe308 <def_list+40> backup_uri_from = 0x560e8abbe2e0 <def_list> backup_uri_to = 0x560e8abbe2e8 <def_list+8> backup_xavps = 0x560e8abbeae0 <_xavp_list_head> backup_xavus = 0x560e8abbeae8 <_xavu_list_head> backup_xavis = 0x560e8abbeaf0 <_xavi_list_head> __func__ = "run_trans_callbacks_internal" #1 0x00007f601d4b35bb in run_trans_callbacks_with_buf (type=512, rbuf=0x7f5ffb53e370, req=0x7f5ffafe3b70, repl=0xffffffffffffffff, flags=0) at t_hooks.c:303 params = {req = 0x7f5ffafe3b70, rpl = 0xffffffffffffffff, param = 0x7f5ffa61e640, code = 487, flags = 0, branch = 0, t_rbuf = 0x7f5ffb53e370, dst = 0x7f5ffb53e3c0, send_buf = {s = 0x7f5ffa6d4d40 "F", len = 621}} trans = 0x7f5ffb53e2a0 #2 0x00007f601d458025 in relay_reply (t=0x7f5ffb53e2a0, p_msg=0xffffffffffffffff, branch=0, msg_status=408, cancel_data=0x7ffd1c349910, do_put_on_wait=0) at t_reply.c:2124 relay = 0 save_clone = 0 buf = 0x7f601e46a4e0 "SIP/2.0 487 Request Terminated\r\nVia: SIP/2.0/TCP 46.182.160.40;branch=z9hG4bK9ef2.1bbf20df35272cd85c43814302770601.2\r\nVia: SIP/2.0/UDP 95.216.75.211;rport=5060;branch=z9hG4bK9ef2.d6265991eb8da9b735273"... res_len = 621 relayed_code = 487 relayed_msg = 0xffffffffffffffff reply_bak = 0x20000000 bm = {to_tag_val = { s = 0x7f601e46a692 "f148edd50b878efc51a72f445b390303-e1110000\r\nCall-ID: c2e2b55c-e574-1239-3cbd-6cb3112351ce\r\nCSeq: 31882977 INVITE\r\nServer: OpenSIPg SIP Proxy (5.4.2-8 (x86_64/linux))\r\nContent-Length: 0\r\n\r\n", len = 41}} totag_retr = 0 reply_status = RPS_COMPLETED uas_rb = 0x7f5ffb53e370 to_tag = 0x7f601d525770 <tm_tag> reason = {s = 0x560e8aad2dbb "Request Terminated", len = 18} onsend_params = {req = 0x7ffd1c349940, rpl = 0x7f5ffaeeb7c0, param = 0x7ffd1c349880, code = 491640932, flags = 32608, branch = 0, t_rbuf = 0x59d, dst = 0x7f5ffb0ea590, send_buf = {s = 0x7ffd1c349940 "p\231\064\034\375\177", len = 491699179}} ip = {af = 1, len = 0, u = {addrl = {140050215200048, 140050209586032}, addr32 = {4216579376, 32607, 4210965360, 32607}, addr16 = {58672, 64339, 32607, 0, 15216, 64254, 32607, 0}, addr = "0\345S\373_\177\000\000p;\376\372_\177\000"}} __func__ = "relay_reply" #3 0x00007f601d49c7b3 in fake_reply (t=0x7f5ffb53e2a0, branch=0, code=408) at timer.c:295 cancel_data = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 517217456}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 517217456}}}} do_cancel_branch = 0 reply_status = 3845367323 #4 0x00007f601d49cc35 in final_response_handler (r_buf=0x7f5ffb53e540, t=0x7f5ffb53e2a0) at timer.c:462 silent = 0 branch_ret = -1 prev_branch = -1 now = 2146494577 #5 0x00007f601d49cd29 in retr_buf_handler (ticks=2146494577, tl=0x7f5ffb53e560, p=0xfa0) at timer.c:518 rbuf = 0x7f5ffb53e540 fr_remainder = 32608 retr_remainder = 473209280 retr_interval = 32765 new_retr_interval_ms = 547787427795 crt_retr_interval_ms = 140725076662752 t = 0x7f5ffb53e2a0 __func__ = "retr_buf_handler" #6 0x0000560e8a7375fa in slow_timer_main () at core/timer.c:1105 n = 12 ret = 0 tl = 0x7f5ffb53e560 i = 610 __func__ = "slow_timer_main"

2 2

Timeout for provisional response other that 100
by David Villasmil 09 Feb '21

09 Feb '21

Hello all, I’m trying to figure out if I can set. Timeout for specific provisional responses. I know I can set timeouts for INV, and restart the timer every time a provisional response is received. But this not what I actually need. What I need is to set a timeout from the 100 Trying to the next 180/183 (or a final response) and set back a 480 if not received. I’m not sure TM would allow me to do that, so maybe using a hash table to store transactions and calculate the time since the INV to the received 180/183. But there’s gotta be a simpler way. Many thanks! David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Best SIP reply for service activation
by Duarte Rocha 09 Feb '21

09 Feb '21

Hello, I'm currently developing a service on Kamailio where it's possible to configure redirections for a subscriber using a code. In order to implement that client should send an INVITE with a specific code on the Request URI and the redirection is configured. What should be the best way to answer this INVITE? A 200 OK will make the call answered and doesn't seem the best way. A negative reply (4/5/6xx) would work, but it seems strange to have the operation concluded with success and answering with a negative reply. I found "199 - Early Dialog Terminated" to be suitable, but the documentation suggests it should be used upstream. Thanks in advance for any suggestion. Cheers

2 1

Kamailio inbound proxy -> Asterisk
by Mihai 09 Feb '21

09 Feb '21

Hi! Don't know if this is a Kamailio Question, maybe someone had deal with it in the past. I use Kamailio with dispatcher to send the request (registration, invites, all) to my asterisk (inbound) and then the asterisk has the outbound trunks, everything works, but this is little thing in asterisk when entering the "sip show peers" it shows me that all the peers are registered with the kamailio IP, if I set the nat option to no, it will show me the correct IP but breaks other clients. Any idea what to do ?

1 0

How to drop 200 OK in onreply_route ?
by Denys Pozniak 09 Feb '21

09 Feb '21

Hello! In the configuration below Kamailio Proxy creates a transaction for the SIP PUBLISH to get info from the HTTP server in async mode. But before creating a transaction, a synthetic 200 OK is sent, so that I need somehow to drop the real 200 OK from the upstream Presence Server. If drop 200 OK in *reply_route*, tm module starts to retransmit. Those it is necessary that the 200 OK be consumed by the tm module, but does not go further. *if ( !is_method("PUBLISH") ) {* * sl_send_reply("200", "Ok"); t_newtran();* * $http_req(suspend) = 1; http_async_query("$var(url)", "CALLBACK");* *}* *route[CALLBACK] {* * <some header manipulation>* * t_on_reply("PUBLISH_REPLY"); route(RELAY); exit;* *}* *onreply_route[PUBLISH_REPLY] { if ( t_check_status("200") ) { drop; # Does not work!!! }}* Any advice is appreciated -- BR, Denys Pozniak

3 8

Kamailio config file for making Kamailio a Sip Proxy
by Derick Greenlee 08 Feb '21

08 Feb '21

Hello, I had some questions regarding the Kamailio config file. I am trying to make Kamailio just behave as a sip proxy between two sip trunks. Could you send me an example for this possibly?

2 1

authenticated INVITE badly managed when topos is activated
by frédéric Gaisnon 08 Feb '21

08 Feb '21

Hi, If topos is activated and kamailio responds with a 407 "proxy authentication required" to an initial INVITE, INVITE received after with credentials (MD5 response in Proxy-Authorization header) is badly managed. Credentials are never validated and Kamailio responds 407 again. If topoh is used instead topos all works fine and credentials are validated. call flow is: Phone sends INVITE to kamailio SBC kamailio respond 407 "proxy authentication required" because SBC wants to authenticate caller Phone resends INVITE with Proxy-Authorization header with all valid information kamailio responds 407 again instead forwarding INVITE. My code: if (!pv_auth_check("$fd", "$sht(auth_cache=>$var(key))", "0", "1")) { auth_challenge("$fd", “1”); exit; } # user authenticated - remove auth header consume_credentials(); So if topos is used, pv_auth_check always returns false even if the phone sets valid information. If topoh is activated, pv_auth_check always returns true (same phone and same kamailio SBC versioning) I made my tests with kamailio 5.4.3 on centos 7. Regards, Frédéric Gaisnon

1 0

Registered with sip user using user pass
by Gaurav Bmotra 08 Feb '21

08 Feb '21

Hi I m using kamailio 5.3 OS Ubuntu 18 I wan to Register on sip server and then send traffic to to it form kamailio. Like : Client ---(sip peer)-->kamailio------(sip peer)---->gateway Please help me Thank you Regards Gaurav Kumar

1 0

filter_body("application/sdp") on multipart/mixed; boundary with double Content-Length
by Karsten Horsmann 08 Feb '21

08 Feb '21

Hi List, I have some weird multipart/mixed;boundary issues. I use kamailio 5.3.6 with rtpengine 7.x.x (same with 8.x.x). If i try to remove a multipart/mixed and leave only the sdp with filter_body() that works except one thing. The sender used IMHO an non RFC compliant version of rfc5621 and wrote the Content-Length Header twice (one in the header, one in the sdp). After manipulating the SDP filter_body() lets Content-Length in the sdp (i guess). And rtpengine is not happy with the sdp after that. rtpengine [rtpengine.c:2588]: rtpp_function_call(): proxy replied with error: Failed to parse SDP Any ideas how to solve this? Kind regards Karsten Horsmann https://tools.ietf.org/html/rfc5621#section-3.1 route[REMOVE_MULTIPART] { if (has_body("multipart/mixed")) { if ($sel(cfg_get.remove.multipart) == 1) { if (filter_body("application/sdp")) { remove_hf("Content-Type"); append_hf("Content-Type: application/sdp\r\n"); msg_apply_changes(); xlog("L_WARN", "[$cfg(route)] Body part multipart/mixed with application/sdp found\n"); } else { xlog("L_ERROR", "[$cfg(route)] Body part multipart/mixed without application/sdp found\n"); } } else { xlog("L_INFO", "[$cfg(route)] Body part multipart/mixed found - manipulation disabled\n"); } } } ------------------- INVITE sip:+49XXXXXXXX@172.20.120.57:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.120.51:5061;branch=z9hG4bK-6589-1-0 From: sipp <sip:sipp@172.20.120.51:5061>;tag=6589SIPpTag001 To: sut <sip:+49XXXXXXXXX@172.20.120.57:5060> Call-ID: 1-6589(a)172.20.120.51 CSeq: 1 INVITE Contact: sip:sipp@172.20.120.51:5061 Max-Forwards: 70 Subject: Performance Test Content-Type: multipart/mixed;boundary=sonus-content-delim Content-Length: 1185 --sonus-content-delim Content-Type: application/sdp Content-Length: 305 v=0 o=Sonus_UAC 895311 862130 IN IP4 8.8.8.8 s=SIP Media Capabilities c=IN IP4 8.8.8.8 t=0 0 m=audio 26088 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:26089 a=ptime:20 --sonus-content-delim Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:cl="urn:ietf:params:xml:ns:pidf:geopriv10:civicLoc" mlns:btd="http://btd.orange-business.com" entity=" pres:geotarget@btip.orange-business.com"><tuple id="XXXX"><status><gp:geopriv><gp:location-info><cl:civicAddress><cl:country>DE</cl:countr <cl:A1>Some State</cl:A1><cl:A3>Somewhere</cl:A3><cl:A4>Somewhere</cl:A4><cl:A6>Somestreet</cl:A6><cl:YYY>1</cl:YYY><cl:PC>9999</cl:PC><cl:EMERGENCYZONE>XXXXXXX</cl:EMERG CYZONE></cl:civicAddress></gp:location-info><gp:usage-rules></gp:usage-rules></gp:geopriv></status></tuple></presence> --sonus-content-delim-- --------------------- INVITE sip:+49XXXXXXXX@foo SIP/2.0 Via: SIP/2.0/TCP 212.XX.XX.XX;branch=z9hG4bKc228.6b8fdf20d79998d1d355feb2dcf5ba63.0 From: sipp <sip:sipp@foo>;tag=6589SIPpTag001 To: sut <sip:+49XXXXXXXX@foo> Call-ID: 1-6589(a)172.20.120.51 CSeq: 1 INVITE Max-Forwards: 69 Content-Length: 347 Content-Type: application/sdp P-Asserted-Identity: <sip:+49XXXXXXXX@212.XX.XX.XX> Contact: <sip:btpsh-60211bbe-674b-1@212.XX.XX.XX;transport=tcp> Content-Length: 305 v=0 o=Sonus_UAC 895311 862130 IN IP4 8.8.8.8 s=SIP Media Capabilities c=IN IP4 8.8.8.8 t=0 0 m=audio 26088 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:26089 a=ptime:20 -- Mit freundlichen Grüßen *Karsten Horsmann*

1 0

Retransmission of 407
by Marat Gareev 08 Feb '21

08 Feb '21

Hello, UAC sends INVITE to UAS, authorization procedure is required, therefore UAS responds with an 407 error code. At this point, the client does not respond with ACK, but the server does not send retransmits. Is this the expected behavior? * UAC -- SIPp, UAS -- Kamailio. Can the Kamailio be configured to send 407 code retransmissions? Thanks!

3 4

TO Header corrupted in RE-INVITE
by alirsm 07 Feb '21

07 Feb '21

Hi, Call to a number forwarded to a FAX by using two Kamailio proxies. (INVITE) --> KamailioA --> GW1 --> KamailioB --> GW2 --> FAX FAX --> (RE-INVITE) --> GW2 --> KamailioB --> (TO Header being corrupted) KamailioB is corrupting TO header when receiving RE-INVITE and causing the call to fail. 1) Only one "uac_replace_from()" is being used in each Kamailio. 2) GW1 is adding +1 (using E164 format) to URI and FROM headers. I was wondering if any of the above might cause the TO header corruption. Any help would be appreciated. AL

2 1

Failover with Dispatcher Module
by alirsm 05 Feb '21

05 Feb '21

Hi I have setup dispatcher failover on Kamailio 5.1 as below. If destination X.X.X.X is down, traffic should be routed to Y.Y.Y.Y. The failover is working but every time after 30 sec Timeout. Is there a way to setup health check properly so the failed destination can be ignored while it is down? Your help is greatly appreciated, AL # dispatcher.list 1 sip:X.X.X.X:5060 0 0 1 sip:Y.Y.Y.Y:5060 0 0 # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_inactive_threshold", 4) modparam("dispatcher", "ds_ping_latency_stats", 1) modparam("dispatcher", "ds_probing_threshold", 1) # Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "4")) { send_reply("503", "No destination"); exit; } t_on_failure("DISPATCH_FAILOVER"); route(RELAY); exit; } # Failure Route failure_route[DISPATCH_FAILOVER]{ if (t_is_canceled()) exit; if(!ds_next_dst()) { send_reply("503", "Out of gateways"); exit; } t_on_failure("DISPATCHER_FAILOVER"); t_relay(); }

2 1

Make configuration as proxy, push, renew server
by James Chan 02 Feb '21

02 Feb '21

Hi We use a Cloud-based SIP-server that does not support the Push-Notifications Service. We try to use Kamailio as the front-end to work as a proxy and push-server. We need help configuring Kamailio as a proxy and push-server. In addition, we want to implement auto renew configuration as UAC using Kamailio. Because all clients need to renew registration our Cloud-SIP-server every 59 minutes. ( registration expires after 60 minutes) Summary: UAC -> Kamailio (proxy & push & auto renew) -> Cloud-SIP-Server -> UAC I have attached our config file and screens And if phone has multiple contacts, there is call bug, call will be drop since there are two incoming call #!KAMAILIO #!define WITH_DEBUG #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_ACCDB #james #!define WITH_UAC #!define WITH_TSILO # # Kamailio (OpenSER) SIP Server v5.3 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!ifdef WITH_TSILO #!define FLB_TSILO 8 #!endif ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif fork=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:185.224.139.45:5060 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib64/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_UAC loadmodule "uac.so" #!endif #!ifdef WITH_TSILO loadmodule "tsilo.so" loadmodule "htable.so" loadmodule "cfgutils.so" loadmodule "http_client.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_UAC modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- 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", 1) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif #!ifdef WITH_UAC modparam("uac", "reg_contact_addr", "pn.telesonique.ch:5060;transport=udp") modparam("uac", "reg_db_url", DBURL) modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 300) modparam("uac", "reg_db_table", "uacreg") #!endif #!ifdef WITH_TSILO modparam("htable", "htable", "vtp=>size=10;autoexpire=180") modparam("http_client", "httpcon", "pushapi=>https://www.myswissline.com") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if (is_method("REGISTER")) { save("location", "0x06"); } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } #james set_forward_no_connect(); if (!t_relay()) { sl_reply_error(); } #!ifdef WITH_TSILO else if(isflagset(FLB_TSILO)) { resetflag(FLB_TSILO); if (ts_store()) { $sht(tsilo=>stored::$rU) = 1; xdbg("stored transaction [$T(id_index):$T(id_label)] $fU => $rU\n"); } route(PUSH_NOTIF); } #!endif exit; } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies #set_reply_no_connect(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # 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 statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } save("location", "0x06"); switch($rc) { case -1: case -2: sl_reply_error(); break; #!ifdef WITH_TSILO case 1: $var(new_contact) = 1; #!endif } #!ifdef WITH_TSILO if ($var(new_contact)) route(WAKE_SUSPENDED); #!endif exit; } #!ifdef WITH_TSILO route[WAKE_SUSPENDED] { $var(suspended) = 0; lock("$tU"); $var(suspended) = $sht(tsilo=>suspended::$tU); $sht(tsilo=>suspended::$tU) = $null; $var(stored) = $sht(tsilo=>stored::$tU); unlock("$tU"); if ($var(suspended) == 0) { if ($var(stored)) { # no suspended INVITE (i.e. there was at least one # registered contact for target) and 'stored' set # (i.e. ongoing transaction) -> this newly just # registered contact needs alerting. ts_append("location", "$tu"); } return; } $var(id_index) = $(var(suspended){s.select,0,:}{s.int}); $var(id_label) = $(var(suspended){s.select,1,:}{s.int}); xdbg("resuming trasaction [$var(id_index):$var(id_label)] $tU .\n"); t_continue("$var(id_index)", "$var(id_label)", "RESUME"); } route[RESUME] { if(! lookup("location")) { xlog("failed to load just added destinations\n"); exit; } setflag(FLB_TSILO); route(RELAY); } #!endif # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; #!ifdef WITH_TSILO send_reply("100", "Trying"); route(SUSPEND); route(PUSH_NOTIF); #!else route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } #!endif } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } #!ifdef WITH_TSILO setflag(FLB_TSILO); #!endif route(RELAY); exit; } #!ifdef WITH_TSILO route[SUSPEND] { if(! t_suspend()) { xlog("SUSPEND: failed to suspend transaction.\n"); # TODO: actually, the destination is not really checked: don't know if it's # own subscriber or invalid dialing in the first place. send_reply("503", "Temporarily unavailable"); exit; } $sht(tsilo=>suspended::$rU) = $_s($T(id_index):$T(id_label)); xdbg("suspended transaction [$T(id_index):$T(id_label)] $fU => $rU\n"); } route[PUSH_NOTIF] { xdbg("TODO: send push for $rU\n"); $var(res) = http_connect("pushapi", "/send_push.php?username=$tU&dev=1", "application/json", "", "$avp(gurka)"); xlog("L_INFO", "Push notification: $avp(gurka) Result code $var(res)\n"); if ($var(res)==200) { if ($var(gurka)=~"^.*rejected.*$") { sl_send_reply("503", "Not available right now"); } else { sl_send_reply("101", "Push Sent"); } } if ($var(res)==28) { xlog("L_INFO", "Push notification timed out. Do we care?"); } exit; } #!endif # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!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; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_TSILO return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } [image: call_issue.png][image: register.png]

1 0

failover on SIP servers
by AL RSM 02 Feb '21

02 Feb '21

Hi, I was wondering what is the recommended approach to implement a failover on several destination SIP servers: 1) using DNS SRV records 2) using dispatcher list 3) combination of both. Thanks, AL

3 2

Acc module unable to insert cdr in mysql server
by M Arqum CH 01 Feb '21

01 Feb '21

Hello Members, Kamailio acc module is not inserting cdr into MySQL DB, sometimes it works and sometimes didn't, the number of parameters/values are similar in failed as well success scenario. I also check with tcpdump for MySQL requests, unfortunately, Kamailio didn't issue insert query toward DB server, so it is clear to me that the MySQL server is nothing to do with failed queries. Please guide me on what I should do to find and fix the issue. Kamailio Installed version : kamailio -v version: kamailio 5.4.1 (x86_64/linux) 2ca223 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 2ca223 compiled on 12:51:24 Sep 21 2020 with gcc 6.3.0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:691]: tmcb_func(): acc callback called for t(0x7fe29b2a7870) event type 512, reply code 200 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:413]: should_acc_reply(): probing acc state - code: 200 flags: 0x6 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:446]: should_acc_reply(): acc is on Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 6 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 7 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 10 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 13 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 16 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 17 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 19 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 22 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 26 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 28 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 29 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 32 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 33 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: tm [t_hooks.c:256]: run_trans_callbacks_internal(): DBG: trans=0x7fe29b2a7870, callback type 512, id 0 entered Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 3 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 14 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 18 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 21 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 24 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 25 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: ERROR: acc [acc_cdr.c:267]: db_write_cdr(): failed to insert into database Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 3 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 14 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 18 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 21 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 24 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 25 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: ERROR: acc [acc_cdr.c:682]: cdr_on_end_confirmed(): failed to write cdr! Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: tm [t_reply.c:1758]: cleanup_uac_timers(): RETR/FR timers reset Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103682]: DEBUG: tm [timer.c:634]: wait_handler(): finished transaction: 0x7fe29727ba88 (p:0x7fe294d27288/n:0x7fe294d27288) -- Regards M Arqum

2 2

UACREG to many Asterisk
by Javier Valencia 01 Feb '21

01 Feb '21

Hi there! Go ahead I apologize for my English. I have a Kamailio 5.4 with the DISPATCHER module and UAC enabled (for remote REGISTER), the DISPATCHER has several SETIDs, and they have one or more Asterisk balanced behind it. In Kamalio script, when an endpoint does REGISTER, I enable the REGISTER of the UAC module. And when it expires or expires=0 I disable the REGISTER in the UAC module. In the UACREG table in the database, L_UUID (unique key), L_USERNAME, R_USERNAME, and AUTH_USERNAME have the same value. For example CST100EXT100. Knowing that the DISPATCHER and UAC modules have no relationship, but that when a SETID has several balanced Asterisks, how can I make the UAC module send the REGISTER to all the Asterisks of a SETID? Clarifications; 1.- I cannot update the Asterisk, which are currently in version 11. They do not support PATH. 2.- I can modify the content of the UACREG table so that l_uuid has a suffix '_X'. 3.- Call jsonrpc_exec several times to enable the REGISTER in the UAC module. (diff emulated) < jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au"]}') ; > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au_1"]}') ; > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au_2"]}') ; > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au_3"]}') ; But I want to know if there is any more elegant way to do this. Cheers, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

3 2

externally triggered message sending to specific clients
by Syed Shahryar 01 Feb '21

01 Feb '21

Is it possible to send arbitrary text messages triggered by an external application (evapi/http) to registered SIP clients via websocket? The following doesn't seem to do anything. KSR.corex.send_data("sip:user@example.com", "Message at $Ts"); -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

about kamailio-sbc-teams tutorial
by Juha Heinanen 01 Feb '21

01 Feb '21

I read kamailio-sbc-teams tutorial https://skalatan.de/blog/kamailio-sbc-teams and it does not mention, which CA root certificates need to be included tls config ca_list so that Kamalio is able to verify sipX.pstnhub.microsoft.com server certificates. -- Juha

2 1

logging error events to database
by Sergiu Pojoga 30 Jan '21

30 Jan '21

Hi there, I was wondering if there's a way to log various core error events in some DB. For example, below error occurred Jan 29 11:43:56 kamailio[11076]: {1 11440 INVITE 28b653bd-8f62-4085-a5ef-f2a4ac0f393d } ERROR: <core> [core/parser/sdp/sdp_helpr_funcs.c:499]: extract_mediaip(): no `IP[4|6]' address in `c=' field Jan 29 11:43:56 kamailio[11076]: {1 11440 INVITE 28b653bd-8f62-4085-a5ef-f2a4ac0f393d } ERROR: <core> [core/parser/sdp/sdp.c:430]: parse_sdp_session(): can't extract common media IP from the message Thanks in advance. Regards, --Sergiu

4 5

Establish connection between MsTeams and Kamailio.
by Willy Valles Rios 30 Jan '21

30 Jan '21

Hello community, Please your support, I tell you that I am integrating Kamailio with MsTeams, after some time reviewing this, I finally achieved the connection from my Kamailio server to MsTeams and I can validate that the connection to MsTeams is in AP. [root @ kamailio-server kamailio] # kamcmd dispatcher.list | egrep "URI | FLAGS" URI: sip: sip.pstnhub.microsoft.com; transport = tls FLAGS: AP However, from the admin panel of MsTeams (Direct Routing) I see that the connection to my sbc "sbc.netvoiceperu.com" is with TLS connectivity status in "Active" but the SIP options status is in "Warning". I have made calls from MsTeams thinking that the SIP options status would change to "active" but it is still in "Warning" state. On the other hand, I have enabled a siptrace in Kamailio and verify that the SIP OPTIONS from kamailio are being sent in the following format to MsTeams. OPTIONS sip: sip.pstnhub.microsoft.com; transport = tls SIP / 2.0 Via: SIP / 2.0 / TLS 161.35.44.66:5061 ;branch=z9hG4bKea07.52224687000000000000000000000000.0 To: <sip: sip.pstnhub.microsoft.com; transport = tls> From: <sip: sbc.netvoiceperu.com>; tag = d3569c818b500aeb8c373426e76c2884-81763c71 CSeq: 10 OPTIONS Call-ID: 13ea237a751e0c48-9148(a)161.35.44.66 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.4.0 (x86_64 / linux)) As you can see, the SIP OPTIONS sent from Kamailio to MsTeams does not contain the "Contact" field, which in theory said "Contact" field should have been added by Kamailio according to the configuration added in kamailio.cfg event_route [tm: local-request] { sip_trace (); if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") { append_hf ("Contact: <sip: sbc.netvoiceperu.com: 5061; transport = tls> \ r \ n"); } xlog ("L_INFO", "Sent out tm request: $ mb \ n"); } As additional information, I inform you that I also managed to observe the SIP OPTIONS that MsTeams sends to Kamailio. OPTIONS sip: sbc.netvoiceperu.com: 5061; transport = tls SIP / 2.0 FROM: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061>; tag = f1bdeb5f-662f-4544-a436-e9aa9ad78da4 TO: <sip: sbc.netvoiceperu.com> CSEQ: 1 OPTIONS CALL-ID: c47e2782-16c3-49cb-8931-24e9709d260a MAX-FORWARDS: 70 VIA: SIP / 2.0 / TLS 52.114.75.24:5061;branch=z9hG4bK48b0e6be CONTACT: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.1.15.7 i.EUWE.10 ALLOW: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY However I don't see the 200 OK SIP responses from Kamailio to MsTeams. I think this may be the reason why I see the SIP OPTIONS status in "Warning" from the MsTeams panel. Maybe the contact field is not being added in the SIP OPTIONS messages that Kamailio sends to MsTeams and for that reason I don't see 200OK responses from MsTeams. Could you help me solve this please. Cheers Saludos Cordiales -- *Willy Valles Rios* *Unified Communications Specialist* phone: +51955747343 em@il: willyvalles17(a)gmail.com

2 1

registering gsm gateway by siremis
by shashank shekhar 29 Jan '21

29 Jan '21

Hi, I want to register a gsm gateway on my kamailio server with siremis ...kindly guide me on this Thankyou Regards Shashank shekhar 9162735161

2 1

Kamailio as inbound and outbound proxy
by Michael Friesen 29 Jan '21

29 Jan '21

Hello, I am trying to configure Kamailio as both an inbound and outbound proxy. Inbound requests flow Sip Trunk -> Kamailio -> Asterisk. Outbound requests flow Asterisk -> Kamailio -> Sip Trunk. Inbound traffic is sent to port 5060 on kamailio which listens on the PRIVATE_IP:5060 and advertises PUBLIC_IP:5060. How can we ensure that when messages are sent to the Sip Trunk they have Record-Route headers with the PUBLIC_IP and when messages are sent to asterisk they have Record-Route headers with the PRIVATE_IP?

2 1

kamailio 5.4.3 ubuntu 20.04 tls - http_async_client
by Filippo Graziola 28 Jan '21

28 Jan '21

Hi all, I have an issue related (my guess) to tls and http_async_client module that result in a segmentation fault and a not correct handle of tls connections. First with only tls module loaded, not forked: 0(1021) INFO: <core> [core/tcp_main.c:4983]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1021) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1021) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1021) INFO: tls [tls_mod.c:389]: mod_init(): With ECDH-Support! 0(1021) INFO: tls [tls_mod.c:392]: mod_init(): With Diffie Hellman 0(1021) WARNING: tls [tls_init.c:784]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 4718592 and 2359296 bytes 0(1021) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 4718592 0(1021) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2359296 0(1021) INFO: <core> [main.c:2833]: main(): processes (at least): 9 - shm size: 67108864 - pkg size: 67108864 0(1021) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1021) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1021) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs<default>: tls_method=12 0(1021) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/fullchain.pem' 0(1021) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' 0(1021) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' 0(1021) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 0(1021) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' 0(1021) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/privkey.pem' 0(1021) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 0(1021) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 0(1021) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs<default>: verify_client=0 0(1021) NOTICE: tls [tls_domain.c:1105]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... 0(1021) INFO: tls [tls_domain.c:711]: set_verification(): TLSs<default>: No client certificate required and no checks performed 0(1021) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 0(1021) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' 0(1021) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' 0(1021) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' 0(1021) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=0 0(1021) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' 0(1021) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' 0(1021) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=0 0(1021) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 0(1021) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc<default>: verify_client=0 0(1021) INFO: tls [tls_domain.c:714]: set_verification(): TLSc<default>: Server MAY present invalid certificate 6(1027) ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error 6(1027) ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:141FC044:SSL routines:tls_setup_handshake:internal error 6(1027) ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: XXXXXXXXXXXXXXX 6(1027) ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: XXXXXXXXXX 6(1027) ERROR: <core> [core/tcp_read.c:1498]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f2cbc1b3948 r: 0x7f2cbc1b3a70 (-1) so no segmentation fault but error in handling. Second one also with http_async_client loaded: 0(1059) INFO: <core> [core/tcp_main.c:4983]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1061) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1061) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1061) INFO: tls [tls_mod.c:389]: mod_init(): With ECDH-Support! 0(1061) INFO: tls [tls_mod.c:392]: mod_init(): With Diffie Hellman 0(1061) INFO: http_async_client [http_async_client_mod.c:222]: mod_init(): Initializing Http Async module 0(1061) WARNING: tls [tls_init.c:784]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 5242880 and 2621440 bytes 0(1061) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 5242880 0(1061) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2621440 0(1061) INFO: <core> [main.c:2833]: main(): processes (at least): 10 - shm size: 67108864 - pkg size: 67108864 0(1061) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1061) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1061) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs<default>: tls_method=12 0(1061) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/fullchain.pem' 0(1061) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' 0(1061) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' 0(1061) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 0(1061) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' 0(1061) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/privkey.pem' 0(1061) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 0(1061) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 0(1061) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs<default>: verify_client=0 0(1061) NOTICE: tls [tls_domain.c:1105]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... 0(1061) INFO: tls [tls_domain.c:711]: set_verification(): TLSs<default>: No client certificate required and no checks performed 0(1061) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 0(1061) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' 0(1061) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' 0(1061) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' 0(1061) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=0 0(1061) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' 0(1061) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' 0(1061) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=0 0(1061) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 0(1061) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc<default>: verify_client=0 0(1061) INFO: tls [tls_domain.c:714]: set_verification(): TLSc<default>: Server MAY present invalid certificate 0(1061) INFO: http_async_client [async_http.c:101]: async_http_init_sockets(): inter-process event notification sockets initialized 0(1061) INFO: http_async_client [async_http.c:84]: async_http_init_worker(): started worker process: 1 0(1059) CRITICAL: <core> [core/mem/q_malloc.c:501]: qm_free(): BUG: bad pointer 0x1 (out of memory block!) called from tls: tls_init.c: ser_free(323) - ignoring Segmentation fault this time, there is a segmentation fault. The above is a result of this minimal configuration: #!KAMAILIO ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " children=2 tcp_children=2 auto_aliases=no alias="XXXXXXXXXXXXX" listen=udp:eth0 server_signature=no tcp_connection_lifetime=3605 tcp_max_connections=40960 tcp_accept_no_cl=yes enable_tls=yes listen=tls:XXXXXXXXXX:5061 advertise XXXXXXXXXXXX:5061 tls_max_connections=40000 enable_sctp=no ####### Modules Section ######## loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "tls.so" loadmodule "http_async_client.so" #----------------- setting module-specific parameters --------------- #----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #----- http client ---- modparam("http_async_client", "workers", 1) ####### Routing Logic ######## request_route { exit; } I used the above configuration to take out as much as possible my mistakes in the configuration, but with my full kamailio configuration, tls connections give the above errors but everything else works just fine (also http_async_client module functions which are used on INVITES) and calls are going properly (unfortunately tls is required). I found a couple of issues that are similar https://github.com/kamailio/kamailio/issues/2560 and https://github.com/kamailio/kamailio/issues/2466# but as far as I understood the issue 2466 is closed because fixes are already included. I tried in any case to compile from source a few older releases with the same result, changed also the certificate and private key (letsencrypt), moreover I have another kamailio (v5.3.4) running on ubuntu 18.04 (same configuration) without any issues. I saw that there is a different version of openssl version 1.0.. in ubuntu 18.04, version 1.1 in ubuntu 20.04, but the segmentation fault seems to happen after an error on free some memory. Have you some ideas? tell me if you need more info from me. Thanks Filippo

4 7

Problems establishing SIP signaling between MsTeams and Kamailio
by Willy Valles Rios 28 Jan '21

28 Jan '21

Hello community, I am having trouble establishing SIP signaling between MsTeams and Kamailio. I currently have this configuration in my tls.cfg file [server: default] method = TLSv1.2 + verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certificates/private-key.pem certificate = /etc/kamailio/certificates/certificate.pem [client: default] method = TLSv1.2 + verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certificates/private-key.pem certificate = /etc/kamailio/certificates/certificate.pem My domain was certified with ssl through an authoritative certifier (GoDaddy), however I see these errors in the / var / log / messages of the Kamailio server. Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_mod.c:389]: mod_init(): With ECDH-Support! Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_mod.c:392]: mod_init(): With Diffie Hellman Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_init.c:722]: tls_h_mod_init_f(): compiled with openssl version "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, compression: on Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_init.c:730]: tls_h_mod_init_f(): installed openssl library version "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, zlib compression: on#012 compiler: gcc -I. -I.. -I../include -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic -Wa,--noexecstack -DPURIFY -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM -DECP_NISTZ256_ASM Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: WARNING: tls [tls_init.c:787]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 13107200 and 6553600 bytes Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 13107200 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 6553600 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [main.c:2834]: main(): processes (at least): 25 - shm size: 67108864 - pkg size: 4194304 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs: tls_method=22 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs: certificate='/etc/kamailio/certificados/certificate.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs: ca_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs: crl='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:335]: ksr_tls_fill_missing(): TLSs: require_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs: cipher_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs: private_key='/etc/kamailio/certificados/private-key.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:353]: ksr_tls_fill_missing(): TLSs: verify_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs: verify_depth=9 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs: verify_client=0 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: NOTICE: tls [tls_domain.c:1107]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='' ... Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:697]: set_verification(): TLSs: Client MUST present valid certificate Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc: tls_method=22 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc: certificate='/etc/kamailio/certificados/certificate.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc: ca_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc: crl='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:335]: ksr_tls_fill_missing(): TLSc: require_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc: cipher_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc: private_key='/etc/kamailio/certificados/private-key.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:353]: ksr_tls_fill_missing(): TLSc: verify_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc: verify_depth=9 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc: verify_client=0 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:697]: set_verification(): TLSc: Server MUST present valid certificate Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32422]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/32422 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32424]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.75.24 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f45242be028 r: 0x7f45242be150 (-1) Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.132.46 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f45242d9278 r: 0x7f45242d93a0 (-1) Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.14.70 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f45242be028 r: 0x7f45242be150 (-1) Could you help me identify the problem please. Cheers Saludos Cordiales -- *Willy Valles Rios* *Unified Communications Specialist* phone: +51955747343 em@il: willyvalles17(a)gmail.com

9 18

Info: Upcoming online events - February 2021
by Daniel-Constantin Mierla 27 Jan '21

27 Jan '21

Hello, short note to announce a news article listing the upcoming online events where you can watch presentation related to Kamailio, respectively FOSDEM and ClueCon TGI. More details at: * https://www.kamailio.org/w/2021/01/upcoming-online-events-february-2021/ During this time where the travelling is under heavy restrictions, such events can be a chance to keep in touch with peers in the RTC space or learn new usefuk things. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

Loop with unable to find dialog for xxx
by David VILLAUME 27 Jan '21

27 Jan '21

Hello, I got some time Ack packets looping on my servers during 4 hours if I don’t restart the system ( 4 hours is my dialog timeout). Jan 27 16:59:02 vma-2 kamailio[4544]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4544]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' It generates approximatively 2 millions packets per loop. On the last occurrence I noticed a ‘SIP/2.0 200 canceling’ generated per Kamailio before the loop started (which seems uncomon on this host) Bellow is a screenshot of the beginning of the loop and may dialog and topos config. Has anyone encountered such behavour ? Regards, David [cid:image001.png@01D6F4E2.F68B5140] modparam("dialog", "dlg_flag", 1) modparam("dialog", "hash_size", 1024) modparam("dialog", "default_timeout", 14500) modparam("dialog", "event_callback", "ksr_dialog_event") modparam("dialog", "db_url", DBURL_DLG) modparam("dialog", "db_update_period", 15) modparam("dialog", "db_mode", 1) modparam("topos", "storage", "redis") modparam("topos", "dialog_expire", 14500) modparam("topos", "contact_host", "PUBLICIP") modparam("topos_redis", "serverid", "topos")

1 0

websocket rd buffer size limit?
by Andrew Chen 27 Jan '21

27 Jan '21

Hey group, I was wondering if there is a way to control the websocket buffer size to support large SIP messages. I received the following error when sending content-length of 65000 2021-01-26T17:32:19.251408+00:00 sjomainkama50 kamailio[31871]: 109(32008) WARNING: websocket [*ws_frame.c*:482]: decode_and_validate_ws_frame(): message is too long for our buffer size (65535 / 66151) I am assuming this is controlled by the tcp_rd_buf params which I have set to 131k: *tcp_*rd_buf_size=131072 and this is my tcp_wr_buf params: *tcp_*wq_blk_size=131072 *tcp_*conn_wq_max=131072 Please advise. Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 4

event_route[core:receive-parse-error] missing from docs
by Henning Westerholt 27 Jan '21

27 Jan '21

Hello, I noticed that the core "event_route[core:receive-parse-error]" was missing from the cookbook, so I have added it there. https://www.kamailio.org/wiki/cookbooks/devel/core#event_route Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

FQDN endpoint as rd
by AL RSM 27 Jan '21

27 Jan '21

Hi, I was wondering if Kamailio can support FQDN endpoint which resolves to multiple IPs, checks if the first IP is not reachable, try the next one in the list. Thanks, AL

2 1

Re: [SR-Users] sr-users Digest, Vol 188, Issue 9
by Willy Valles Rios 26 Jan '21

26 Jan '21

Hello community, After a while, I finally got the connection from my Kamailio server to MsTeams and I can validate that the connection to MsTeams is in AP. [root @ kamailio-server kamailio] # kamcmd dispatcher.list | egrep "URI | FLAGS" URI: sip: sip.pstnhub.microsoft.com; transport = tls FLAGS: AP However, from the admin panel of MsTeams (Direct Routing) I see that the connection to my sbc "sbc.netvoiceperu.com" is with TLS connectivity status in "Active" but the SIP options status is in "Warning". I have made calls from MsTeams thinking that the SIP options status would change to "active" but it is still in "Warning" state. On the other hand, I have enabled a siptrace in Kamailio and verify that the SIP OPTIONS from kamailio are being sent in the following format to MsTeams. OPTIONS sip: sip.pstnhub.microsoft.com; transport = tls SIP / 2.0 Via: SIP / 2.0 / TLS 161.35.44.66:5061 ;branch=z9hG4bKea07.52224687000000000000000000000000.0 To: <sip: sip.pstnhub.microsoft.com; transport = tls> From: <sip: sbc.netvoiceperu.com>; tag = d3569c818b500aeb8c373426e76c2884-81763c71 CSeq: 10 OPTIONS Call-ID: 13ea237a751e0c48-9148(a)161.35.44.66 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.4.0 (x86_64 / linux)) As you can see, the SIP OPTIONS sent from Kamailio to MsTeams does not contain the "Contact" field, which in theory said "Contact" field should have been added by Kamailio according to the configuration added in kamailio.cfg event_route [tm: local-request] { sip_trace (); if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") { append_hf ("Contact: <sip: sbc.netvoiceperu.com: 5061; transport = tls> \ r \ n"); } xlog ("L_INFO", "Sent out tm request: $ mb \ n"); } As additional information, I inform you that I also managed to observe the SIP OPTIONS that MsTeams sends to Kamailio. OPTIONS sip: sbc.netvoiceperu.com: 5061; transport = tls SIP / 2.0 FROM: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061>; tag = f1bdeb5f-662f-4544-a436-e9aa9ad78da4 TO: <sip: sbc.netvoiceperu.com> CSEQ: 1 OPTIONS CALL-ID: c47e2782-16c3-49cb-8931-24e9709d260a MAX-FORWARDS: 70 VIA: SIP / 2.0 / TLS 52.114.75.24:5061;branch=z9hG4bK48b0e6be CONTACT: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.1.15.7 i.EUWE.10 ALLOW: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY However I don't see the 200 OK SIP responses from Kamailio to MsTeams. I think this may be the reason why I see the SIP OPTIONS status in "Warning" from the MsTeams panel. Maybe the contact field is not being added in the SIP OPTIONS messages that Kamailio sends to MsTeams and for that reason I don't see 200OK responses from MsTeams. Could you help me solve this please. Cheers Saludos Cordiales -- *Willy Valles Rios* *Unified Communications Specialist* phone: +51955747343 em@il: willyvalles17(a)gmail.com El dom, 10 ene 2021 a las 6:03, <sr-users-request(a)lists.kamailio.org> escribió: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Problems establishing SIP signaling between MsTeams and > Kamailio (Carlos Mestanza T.) > 2. Re: Problems establishing SIP signaling between MsTeams and > Kamailio (Rob van den Bulk) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 8 Jan 2021 19:16:30 -0500 > From: "Carlos Mestanza T." <mestacart(a)gmail.com> > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org>, "Daniel Constantin Mierla > (asipto)" > <miconda(a)gmail.com> > Subject: Re: [SR-Users] Problems establishing SIP signaling between > MsTeams and Kamailio > Message-ID: > < > CAHcXM7PN0zuJC9jONTERZdfur3b+ucwnCC712Ss1OBxCiMctUg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Dear Kamilio Community, > > After doing several reads to our Kamailio configuration, the dispatcher > list is in AP: > > URI: sip: sip.pstnhub.microsoft.com; transport = tls > FLAGS: AP > URI: sip: sip2.pstnhub.microsoft.com; transport = tls > FLAGS: AP > URI: sip: sip3.pstnhub.microsoft.com; transport = tls > FLAGS: AP > > But in the MS Teams dashboard the SIP OPTIONS STATUS column is WARNING. > > In the LOGs sent to OPTIONS, it gives us to understand that with sent to MS > TEAMS: > > Jan 8 19:01:40 Kamailio-Server /usr/sbin/kamailio[1444]: INFO: <script>: > Sent out tm request: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls > SIP/2.0#015#012Via: SIP/2.0/TLS > 161.35.44.66:5061 > ;branch=z9hG4bKd25f.2835f676000000000000000000000000.0#015#012To: > <sip:sip.pstnhub.microsoft.com;transport=tls>#015#012From: <sip: > sbc.netvoiceperu.com > >;tag=69ae0da9200ed8d142f2e4a69f531080-213e3c71#015#012CSeq: > 10 OPTIONS#015#012Call-ID: > 07561978687e60d0-1444(a)10.131.245.99#015#012Max-Forwards > <http://07561978687e60d0-1444@10.131.245.99#015%23012Max-Forwards>: > 70#015#012Content-Length: 0#015#012User-Agent: kamailio (5.4.0 > (x86_64/linux)) > Jan 8 19:01:40 Kamailio-Server /usr/sbin/kamailio[1444]: INFO: <script>: > Sent out tm request: OPTIONS sip:sip2.pstnhub.microsoft.com;transport=tls > SIP/2.0#015#012Via: SIP/2.0/TLS > 161.35.44.66:5061 > ;branch=z9hG4bKe25f.b14dc514000000000000000000000000.0#015#012To: > <sip:sip2.pstnhub.microsoft.com;transport=tls>#015#012From: <sip: > sbc.netvoiceperu.com > >;tag=69ae0da9200ed8d142f2e4a69f531080-44c3af70#015#012CSeq: > 10 OPTIONS#015#012Call-ID: > 07561978687e60d1-1444(a)10.131.245.99#015#012Max-Forwards > <http://07561978687e60d1-1444@10.131.245.99#015%23012Max-Forwards>: > 70#015#012Content-Length: 0#015#012User-Agent: kamailio (5.4.0 > (x86_64/linux)) > Jan 8 19:01:40 Kamailio-Server /usr/sbin/kamailio[1444]: INFO: <script>: > Sent out tm request: OPTIONS sip:sip3.pstnhub.microsoft.com;transport=tls > SIP/2.0#015#012Via: SIP/2.0/TLS > 161.35.44.66:5061 > ;branch=z9hG4bKb25f.8442f914000000000000000000000000.0#015#012To: > <sip:sip3.pstnhub.microsoft.com;transport=tls>#015#012From: <sip: > sbc.netvoiceperu.com > >;tag=69ae0da9200ed8d142f2e4a69f531080-fa555adb#015#012CSeq: > 10 OPTIONS#015#012Call-ID: > 07561978687e60d2-1444(a)10.131.245.99#015#012Max-Forwards > <http://07561978687e60d2-1444@10.131.245.99#015%23012Max-Forwards>: > 70#015#012Content-Length: 0#015#012User-Agent: kamailio (5.4.0 > (x86_64/linux)) > Jan 8 19:01:42 Kamailio-Server journal: Suppressed 103 messages from > /system.slice/kamailio.service > > In the kamailio.cfg configuration it is declared: > > listen=tls:161.35.44.66:5061 > listen=tcp:10.131.245.99:5061 > > modparam ("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") > modparam ("dispatcher", "ds_probing_mode", 1) > modparam ("dispatcher", "ds_ping_interval", 60) > > And this fragment was also added: > > event_route [tm: local-request] { > > if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") { > append_hf ("Contact: <sip: sbc.netvoiceperu.com: 5061; > transport = tls> \ r \ n"); > } > xlog ("L_INFO", "Sent out tm request: $ mb \ n"); > } > > There is something additional that has to be declared so that in the MS > Teams panel the SIP OPTIONS STATUS column is shown as ACTIVE and not as > WARNNING, in the MS TEAMS documentation it is a possible problem related to > OPTIONS events. > > Atentamente > > *Adalberto Carlos Mestanza T.* > > > > El jue, 7 ene 2021 a las 21:54, Ovidiu Sas (<osas(a)voipembedded.com>) > escribi?: > > > That certificate should already be present under the OS's trusted > > certificates directory (debian and ubuntu certs are stored under > > /etc/ssl/certs), maybe under a different name, and is required for > > remote endpoint's certificate validation. > > One can load a particular certificate or a list of certificates. > > Multiple certificates can be concatenated into one single file as > > stated in the documentation: > > https://kamailio.org/docs/modules/devel/modules/tls.html#tls.p.ca_list > > Hope this helps a little bit in understanding of the ca_list param. > > > > Regards, > > Ovidiu Sas > > > > On Thu, Jan 7, 2021 at 8:10 AM <rob.van.den.bulk(a)gmail.com> wrote: > > > > > > I Used this tls.cfg > > > > > > > > > > > > Use bc2025.pem as extra, Microsoft needs this? > > > > > > > > > > > > And works fine on different Kamailio-msteams sbcs > > > > > > > > > > > > > > > > > > [server:default] > > > > > > method = TLSv1.2+ > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/privkey.pem > > > > > > certificate = > > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/fullchain.pem > > > > > > ca_list = /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/bc2025.pem > > > > > > server_name = sbc.combivoipdom.nl > > > > > > > > > > > > [client:default] > > > > > > method = TLSv1.2+ > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/privkey.pem > > > > > > certificate = > > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/fullchain.pem > > > > > > ca_list = /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/bc2025.pem > > > > > > > > > > > > > > > > > > > > > > > > Cheers Rob > > > > > > > > > > > > Van: sr-users <sr-users-bounces(a)lists.kamailio.org> Namens > > Daniel-Constantin Mierla > > > Verzonden: donderdag 7 januari 2021 08:53 > > > Aan: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org > >; > > Willy Valles Rios <willyvalles17(a)gmail.com> > > > CC: Carlos Mestanza T. <mestacart(a)gmail.com> > > > Onderwerp: Re: [SR-Users] Problems establishing SIP signaling between > > MsTeams and Kamailio > > > > > > > > > > > > Does this happen when Kamailio connects to MS Teams? The logs indicate > > the received TLS certificate is not trusted: > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > > > > > > > You can set debug=3 in kamailio.cfg and see if the DEBUG messages > > provide more hints. For me it worked fine with Letsencrypt certs in > > Kamailio and accepting what ever MS sent back. I used Debian 10 and > libssl > > 1.1. > > > > > > > > > > > > Cheers, > > > Daniel > > > > > > > > > > > > On 06.01.21 21:47, Willy Valles Rios wrote: > > > > > > Hello community, > > > > > > > > > > > > I am having trouble establishing SIP signaling between MsTeams and > > Kamailio. I currently have this configuration in my tls.cfg file > > > > > > > > > > > > [server: default] > > > > > > method = TLSv1.2 + > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = /etc/kamailio/certificates/private-key.pem > > > > > > certificate = /etc/kamailio/certificates/certificate.pem > > > > > > > > > > > > [client: default] > > > > > > method = TLSv1.2 + > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = /etc/kamailio/certificates/private-key.pem > > > > > > certificate = /etc/kamailio/certificates/certificate.pem > > > > > > > > > > > > My domain was certified with ssl through an authoritative certifier > > (GoDaddy), however I see these errors in the / var / log / messages of > the > > Kamailio server. > > > > > > > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_mod.c:389]: mod_init(): With ECDH-Support! > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_mod.c:392]: mod_init(): With Diffie Hellman > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_init.c:722]: tls_h_mod_init_f(): compiled with openssl version > > "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, > > compression: on > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_init.c:730]: tls_h_mod_init_f(): installed openssl library version > > "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, > zlib > > compression: on#012 compiler: gcc -I. -I.. -I../include -fPIC > -DOPENSSL_PIC > > -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H > -DKRB5_MIT > > -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 > > -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 > > -grecord-gcc-switches -m64 -mtune=generic -Wa,--noexecstack -DPURIFY > > -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 > > -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM > > -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM > > -DECP_NISTZ256_ASM > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: WARNING: tls > > [tls_init.c:787]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks > on > > low memory) workaround enabled (on low memory tls operations will fail > > preemptively) with free memory thresholds 13107200 and 6553600 bytes > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold1 has been > > changed to 13107200 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold2 has been > > changed to 6553600 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [main.c:2834]: main(): processes (at least): 25 - shm size: 67108864 - > pkg > > size: 4194304 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is > initially > > 212992 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally > > 425984 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs: tls_method=22 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs: > > certificate='/etc/kamailio/certificados/certificate.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs: ca_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs: crl='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:335]: ksr_tls_fill_missing(): TLSs: require_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs: cipher_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs: > > private_key='/etc/kamailio/certificados/private-key.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:353]: ksr_tls_fill_missing(): TLSs: verify_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs: verify_depth=9 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs: verify_client=0 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: NOTICE: tls > > [tls_domain.c:1107]: ksr_tls_fix_domain(): registered server_name > callback > > handler for socket [:0], server_name='' ... > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:697]: set_verification(): TLSs: Client MUST present valid > > certificate > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc: tls_method=22 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc: > > certificate='/etc/kamailio/certificados/certificate.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc: ca_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc: crl='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:335]: ksr_tls_fill_missing(): TLSc: require_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc: cipher_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc: > > private_key='/etc/kamailio/certificados/private-key.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:353]: ksr_tls_fill_missing(): TLSc: verify_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc: verify_depth=9 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc: verify_client=0 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:697]: set_verification(): TLSc: Server MUST present valid > > certificate > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32422]: INFO: > jsonrpcs > > [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/32422 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32424]: INFO: ctl > > [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io > > watch method (config) > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_server.c:1283]: tls_h_read_f(): protocol level error > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.75.24 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: > > [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error > reading > > - c: 0x7f45242be028 r: 0x7f45242be150 (-1) > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_server.c:1283]: tls_h_read_f(): protocol level error > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.132.46 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: > > [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error > reading > > - c: 0x7f45242d9278 r: 0x7f45242d93a0 (-1) > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_server.c:1283]: tls_h_read_f(): protocol level error > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.14.70 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: > > [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error > reading > > - c: 0x7f45242be028 r: 0x7f45242be150 (-1) > > > > > > > > > > > > Could you help me identify the problem please. > > > > > > > > > > > > Cheers > > > > > > > > > > > > Saludos Cordiales > > > > > > -- > > > > > > Willy Valles Rios > > > > > > Unified Communications Specialist > > > > > > > > > > > > phone: +51955747343 > > > > > > em@il: willyvalles17(a)gmail.com > > > > > > > > > > > > _______________________________________________ > > > > > > Kamailio (SER) - Users Mailing List > > > > > > sr-users(a)lists.kamailio.org > > > > > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > -- > > > > > > Daniel-Constantin Mierla -- www.asipto.com > > > > > > www.twitter.com/miconda -- www.linkedin.com/in/miconda > > > > > > Funding: https://www.paypal.me/dcmierla > > > > > > _______________________________________________ > > > Kamailio (SER) - Users Mailing List > > > sr-users(a)lists.kamailio.org > > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > > VoIP Embedded, Inc. > > http://www.voipembedded.com > > >

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by Dmitry Sytchev 22 Jan '21

22 Jan '21

Hello! In 5.4 release announce there is a line about contact rewriting. "new Contact address encoding/decoding functions – replace/restore Contact address with SIP server address, by encoding/decoding old address in a parameter" Where can I find some documentation or examples about NEW functions? At present we handle this task exactly the same way via parameters manually. It is kinda verbose and hard to understand. Maybe new functions will simplify this. Please advise! Thanks! -- Best regards, Dmitry Sytchev, IT Engineer

2 3

Help required in sipwise rtpengine-ng-client and rtpengine-ctl
by mahesh prasad behera 22 Jan '21

22 Jan '21

Hi Team, We are using sipwise rtpengine on platform centos 7. To get call related statistics from rtpengine. We tried to use utils "rtpengine-ng-client" and "rtpengine-ctl". But for us both of them are not working. *Rtpengine process status:* [root@ctl utils]# ps -ef |grep rtpengine root 3924 668 0 00:11 pts/0 00:00:00 grep --color=auto rtpengine root 20502 20499 0 Jan19 ? 00:10:02 ../sbin/rtpengine -f --num-threads 4 -i pub/10.211.160.132 -i priv/10.211.160.132 -n 127.0.0.1:8500 -c 127.0.0.1:8500 -m 32001 -M 32500 -T 184 -o 90 -d 4 -s 900 -p /var/run/rtpengine1.pid --scheduling rr --priority 37 [root@ctl utils]# * rtpengine-ng-client:* When i ran rtpengine-ng-client, I was getting Bencode.pm missing so I have manually installed perl bencode library "perl-Convert-Bencode-1.03-9.el7.noarch.rpm" Now when i ran rtpengine-ng-client i am getting below error [root@ctl utils]# ./rtpengine-ng-client list *Undefined subroutine &Bencode::bencode called at /usr/local/lib64/perl5/NGCP/Rtpengine.pm line 33. * *rtpengine-ctl:* When i ran ./rtpengine-ctl -ip 127.0.0.1:8500 list , We are not getting any valid response from rtpengine. [root@ctl utils]# ./rtpengine-ctl -ip 127.0.0.1:8500 list *Inside do while after call socket->recv(response, 1024*1024*10)* Please find the below attached source code for rtpengine-ng-client and rtpengine-ctl. Any help and suggestion will be highly appreciated. Thanks, Mahesh

2 1

Kemi execute a route from configuration file
by David VILLAUME 22 Jan '21

22 Jan '21

Hello, I was pretty sure there was a way to call a route from config file from Kemi, but I can’t find documentation about it. Does that really exists or did I dreamed it ? Regards, David

2 2

Kamailio, debian and MySQL caching_sha2_password
by Nicki Otte 22 Jan '21

22 Jan '21

Hi all Have any of you experienced problems using MySQL(Not MariaDB) on a debian machine, with caching_sha2_password? I am having some trouble: Jan 21 08:13:41 MobileValue-Kamailio systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Jan 21 08:13:42 MobileValue-Kamailio kamailio: INFO: <core> [core/tcp_main.c:4983]: init_tcp(): using epoll_lt as the io watch method (auto detected) Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: Listening on Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: udp: 185.31.176.39:5060 Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: tcp: 185.31.176.39:80 Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: Aliases: Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: tcp: voipgw01.evercall.dk:80 Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: udp: voipgw01.evercall.dk:5060 Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Plugin caching_sha2_password could not be loaded: /usr/lib/x86_64-linux-gnu/mariadb19/plugin/caching_sha2_password.so: cannot open shared object file: No such file or directory Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: dispatcher [dispatch.c:903]: ds_connect_db(): cannot initialize db connection Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: dispatcher [dispatch.c:937]: ds_init_db(): unable to connect to the database Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: dispatcher [dispatcher.c:404]: mod_init(): could not initiate a connect to the database Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module dispatcher (/usr/lib/x86_64-linux-gnu/kamailio/modules/dispatcher.so) Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jan 21 08:13:42 MobileValue-Kamailio systemd[1]: kamailio.service: Control process exited, code=exited, status=255/EXCEPTION Jan 21 08:13:42 MobileValue-Kamailio systemd[1]: kamailio.service: Failed with result 'exit-code'. Jan 21 08:13:42 MobileValue-Kamailio systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. I can get around the problem, by making the kamailio & kamailioro user use mysql_native_password instead. But am i wrong to belieave that it should Work out of the box? Here are the installed packages from APT: kamailio/unknown,now 5.4.3+bpo10 amd64 [installed] kamailio-json-modules/unknown,now 5.4.3+bpo10 amd64 [installed] kamailio-mysql-modules/unknown,now 5.4.3+bpo10 amd64 [installed] kamailio-outbound-modules/unknown,now 5.4.3+bpo10 amd64 [installed] default-libmysqlclient-dev/stable,now 1.0.5 amd64 [installed] default-libmysqld-dev/stable,now 1.0.5 amd64 [installed] libmariadb-dev/stable,now 1:10.3.27-0+deb10u1 amd64 [installed,automatic] libmariadb-dev-compat/stable,now 1:10.3.27-0+deb10u1 amd64 [installed,automatic] mysql-apt-config/unknown,now 0.8.16-1 all [installed] mysql-client/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-common/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-client/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-client-core/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-client-plugins/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-server/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-server-core/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-server/unknown,now 8.0.23-1debian10 amd64 [installed] Thanks in advance Nicki Bo Otte

4 8

RTP Querys
by Ahmed Marsou 22 Jan '21

22 Jan '21

Hi guys; I have some rtpproxy questions: - When a proxy is in a failed state, what happens to the active sessions / streams? - How many times the module should retry to send and receive after timeout was generated? - *How often are the control messages sent to the rtp proxy’s in the pool?* *- I would like to know the exact nc message that’s used for the control messages* *Thank you.*

2 2

tracking usrloc clean
by Michal Popovic 22 Jan '21

22 Jan '21

Hi All, I would like to ask you if there is any way how to track usrloc location cleaning operation with some event route, so I can broadcast the change with for example kafka message. If no, how do you keep track of online/offlice status of the user? I am trying to avoid using change data capture in mysql proxy for location table. Michal

2 2

Kamailio and Serverless
by H Yavari 22 Jan '21

22 Jan '21

Hi to all, Does anyone have experience using Kamailio in front of some Lambda / Cloud functions? For example, passing call info with JSON to functions.Please share your ideas. Regards,HYavari

4 4

Kamailio v5.4 install
by Jewna 21 Jan '21

21 Jan '21

I'm having a problem installing kamailio 5.4 using the following guide https://kamailio.org/docs/tutorials/5.4.x/kamailio-install-guide-git/ . I have a debian 10.7 machine where I install default-mysql-server package for the database. After that I follow the install guide and everything goes smoothly until I have to install the init.d scripts. 1. I do "cd /usr/local/src/kamailio-5.4/kamailio' 2. I do "make install-initd-debian" Then make fails with the following error " root@kamServ:/usr/local/src/kamailio-5.4/kamailio# make install-initd-debian make -C src/ install-initd-debian make[1]: Entering directory '/usr/local/src/kamailio-5.4/kamailio/src' sed -e "s#DAEMON=/usr/sbin/kamailio#DAEMON=/usr/local/sbin/kamailio#g" \ -e "s#NAME=kamailio#NAME=kamailio#g" \ -e "s#DESC=Kamailio#DESC=kamailio#g" \ -e "s#HOMEDIR=/run/kamailio#HOMEDIR=/run/kamailio#g" \ -e "s#DEFAULTS=/etc/default/kamailio#DEFAULTS=/etc/default/kamailio#g" \ -e "s#CFGFILE=/etc/kamailio/kamailio.cfg#CFGFILE=/usr/local/etc/kamailio/kamailio.cfg#g" \ < pkg/kamailio/deb/debian/kamailio.init \ > /etc/init.d/kamailio /bin/sh: 1: cannot open pkg/kamailio/deb/debian/kamailio.init: No such file make[1]: *** [Makefile:982: install-initd-debian] Error 2 make[1]: Leaving directory '/usr/local/src/kamailio-5.4/kamailio/src' make: *** [Makefile:34: install-initd-debian] Error 2 " Am I doing something wrong? Possibly in the wrong directory? If you need any more information please don't hesitate to ask. Sincerely, Matthew Sherwood

2 1

[kamailio 5.4.3] xavp_rm_by_index removes (N+1) first indexes instead of index N
by Chaigneau, Nicolas 20 Jan '21

20 Jan '21

Hello I’m trying to use function xavp_rm_by_index to remove a specific index from a given xavp. I’m observing an unexpected behaviour, I think it might be a bug : Instead of just removing the index, the function removes the index and all others before it. E.g : if called with « idx = 1 », it removes indexes 0 and 1. Reading the code of xavp_rm_internal (which is called by xavp_rm_by_index), I don’t think it behaves at it should : /* Remove xavps * idx: <0 remove all xavps with the same name * >=0 remove only the specified index xavp * Returns number of xavps that were deleted */ For example when I do this : LM_ERR("COUNT BEFORE: %d\n", xavp_count(rname, NULL)); int rm_count = xavp_rm_by_index(rname, 1, NULL); // TEST LM_ERR("REMOVED: %d\n", rm_count); LM_ERR("COUNT AFTER: %d\n", xavp_count(rname, NULL)); « rm_count » returned is 1 (which is expected). However, « count after » is decremented by 2. And when I look at the contents of my xavp, I notice that the first two elements have been removed. Likewise if xavp_rm_by_index is invoked with idx = 2 => the first 3 elements are removed. So… do you think this is a bug ? or something I didn’t understand ? Thanks for your help ! Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 2

Dialog Tracking with Inconsistent RR
by Charles Phillips 20 Jan '21

20 Jan '21

I am working to track concurrent channels with the dialog module, but I have run in to an issue. Due to the nature of my required configuration, I have a need to strip the record route headers from packets that are sent to the upstream proxy. This leaves no way for the dialog module to track the state of the call. I was hoping that DID_FALLBACK or DID_NONE modes would track by call id or some other constant, but it does not appear to. So, before I start down the path of using htable to store and reinsert the did when a packet returns, is there something I am missing with dialog? Is there some more efficient way to do this? - Charles

2 2

Error: invalid json doc in command "kamctl ping" kamailio 5.4.3
by Julia Bo 20 Jan '21

20 Jan '21

Hello, Kamailio 5.4.3 has been installed from source. "kamctl ping sip:10.10.10.10:5060" is sent from the command line. Errors: Jan 12 15:59:24 KAM-TEST /usr/local/kam/kamailio[66587]: ERROR: jsonrpcs [jsonrpcs_mod.c:1352]: jsonrpc_exec_ex(): invalid json doc [[{"jsonrpc": "2.0", "method": "tm.t_uac", "], "reply_name": "kamailio_receiver_85791", "id": 85791}]].", "From:sip:daemon@KAM-TEST Jan 12 15:59:24 KAM-TEST /usr/local/kam/kamailio[66587]: ERROR: jsonrpcs [jsonrpcs_fifo.c:461]: jsonrpc_fifo_server(): failed to execute the json document from fifo stream This command is working in 4.4. Has anyone tried using kamctl ping in 5.4? Is it a bug in 5.4.3 ? Thank you, Julia.

3 10

Header Manipulation with returned REST data
by Mark Lawrence 19 Jan '21

19 Jan '21

Hi All, Sorry for the intrusion and asking directly, but there’s so much information out there that I would spend years looking for the answer myself and spending equal amount of time in a lab playing. I do like the last idea but time is a little short. I’m thinking of using Kamailio to overcome a few downfalls in the main SBC/PBX commercial product space that I’ve typically deployed and wanted to ask if it is possible to use as a stateful proxy and carry out the following header manipulation type scenario with a bit more dynamassium: Use the FROM and TO header to create two value pairs that would be sent via a REST request to an external application, which will return two corresponding values to (a) supply a different value to update the original FROM header and (b) a full PAI sting to add on to the outgoing message. If you have experience of this or if it’s a nonstarter I’d love to know. There’s so many modules and great stuff that you have in your toy chest it is amazing and kudos to all you developers being so inventive out there: it leaves me to think I’ve been deploying SIP wrong all these years. Thanks and keep safe and well, Mark

2 1

Count the rows in a xavp from script
by Chaigneau, Nicolas 19 Jan '21

19 Jan '21

Hello, I'm looking for a way to retrieve the number of rows stored in a xavp. (Something better than manually iterating over the xavp) Basically, I'd like to call xavp_count from the script. Is that possible ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 1

Kamailio High Availability questions
by Andriano 17 Jan '21

17 Jan '21

Hello have some questions about Kamailio HA + loadbalancers (dispatcher modules) 1. is posible to make HA for RTP media? maybe RTPEngine +redis? is not possible to do this with RTProxy? where is some examples? 2. What do Kamailio db cluster module? it can replicate every DB query record like normal DB cluster? or better is to use native MySQL cluster mode? 3. How is working dispatcher modules in HA? how call is handled and htable in memory between loadbalancers endpoint nodes etc... IS it posible to do only with DMQ ? Just need understand how to do basic Kamailio setup with HA and load balancers in general, for example if will be 2 phisical servers, in every server is virtualisation and every server has 2 VPS. 1 VPS for frontend Kamailio loadbanancer, 1 VPS backend Kamailio app with accounts, routing logic, LCR, RTP media app etc... between load balancers i know maybe is possible to do floating ip, dns round robinetc... and if one VPs server down, call will be dropet in every situation? or call will not be droped and how to prevent it. Just need to know in general schema and architecture how to do this real HA in Kamailio and avoind sinlge point of failure and where to read and learn it ho to make HA in real. In the internet no many docs about it, many conference video how good it is, but no good examples with config, i did not founded, so will be very glad for any help in these newbie questions ;) with single kamailio is many answers, but is realy difficult to understand how to do voip HA with Kamailio without single point of failure. Thanks for any answers...

1 0

http_client module not found
by Joli Martinez 15 Jan '21

15 Jan '21

Hello, I am running kamailio 5.3.8 on Debian 9. Kamailio is installed via apt package manager. I am trying to get apiban.org to work and it seems to be having a problem with the http_client module. I have searched for the kamailio modules and I am unable to find anything that resembles http. I have also installed the extra module and no luck. What would be the correct module to install? kamailio: ERROR: <core> [core/sr_module.c:500]: load_module(): could not find module <http_client> in </usr/lib/x86_64-linux-gnu/kamailio/modules/>

2 2

Stubs for KEMI Javascript
by Nicki Otte 15 Jan '21

15 Jan '21

Hi everyone. I wanted all of you to know, that I have made JavaScript stubs, for autocompletion in IDE’s, for those of you that use KEMI JavaScript. Specifically tested on Jetbrains PHPStorm/Webstorm. The stubs are available for free at my GitHub page. The link is https://github.com/captainDuckay/Kamailio-javascript-stubs My main language isn't typescript or JavaScript, and this is one of my first public releases. With that said though, I am open for critics, or even guidance. I hope it's usable for you. // Nicki Bo Otte

2 3

To header being malformatted.
by AL S 14 Jan '21

14 Jan '21

Hi, We noticed an issue with the way the Kamailio is handling certain messages being sent to them, specifically, the To header being relayed from Kamailio to the next hop is malformatted. Any assistance is greatly appreciated INVITE received by Kamailio: Request: Invite Request URI: sip:2053140346;isub=33343@72.31.12.137:5060;transport=udp From: <sip:+13037819601@71.44.15.84;user=phone>;tag=gK00a85927 To: <sip:+12053140346@72.31.12.137>;tag=SDnpoa601-1c711123926 Call-ID: SDnpoa601-f1bc387aa5a486c1dd943d1b53ae8e6a-8q7og02 CSeq: 860668 INVITE Contact: <sip:+13037819601@107.14.122.29:5060> SDP IP: 0.0.0.0 SDP Port: 43126 INVITE forwarded by Kamailio: Request: Invite Request URI: sip:2053140346;isub=33343@72.31.12.137:5060;transport=udp From: <sip:+13037819601@71.44.15.84;user=phone>;tag=gK00a85927 To: <sip:+120531403Oh.lc".3/+0]1..31.12.137>;tag=SDnpoa601-1c711123926 Call-ID: SDnpoa601-f1bc387aa5a486c1dd943d1b53ae8e6a-8q7og02 CSeq: 860668 INVITE Contact: <sip:+13037819601@107.14.122.29:5060> SDP IP: 0.0.0.0 SDP Port: 43126 Another example in BYE: BYE received by Kamailio: To: "WIRELESS CALLER" <sip:+18131111111@71.44.13.84>;tag=gK0a788ca6 BYE forwarded by Kamailio: To: "WIRELESS CALLER" <sip:+18131112222A.++.+,-94 Thanks. AL

2 2

Source Port on TLS OPTIONS from Dispatcher
by Charles Phillips 14 Jan '21

14 Jan '21

Hello all. As they say in radio, “long time listener, first time caller” Anyway, I am having trouble getting past the following road block and any help would be greatly appreciated. Kamailio version is 5.4.3 When attempting to use dispatcher to send OPTIONS packets to several TLS destinations, the packets are leaving the Kamailio server on random ports. This is a problem because the servers I am sending the OPTIONS to (MS Teams) are enforcing rport so the responses are returned to a port that Kamailio is not listening on. I have tried to force the socket in the event route (relevant parts of snippet below) but it does not appear to help. I should also mention that I am not behind NAT and the TLS socket is specified in the dispatcher attrs. event_route[tm:local-request] { sip_trace(); $fs = “tls:**ip-address**:5061”; } I have used Kamailio as a TLS server for many projects, but this is my first time as a client. I am sure I am missing something. - Charles

4 10

BYE and TCP
by Kjeld Flarup 13 Jan '21

13 Jan '21

Hello I have a Kamailio running behind NAT, which sends calls to a VOIP service provider. I have setup the Kamalio to listen on port 5071, and also setup a port forward in the router. Now the problem is that with TCP, 5071 is not used for the dialog, but a new port is chosen everytime. This means that when the mobile phone called hands up, I never sees the BYE, because BYE is a new dialog. To which port is the server supposed to send the BYE, and what field tells the server this. -- -------------------- 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

5 17

handling rejected REFER messages
by Jaime Gajardo 13 Jan '21

13 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environment. behind the nat we have 2 identical nodes using REFER for transfer call to another destiny. On the other NAt side we have our public carrier. When some node sends to kamailio the REFER message and kamailio reroutes this message to the carrier. The call is rejected due company policies. We want to know how we can convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux

2 1

handling rejected REFER messages
by Jaime Gajardo 12 Jan '21

12 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environment. behind the nat we have 2 identical nodes using REFER for transfer call to another destiny. On the other NAt side we have our public carrier. When some node sends to kamailio the REFER message and kamailio reroutes this message to the carrier. The call is rejected due company policies. We want to know how we can convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux rtpengine -v: Version: 9.2.0.0+0~mr9.2.0.0 git-master-aa5e0244

1 0

handling rejected REFER messages
by Jaime Gajardo 12 Jan '21

12 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environmet. behind the nat we have 2 identycal nodes using REFER for transfer call to another destiny. in the other NAt side we have our public carrier. When some node send to kamailio the REFER message and kamailio reroute this message to the carrier. The call is rejected due company policies. We want know how can we do for convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux rtpengine -v: Version: 9.2.0.0+0~mr9.2.0.0 git-master-aa5e0244

1 0

Re: [SR-Users] Using websockets with kamailio 5.4
by Jeremy McNamara 11 Jan '21

11 Jan '21

Hi Daniel - Google is your friend. I use it often. https://github.com/havfo/WEBRTC-to-SIP

1 0

TSILO - transaction is not suspended
by Jeremy McNamara 11 Jan '21

11 Jan '21

Hi Folks - I am attempting to set up a mobile push configuration, by following the published example(s). If we have more than one endpoint registered, this configuration works as expected (via the ts_store() path below). If we do not have another registered endpoint the transaction seems to be suspended but then cannot be resumed (t_suspend) after REGISTER arrives. INFO: <script>: [R-LOG]: INVITE from IP:[24.XXX.XXX.XXX:62665] [sip:jeremy2@stage][<null>] NFO: <script>: [R-SUSPEND]: Suspended transaction [41359:415220139] INFO: <script>: [R-SUSPEND]: fU:[jeremy] rU:[jeremy2] tU:[jeremy2] INFO: <script>: [R-SUSPEND]: fU:[jeremy] => tU:[jeremy2] INFO: <script>: [R-SUSPEND]: vtp:[41359:415220139] INFO: <script>: [R-SENDPUSH]: Push sent [(55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78][com][hq01gt5utegi18ipkh1b] [Node-Push]: About to push: 55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78 INFO: <script>: R-REGISTER]: jeremy2 registered INFO: <script>: [R-REGISTER]: Contact: [<sip:ilyartjr@qjwhdafxlbtk.invalid ;transport=ws>;expires=10;app-id=stage;pn-voip-tok=55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78] INFO: <script>: [R-REGISTER]: App id: stage INFO: <script>: [R-REGISTER]: Expires: 10 INFO: <script>: [R-JOIN]: jeremy2 INFO: <script>: [R-JOIN]: Resuming trasaction [41359:415220139] jeremy2 (41359:415220139) WARNING: tm [t_suspend.c:197]: t_continue_helper(): transaction is not suspended [41359:415220139] WARNING: tmx [tmx_mod.c:847]: w_t_continue(): resuming the processing of transaction [41359:415220139] failed Relvant kamailio.cfg [Note: I did notice we had to switch things up a bit if calls come from a PSTN number via SBC] modparam("htable", "htable", "vtp=>size=10;autoexpire=300") ... if (!lookup("location")) { send_reply("100", "Suspending"); route(SUSPEND); // *testing this path here * } else { route(RELAY); ts_store("$tu"); $sht(vtp=>stored::$tU) = 1; xlog("L_INFO", "[R-LOCATION]: Stored transaction [$T(id_index):$T(id_label)] $fU => $tU\n"); } ... route[SUSPEND] { if (!t_suspend()) { xlog("L_ERROR", "[R-SUSPEND]: Failed to suspend transaction [$T(id_index):$T(id_label)]\n"); send_reply("501", "Unknown Destinaton"); exit; } xlog("L_INFO", "[R-SUSPEND]: Suspended transaction [$T(id_index):$T(id_label)]\n"); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] rU:[$rU] tU:[$tU]\n"); if (isbflagset(FLB_PSTN)) { $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] => rU:[$rU]\n"); xlog("L_INFO", "[R-SUSPEND]: vtp:[$sht(vtp=>join::$rU)]\n"); } else { $sht(vtp=>join::$tU) = "" + $T(id_index) + ":" + $T(id_label); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] => tU:[$tU]\n"); xlog("L_INFO", "[R-SUSPEND]: vtp:[$sht(vtp=>join::$tU)]\n"); } } route[REGISTER] { if (!is_method("REGISTER")) { return; } if (isflagset(FLT_NATS)) { setbflag(FLB_NATB); } if (!save("location")) { sl_reply_error(); } route(JOIN); } # Append branches or resume the transaction route[JOIN] { xlog("L_INFO","[R-JOIN]: $tU\n"); $var(hjoin) = 0; lock("$tU"); $var(hjoin) = $sht(vtp=>join::$tU); $var(hstored) = $sht(vtp=>stored::$tU); $sht(vtp=>join::$tU) = $null; unlock("$tU"); if ($var(hjoin)==0) { if ($var(hstored)) { xlog("L_INFO","[R-JOIN]: Appending transaction ($tu)\n"); ts_append("location", "$tu"); exit; } xlog("L_INFO","[R-JOIN]: No INVITE to resume\n"); exit; } $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); xlog("L_INFO", "[R-JOIN]: Resuming trasaction [$var(id_index):$var(id_label)] $tU ($var(hjoin))\n"); t_continue("$var(id_index)", "$var(id_label)", "RESUME"); exit; } # Lookup and relay after resuming transaction route[RESUME] { xlog("L_INFO", "[R-RESUME]: ru: [$ru] tu: [$tu]\n"); lookup("location"); route(SETUP_ROUTES); route(RELAY); ts_store("$tu"); $sht(vtp=>stored::$tU) = 1; xlog("L_INFO", "[R-RESUME]: Stored transaction [$T(id_index):$T(id_label)] $fU => $tU\n"); exit; } What did I miss?

4 6

Profile name as variable in set_dlg_profile
by Muhammad Sohaib 11 Jan '21

11 Jan '21

Is it possible to insert the current dialog into the profile by using the following way? modparam("dialog", "profiles_no_value", "mytest;xyz") $var(profilename)="xyz"; set_dlg_profile(" $var(profilename) "); Currently it is giving errors and not allowing to use variable instead of static profile name. Or is there any alternative to pass profile name as variable? --- Thanks,

4 6

Unload UAC remote registry
by Javier Valencia 11 Jan '21

11 Jan '21

Hello, I'm on Kamailio 4.4.7 and I'm trying to do uac.reg_refresh over a row non existing on DB to try to unload, but data still persist on Kamailio. How to unload only one registry from UACs remote registrations when delete a record from DB without do a uac.reg_reload? UAC remote registrations is above 2k rows, I don't know if it's much or not. Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

3 2

Different responses for domain.dump
by Javier Valencia 11 Jan '21

11 Jan '21

Hi there! I'm getting different responses from Kamailio 4.4.7 when domain.dump exec; From *kamcmd*: I'm getting two domains; .es and .net. Request to jsonrpc-s module: Response from jsonrpc-s module: And now, I'm getting only one domain. What's I'm doing wrong?

2 2

uac.reg_refresh command
by Javier Valencia 11 Jan '21

11 Jan '21

Hello, I'm playing with RPC on my Kamailio 4.4.7, and I getting rare behaivor with *uac.reg_refresh* command. When I run it thru kamcmd like docs describes, this is without *l_uuid* parameter, I'm geting "*error: 500 - Invalid Parameters*" output. And when I run with *l_uuid* parameter, I get no output. But in both cases, Kamalio don't refresh data from DB. But I run this command thru HTTP (JSONRPC-S), both call modes are equivalent. With or without l_uuid parameter geting no output and refresh data from DB. It's a normal behaivor? Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

2 3

list myself values
by Javier Valencia 11 Jan '21

11 Jan '21

Hi, a question from a Kamailio newbie. How can I list or show variable *myself* values? Regards, JV

2 2

UAC module registrations interval
by Javier Valencia 11 Jan '21

11 Jan '21

Hi there, I'm using Kamailio 4.4.7 with UAC remote registrations module. In my database have all rows with expires column value's 3600, and my PBX (asterisk) min, max and default expires to 3600. But I can see how Kamailio it's sending REGISTER each 150segs. UAC module parameters values; modparam("uac", "reg_db_url", "mysql://XXX") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_contact_addr", "XXX") modparam("uac", "reg_timer_interval", 5) modparam("uac", "reg_retry_interval", 60) My uacreg table it's above 2k rows. Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

2 1

how does rtjson work?
by Syed Shahryar 11 Jan '21

11 Jan '21

My use case is such that I want to be able to send a SIP MESSAGE destined for a single username to multiple users. In order to be able to map that target username to multiple usernames, I tried using rtjson and evapi/http (to generate the routing json), but none of them worked. Is it the right approach? Is there any other way to programmatically generate multiple targets for SIP request? I am familiar with dynamic routing but I need more flexibility. Initially, I tried sending the request to a different username but that didn't work and got the following message in the http async result route. kamailio | 19(25) ERROR: nathelper [nathelper.c:1461]: sdp_1918(): Unable to parse sdp body kamailio | 19(25) NOTICE: <core> [core/kemi.c:124]: sr_kemi_core_notice(): In evapi msg branch kamailio | 28(34) NOTICE: {2 6661 MESSAGE q6hikd6tumsr1dh35kvj} <core> [core/kemi.c:124]: sr_kemi_core_notice(): In evapi reply branch The message was received on the originally requested to user and also $tu still points to the original user instead of the new user even after executing the following code. KSR.rtjson.init_routes(response); KSR.rtjson.push_routes(); KSR.nathelper.fix_nated_contact(); In http response, I receive the following and don't see any rtjson errors when init_routes is executed on it. {"version":"1.0","routing":"parallel","routes":[{"headers":{"from":{"display":"test_1","uri":"sip: *test_1*@test.com:9908"},"to":{"display":"test_4","uri":"sip:*test_4*@ test.com:9908 "},"extra":""},"branch_flags":8,"fr_timer":5000,"fr_inv_timer":30000}]} Please note that the request was sent to *test_2*(a)test.com. I have tried other combinations of above but nothing works. I have already searched on Google and read relevant articles and blog posts. Thanks -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Lost MySQL connections - Error 500
by Duarte Rocha 11 Jan '21

11 Jan '21

Hi all, I've made a post last month regarding losing MySQL connections - https://lists.kamailio.org/pipermail/sr-users/2020-December/111389.html At the time I thought connections were dying as a consequence of low activity and traffic on the proxy. Meanwhile, I've migrated a great number of equipments to the proxy with Registers being refreshed every 10minutes and the problem still persists. In order to try to fix this i've added timeout_interval and ping_interval from the db_mysql module. My SQL client on the Kamailio machine is mysql-community-client 5.6.50-2.el7. It writes and reads in a remote InnoDB database. This are the logs i get from Kamailio when the problem appears : Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lock wait timeout exceeded; try restarting transaction (1205) Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} <core> [db_query.c:348]: db_do_update(): error while submitting query Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} usrloc [ucontact.c:1147]: db_update_ucontact_ruid(): updating database failed Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} usrloc [ucontact.c:1663]: update_contact_db(): failed to update database Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} registrar [save.c:784]: update_contacts(): failed to update contact Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) Originally I had usrloc db_mode on mode 3 - DB-Only Scheme. In order to try to mitigate the issue I changed it to mode 1 - Write-Through scheme but even then I get the same log errors and an "500" error is still sent to the client. I chose this mode since, as far as I can understand it applies changes directly to DB but also uses cache. Please correct me if i'm wrong on that. Has this issue happened with anyone before? Is there a way to mitigate this issue? My only constraint is that I need the database to be always updated since I have an HA setup, and as such, I can't use cache only methods. Best Regards,

2 5

loose_route() - Local Proxy
by Duarte Rocha 08 Jan '21

08 Jan '21

Greetings, I'm using loose_route() from the RR module and i'm having troubles making it use the following exception from the code : "There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE." My example is a REGISTER without To-TAG which has a Route header with kamailio address. If i use proxy IP on Route header , loose_route() returns false as it should. However, if i use an hostname belonging to the proxy in the route, loose_route() returns true. I have hostnames and local ips defined in the "DOMAIN" table but it doesn't seem to be working. Which other places can local hostnames and ips be configured in order to be seen as local to loose_route() ? Best Regards

4 4

Contact header not removed properly Getting some junks
by Devang Dhandhalya 07 Jan '21

07 Jan '21

in my scenario call coming from sbc to kamailio proxy and we route to pbx server in that when kamailio proxy get the call at that time i am removing Contact header using remove_hf("Contact") and adding contact Header using append_hf("$uac_req(hdrs)") for route to PBX server . in that signaling i am getting modified Contact header but old contact header not removed properly some junks are remain in the signaling scenario : SBC ---------> Kamailio proxy ------> PBX before using remove_hf("Contact") INVITE sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 70 Contact: <sip:user1@10.212.xxx.xxx:39931;transport=UDP> To: <sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP> From: <sip:user1@ <sip%3A1234(a)opensips.org>domain.org ;transport=UDP>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp Proxy-Authorization: Digest username="user1",realm="domain.org <http://opensips.org/> ",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2@domain.org <sip%3A5678(a)opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5 82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5 User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 243 v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 10.212.xxx.xxx t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv after using remove-hf("Contact") function removing contact header but some junks are left and that eats next To header line and breaks Invite packet INVITE sip:user2@domain.org:;transport=UDP SIP/2.0 Record-Route: <sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA-> Via: SIP/2.0/UDP 172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 69 sip:user1@172.xx.xx.x:39931;transport=UDPTo: <sip:user2@domain.org <sip%3Adevang3032(a)opensips.org>> From: <sip:user1@domain.org <sip%3Adppatel(a)opensips.org>>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 534 Contact: <sip:user2@172.xx.xx.xxx:5060 <http://sip:dppatel@172.16.16.163:5060/>> v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 172.xx.xx.xxx t=0 0 m=audio 11210 RTP/AVP 3 110 8 0 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:97 mode=30 a=fmtp:101 0-16 a=sendrecv a=rtcp:11211 a=ice-ufrag:lzjregPe a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.xx.xx.xxx 11210 typ host a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.xx.xx.xxx 11211 typ host As you can see Contact header is gone but sip:1234@172.xx.xx.x:39931;transport=UDP part left before To header . and its breake invite packet . Any suggestion will be highly appreciated. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 1

about Contact Header Junks
by Devang Dhandhalya 07 Jan '21

07 Jan '21

in my requirement call coming from sbc to kamailio proxy and we route to pbx server in that when kamailio proxy get the call at that time i am removing Contact header using remove_hf("Contact") and adding contact Header using append_hf("$uac_req(hdrs)") for route to PBX server . in that signaling i am getting modified Contact header but old contact header not removed properly some junks are remain in the signaling scenario : SBC ---------> Kamailio proxy ------> PBX before using remove_hf("Contact") INVITE sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 70 Contact: <sip:user1@10.212.xxx.xxx:39931;transport=UDP> To: <sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP> From: <sip:user1@ <sip%3A1234(a)opensips.org>domain.org ;transport=UDP>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp Proxy-Authorization: Digest username="user1",realm="domain.org <http://opensips.org> ",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2@domain.org <sip%3A5678(a)opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5 82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5 User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 243 v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 10.212.xxx.xxx t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv after using remove-hf("Contact") function removing contact header but some junks are left and that eats next To header line and breaks Invite packet INVITE sip:user2@domain.org:;transport=UDP SIP/2.0 Record-Route: <sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA-> Via: SIP/2.0/UDP 172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 69 sip:user1@172.xx.xx.x:39931;transport=UDPTo: <sip:user2@domain.org <sip%3Adevang3032(a)opensips.org>> From: <sip:user1@domain.org <sip%3Adppatel(a)opensips.org>>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 534 Contact: <sip:user2@172.xx.xx.xxx:5060 <http://sip:dppatel@172.16.16.163:5060>> v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 172.xx.xx.xxx t=0 0 m=audio 11210 RTP/AVP 3 110 8 0 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:97 mode=30 a=fmtp:101 0-16 a=sendrecv a=rtcp:11211 a=ice-ufrag:lzjregPe a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.16.16.163 11210 typ host a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.16.16.163 11211 typ host As you can see Contact header is gone but sip:1234@172.xx.xx.x:39931;transport=UDP part left before To header . and its breake invite packet . Any suggestion will be highly appreciated. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

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Dispatcher SNI Questions
by Tim Chubb 06 Jan '21

06 Jan '21

Hi All Is it possible to explicitly define the tls cert to be used per destination by dispatcher? I'm attempting to integrate with a service that requires domain name presented to match that of the cert. Where the use of dispatcher is probably not as intended anyway as it targets the same set of destinations multiple times (primary and secondary endpoints for the service im integrating with), with multiple distinct set id's (basically a set per customer for multi tenancy) which expects a sip ping per account to verify that connection is alive and healthy. #Example Dispatcher list #Account 1 1 sip:a.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account1.com;ping_from=sip:sip.account1.com:5061 1 sip:b.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account1.com;ping_from=sip:sip.account1.com:5061; #Account 2 2 sip:a.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account2.com;ping_from=sip:sip.account2.com:5061; 2 sip:b.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account2.com;ping_from=sip:sip.account2.com:5061; Currently I'm running a solution based around certs with multiple SAN (subject alternative name) defined, but this is a pain to administrate, and not as scalable as I would like. I want to be able to define multiple client:any tls profiles and explicitly send via that profile. Its easy enough to do that in config using the tls xavps to define a server name and/or id. But for the sip options ping by dispatcher to work you need to specify the server name or id before the tm:local-request event route fire's. I've resorted to hacking away at dispatcher to see what would happen if I set the tls server name or id before the OPTIONS message is sent, and it works great for the first tls client profile matched, but any others you define re-use the initial tcp connection so reuses the first connections tls config, thus presenting the wrong cert (I'm probably wrong but logs show only one tls complete_init() related to dispatcher pings and the second set's pings skips all the tls init logging so suggests some thing is cached/reused, and logging on the service indicates cert name mismatch with set1s cert being offered) So I've hit a bit of a dead end, is there a way to force a new tcp connection per server name/id I fear outbound SNI might be a bit of a can of worms, for it to work nicely with dispatcher as is, would be an extension of the proto:host:port format to include a name or id, maybe something like: tls(name=sub.domain.com):10.0.0.5:5061 But that looks like it would have a lot of wide reaching knock-on effects, so setting it as an attribute like ping_from works fine as I'm currently doing, if I could force a new connection from dispatch.c.... The other thing that's occurred to me is that it might be better to stop trying to adapt a small percentage of dispatcher's functionality and trying to hammer a round peg into a square hole and instead try and create a new module similar to dispatcher in so much that it sends pings (that's all I really need in this case) and use dispatcher as intended with a single set of non-repeated destinations for routing. This would also allow much simpler integration with MS teams (if added ability to set contact address on the background pings) as well as the service im trying to integrate, and a more natural data set to be used as in my example you can see the only difference is the customers domain between each set. How I would imagine it to work would be to create a tls connection per customer domain, then iterate through the connections per destination in pseudo json the data would look like this: { Destinations:[ [1,["a.cloudservice.com","b.cloudservice.com"]], [2,["sip.pstnhub.microsoft.com", "sip2.pstnhub.microsoft.com", "sip3.pstnhub.microsoft.com"] ], Sources:[ {uri:"sip.account1.com",destinationMapping:[1]}, {uri:"sip.account2.com",destinationMapping:[1,2]} ] } And would just run in the background sending pings with SNI support, and if based on dispatcher should make it easy enough adapt things like the destination up and down event routes, and would allow me to use Kamailio much like I would use nginx for web stuff as a reverse proxy and handle TLS offload on my edge gateway proxy. Any thoughts?, im frustratingly close to having something that will do if I could get a new connection to be created, but I fear my use case might be sufficiently different from dispatchers to make a new module a sensible approach.. Regards Tim.

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Pre-loaded routes verification
by Duarte Rocha 04 Jan '21

04 Jan '21

Greetings, I'm doing a migration from a opensips system to kamailio one and i'm trying to replicate its functionalities. The system is a Registrar with a pstn gateway. I noticed that on the opensips version, loose_route() is called on initial requests and it seems to be used to protect against preloaded routes. However, on the default Kamailio configuration file loose_route() is only called on requests in-dialog and the verification mentioned above is not used. Why doesn't Kamailio includes this verification? Is there are any security concerns not using it? Best Regards,

2 1

using kamailio 5.4 with the websocket or xmlhttprequest
by Daniel Hermann N'don 04 Jan '21

04 Jan '21

Hello everyone, I'm a beginner in kamailio technology, and I would like to know how to communicate with kamailio from the web.

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Unable to connect sip with Kamailio
by Rohit Kumar 04 Jan '21

04 Jan '21

Hello everyone, I'm beginner in Kamailio, I want to connect my asterisk Server with Kamailio. Can you please help me in this context. I shall be very thankful to you. Regards, Rohit Kumar On Thu, Dec 31, 2020, 16:31 <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. using kamailio 5.4 with the websocket or xmlhttprequest > (Daniel Hermann N'don) > 2. Re: using kamailio 5.4 with the websocket or xmlhttprequest > (Antony Stone) > 3. Re: using kamailio 5.4 with the websocket or xmlhttprequest > (Daniel Hermann N'don) > 4. Using a specific log level for "script" ? (Chaigneau, Nicolas) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 30 Dec 2020 14:15:19 +0000 > From: Daniel Hermann N'don <danielhermann.ndon(a)outlook.com> > To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> > Subject: [SR-Users] using kamailio 5.4 with the websocket or > xmlhttprequest > Message-ID: > < > MWHPR05MB291167705F86195717CA7E3181D70(a)MWHPR05MB2911.namprd05.prod.outlook.com > > > > Content-Type: text/plain; charset="iso-8859-1" > > Hello everyone, I'm a beginner in kamailio technology, and I would like to > know how to communicate with kamailio from the web. >

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Using a specific log level for "script" ?
by Chaigneau, Nicolas 04 Jan '21

04 Jan '21

Hello, Is it possible to set a specific log level for the Kamailio script ? I've seen that the < debugger > module allows to do that with modules or < core > : https://kamailio.org/docs/modules/5.4.x/modules/debugger.html#dbg.p.mod_lev… So I've tried to do : modparam("debugger", "mod_level_mode", 1) modparam("debugger", "mod_hash_size", 5) modparam("debugger", "mod_level", "my_module=3") // this works modparam("debugger", "mod_level", "core=3") // also works modparam("debugger", "mod_level", "script=3") // nope... But it doesn't seem to work for < script >... :/ Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

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[Presence] publ_cache=2 usage
by Denys Pozniak 30 Dec '20

30 Dec '20

Hello! Could somebody explain how the parameters *publ_cache=2* and *subs_db_mode* depend on each other? Do I understand correctly that only the following modes of *subs_db_mode* are allowed 0,1,2? *subs_db_mode:0 - This disables database completely. Only memory will be used. Subscriptions will not survive restart. Use this value if you need a really fast presence module and subscription persistence is not necessary or is provided by other means.1 - Write-Through scheme. Subscriptions are updated synchronously in database and in memory(used for read operations). Use this scheme if speed is not top priority, but it's important that no subscriptions will be lost during crash or reboot or if you have an external application that reads the state of the subscriptions from database and they need to be updated synchronously.2 - Write-Back scheme. This is a combination of previous two schemes. All changes are made to memory and database synchronization is done in the timer. The timer deletes all expired contacts and flushes all modified or new subscriptions to database. Use this scheme if you encounter high-load peaks and want them to process as fast as possible. Latency of this mode is much lower than latency of mode 1, but slightly higher than latency of mode 0. To control the interval at which data is flushed to database, set the db_update_period parameter.3 - DB-Only scheme. No memory cache is kept, all operations being directly performed with the database. The timer deletes all expired subscriptions from database. The mode is useful if you configure more servers sharing the same DB without any replication at SIP level. The mode may be slower due the high number of DB operation.* -- BR, Denys Pozniak

1 0

Presence - issue with NOTIFY "terminated" state
by Zhan Bazarov 29 Dec '20

29 Dec '20

Hello everyone! I faced some unfortunately issue with presence module. Let me describe setup scheme a bit: I'm publishing dialog states via pua_dialoginfo from one kamailio server(for calls proccessing itself) to another one kamailio(standalone kamailio just for handling presence,blf,mwi etc) So, about issue, let's say we have device A which has outgoing subscription to extension which belongs to device B and device C(both devices has same extension but different usernames, webrtc+sip phone, effectively its the same Phone, I'm using pubruri_avps to change entities from usernames to extension) . And, in some cases,for examle, when thirtparty is making call to this extension(B,C devices) and making CANCEL after couple seconds - I see the PUBLISH from callserver to presence server with state "terminated', I see 200ok response.. but presence server isn't sending NOTIFY to device A about state "terminated", everything is good with "trying" and "early" states... That problem is happening 1 of 50 tries.. I've tried a lot of things to fix that - still are not succeed. p.s. everything is good from logs/sip point of view... And also, when I disable the webrtc device - I'm not able to reproduce that issue anymore. Looks like something wrong with case, when you have multiple different username behind the same presentity. Maybe someone had same issue and may help me. Thanks! -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Kamailio as a presence watcher
by Jukka Heikkilä 29 Dec '20

29 Dec '20

Hello, I'm working with IMS network and VoLTE related services. My problem is that I would like to use Kamailio as a presence watcher but I haven't managed to make it happen yet. The approach would be that once Kamailio gets 3rd party registration from S-CSCF the Kamailio would send SUBSCRIBE back to S-CSCF to subscribe the presence of subscription. Once a subscription place or get calls the status of subscription would be updated to Kamailio and to the database where it can be fetched. I have checked presence and pua modules, but haven't found a way to make it work as expected. Is there way to do that or is this totally incompatible approach to do that? Best Regards, Jukka Heikkilä jutsco(a)gmail.com

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How to implement a self-sustained Presence server
by Michel Pelletier 28 Dec '20

28 Dec '20

Hi, What is the recommended practice for implementing a self-sustained Presence server from my current Kamailio configuration? That is I want the Presence module to be fed status updates as Kamailio is routing calls and handling registrations. So when a user registers, his phone is ringing, he is on the phone, etc.., the Presence module gets notified and can update its subscribers accordingly. Must I add snippets of codes to all of these events and generate local Publish messages, or is there a better way? I'm using the latest version 5.4.x. Regards, Michel Pelletier

2 1

Kamailio suddenly stops
by Javier Valencia 28 Dec '20

28 Dec '20

Hi there! When I start my webrtc kamailio suddenly stops. I can see this on syslog: Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c2ab8), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c3138), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring ... omit vervose ... Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c1f38), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_free(323) - ignoring ...omit verbose ... Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f557569f250), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_realloc(299) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f557565cf60), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized OS info: root@webrtc:/etc/kamailio# lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster root@webrtc:/etc/kamailio# free -m total used free shared buff/cache available Mem: 16041 134 15551 108 356 15524 Swap: 974 0 974 Kamailio version: version: kamailio 5.4.3 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, 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 with gcc 8.3.0 Kamailio params: SHM_MEMORY=512 PKG_MEMORY=32 SSL info: root@webrtc:/etc/kamailio# apt show openssl Package: openssl Version: 1.1.1d-0+deb10u4 Priority: optional Section: utils Maintainer: Debian OpenSSL Team <pkg-openssl-devel(a)lists.alioth.debian.org> Installed-Size: 1.495 kB Depends: libc6 (>= 2.15), libssl1.1 (>= 1.1.1) Suggests: ca-certificates Homepage: https://www.openssl.org/ Download-Size: 843 kB APT-Manual-Installed: no APT-Sources: http://security.debian.org/debian-security buster/updates/main amd64 Packages Description: juego de herramientas Secure Sockets Layer - utilidad criptográfica Este paquete forma parte de la implementación del proyecto OpenSSL de los protocolos criptográficos SSL y TLS para la comunicación segura a través de Internet. . Contiene el binario de propósito general para la línea de órdenes /usr/bin/openssl, útil para operaciones criptográficas como: * creación de parámetros clave RSA, DH y DSA; * creación de certificados X.509, CSR y CRL; * calcular resúmenes de mensajes; * cifrado y descifrado con cifras; * probar clientes y servidores SSL/TLS; * gestión de correo firmado o cifrado S/MIME. Should I use another version of openssl, or maybe use gnutls? Regards, Javier Valencia

1 0

Self-sustained Presence server.
by Michel Pelletier 28 Dec '20

28 Dec '20

Hi, What is the recommended practice for implementing a self-sustained Presence server from my current Kamailio configuration. That is I want the Presence module to be fed status updates from the same Kamailio that is routing the calls. So when a user registers, his phone is ringing, he is busy, on the phone, etc.., the Presence module gets notified and can update subscribers accordingly. Must I add snippets of codes to all of these events and generate local Publish messages, or is there a better way? I'm using the latest version 5.4.x. Regards, Michel Pelletier

1 0

Read $fd from mysql?
by Mihai Cezar 27 Dec '20

27 Dec '20

Hello, I am redirecting the agents/reps to their asterisk server based on the hostname which they connect, and, i wonder if it's possible to read from db instead of the file cause it's getting big ;) # Dispatch requests route[DISPATCH] { switch ($fd){ case "oro.streamlinepbx.nl": if (!ds_select_dst("1", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } break; case "oro3.streamlinepbx.nl": if (!ds_select_dst("2", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } break; ............ default: log("unknow destination?"); send_reply(503, "No service defined"); xlog("--- SCRIPT: going to <$ru> via <$du> ...Exiting"); exit; } xlog("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } Thanks,

4 3

Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 27 Dec '20

27 Dec '20

A year that was very tough for the entire world is getting close to its end, no doubt that was the most challenging in our life so far for many of us, however, with a distributed, friendly and collaborative community, Kamailio project continued to develop consistently, bringing out a new major release. I trust it helped people and companies world wide to to connect with the dear ones and build a better way of living. The season holidays are ahead, time to rest a bit, spend time with family and friends, being online or face to face. As usual, I take this chance to express my thanks and greetings to all the friends, developers, supporting companies and community members that made 2020 another great year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

5 4

app_lua
by rob.van.den.bulk＠gmail.com 26 Dec '20

26 Dec '20

Hallo, I try to install the module app_lua in Kamailio 5.3. ls: cannot access '/usr/lib/liblua*': No such file or directory ls: cannot access '/usr/lib/liblua*': No such file or directory CC (gcc) [M app_lua.so] app_lua_mod.o In file included from app_lua_mod.c:32: app_lua_api.h:25:10: fatal error: lua.h: No such file or directory #include <lua.h> ^~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: app_lua_mod.o] Error 1 make[1]: *** [Makefile:511: modules] Error 1 make[1]: Leaving directory '/usr/local/src/kamailio-5.3/kamailio/src' make: *** [Makefile:34: all] Error 2 Any help appreciated Greeting Rob van den Bulk

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Fwd: [CFP] FOSDEM 2021, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 23 Dec '20

23 Dec '20

FOSDEM - Real Time Communications devroom CfP ============================================= NOTE: we have extended the deadline but we will give preference to people who submitted earlier when we set the time for each talk. Please submit your proposal ASAP so that you can get your preferred time slot. Overview -------- [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2021 takes place 6-7 February 2021 and for the very first time, it will be online. This document contains information about: - Real-Time Communications developer room (devroom) - speaking opportunities - volunteering New rules for the online edition -------------------------------- This year FOSDEM will be fully online instead of being held in Brussels, here are the most important things to know about this (quite significant) change: - The reference time will be Brussels local time (CET) - Talks will be pre-recorded in advance, and streamed during the event - The Q/A session will be live - A facility will be provided for people watching to chat between themselves - A facility will be provided for people watching to submit questions Call for participation - Real Time Communications (RTC) ------------------------------------------------------- The Real-Time devroom is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers who can help manage the scheduling and live Q&A sessions.** The devroom is only on Saturday, 6th of February 2021. To discuss the devroom, volunteer or ask questions, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Key dates - 20th Dec: Submission deadline (extended to 8 January) - 24th Dec: Announcement of selected talks - 15th Jan: Presentations upload deadline - 6th & 7th Feb: Conference dates (online) - 6th Feb: RTC devroom date (online) ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 20th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM21/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. This year this is more true than ever, being able to record your presentation offline without an audience in front can greatly help build up one's confidence! ### 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 (devrooms, 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 devroom 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 devroom administrators based on the number of received proposals. As the two previous devrooms 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. ### Recording help The devroom organization is able to provide help with recording your session. The recording would be performed at a scheduled time with one of us, so you won't be alone giving your presentation. Minimal edits will be possible, but the ideal plan is to record it in one shot. Thanks Dan Jenkins for providing us with the means to do this! Volunteers needed ----------------- To make the devroom run successfully, we are looking for volunteers. This year many things be done for the first time, so all the help we can get is more than welcome. Spread the word and discuss --------------------------- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom 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 https://planet.freertc.org planet(a)freertc.org - XMPP https://planet.jabber.org ralphm(a)ik.nu - SIP https://planet.sip5060.net 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 devroom 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>

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MID-Registrar Example
by moisesynfante 23 Dec '20

23 Dec '20

Hello, I'm a newbie when it comes to Kamailio and I would like to know if there's a working example of a Mid-Registrar implementation. I read on other posts that the UAC module can be used to achieve this but, honestly, I don't understand how. If any of you could provide some path I could follow I would be very thankful. -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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Re: [SR-Users] How to use port number using $env in listen
by Syed Shahryar 22 Dec '20

22 Dec '20

Any updates on this? On Tue, Dec 15, 2020 at 8:13 PM Syed Shahryar <syedshahryar(a)gmail.com> wrote: > I am new to Kamailio so may be doing something wrong. I looked at > https://www.kamailio.org/wiki/cookbooks/5.4.x/core. > > I tried the following combinations but getting errors in all cases: > > #!subst "/EVAPI_PORT/$env(EVAPI_PORT)/" > #!substdef "/EVAPI_PORT/$env(EVAPI_PORT)/" > #!substdefs "/EVAPI_PORT/$env(EVAPI_PORT)/" > > *Errors:* > 0(6) ERROR: <core> [core/pvapi.c:924]: pv_parse_spec2(): error searching > pvar "env" > 0(6) ERROR: <core> [core/pvapi.c:1127]: pv_parse_spec2(): wrong char > [E/69] in [$env(EVAPI_PORT)/] at [5 (5)] > 0(6) ERROR: <core> [core/re.c:170]: parse_repl(): bad specifier in > replace part /$env(EVAPI_PORT)/ > 0(6) ERROR: <core> [core/ppcfg.c:69]: pp_subst_add(): bad subst > expression: /EVAPI_PORT/$env(EVAPI_PORT)/ > > At a later step, I would like to do: > #!define EVAPI_LISTEN "0.0.0.0:EVAPI_PORT" > > And use EVAP_LISTEN in mod_param for the evapi module. > > And something similar to above for TLS settings. > > > On Tue, Dec 15, 2020 at 12:41 PM Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> as written on the tracker, look at using #!subst or #!substdef (see core >> cookbook), they should be able to get the substitution value from >> environment using $env(NAME). >> >> The master branch introduced #!defenv as a (simpler) alternative. >> >> Cheers, >> Daniel >> On 12.12.20 05:33, Syed Shahryar wrote: >> >> I tried many combinations of the following but I keep getting syntax >> errors: >> >> listen=tls:TLS_IP:$env(TLS_PORT) advertise PUBLIC_IP:$env(TLS_PORT) >> >> >> TLS_IP and PUBLIC_IP are constants defined at the top using #!define. >> And, TLS_PORT is supposed to be the env variable. >> >> >> -- >> -- >> This message contains confidential information and is intended only for >> the individual named. If you are not the named addressee, you should not >> disseminate, distribute or copy this email. Please notify the sender >> immediately by email if you have received this email by mistake and delete >> this email from your system. Email transmission cannot be guaranteed to be >> secure or error-free, as information could be intercepted, corrupted, lost, >> destroyed, arrive late or incomplete, or contain viruses. The sender, >> therefore, does not accept liability for any errors or omissions in the >> contents of this message which arise as a result of email transmission. If >> verification is required, please request a hard-copy version. >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Funding: https://www.paypal.me/dcmierla >> >> > > -- > -- > This message contains confidential information and is intended only for > the individual named. If you are not the named addressee, you should not > disseminate, distribute or copy this email. Please notify the sender > immediately by email if you have received this email by mistake and delete > this email from your system. Email transmission cannot be guaranteed to be > secure or error-free, as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender, > therefore, does not accept liability for any errors or omissions in the > contents of this message which arise as a result of email transmission. If > verification is required, please request a hard-copy version. > -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

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ndb_redis with keydb master/master
by David VILLAUME 22 Dec '20

22 Dec '20

Hello, I consider using keydb-server in an active/action setup without sentinel. Do you know if there is a way to declare both hosts in ndb_redis with the same server name (target use : topos) ? Regards, David

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Re: [SR-Users] DLGLOAD stuck in Dispatcher Module(HA Mode)
by harneet singh 22 Dec '20

22 Dec '20

Apologies for not having included the sr-users in the previous email. Anyhow, the solution suggested by Social Boh indeed worked, as a side effect of which we would not run into the issue described. *NOTE:* There was a peculiar configuration in keepalived with keeping both the nodes as BACKUP and setting the 'nopreempt' flag only in the actual MASTER(one with higher 'priority') side to make the keepalived thing to work properly. However, I was wondering if someone could confirm if this issue has been fixed inside the source code itself as I am using an older release v5.3.2 Regards, Harneet Singh On Thu, Dec 17, 2020 at 9:52 PM harneet singh <hbilling(a)gmail.com> wrote: > Thanks Social Boh, > > I will investigate this possibility and if it does work in our case, that > should circumvent the issue. > In addition, is there a fix known to the SR-Users which could be available > inside the Dispatcher module to cause syncing just like the syncing is > there in the Dialog module via DMQ. I wanted to check this as I am using > v5.3.2 and unaware if new releases could have fixed this issue? > > Regards, > Harneet Singh > > On Wed, Dec 16, 2020 at 10:46 PM Social Boh <social(a)bohboh.info> wrote: > >> Maybe a option is leave the secondary like primary when primary go up >> newly. >> >> Example: Primary go down, Secondary is the new Primary. Primary came back >> up but Secondary still is the VIP and still process the calls. >> >> Only if Secondary go down, primary is the new VIP. >> >> The keepalived parameter for this behaviour is: >> >> *nopreempt* >> >> --- >> I'm SoCIaL, MayBe >> >> El 16/12/2020 a las 11:57 a. m., harneet singh escribió: >> >> Hi All, >> >> I am using Kamailio in HA mode with Keepalived providing the VIP(Virtual >> IP) functionality, and have run into a rather peculiar issue. >> >> *Setup:* >> >> Caller ------------ KamailioVIP(*Primary *and *Secondary* >> Kamailio)------- Callee >> >> HA provided by Keepalived >> DMQ is used between the Primary and Secondary Kamailio for dialog sync >> >> *Issue Seen:* >> >> *Suppose the Primary Kamailio has been brought down* and the Secondary >> one is actually active and tied to the VIP. >> >> When a call is fired from the Caller, it traverses through the Secondary >> Kamailio and lands onto the callee. The dialogs are updated properly. At >> this point, the Primary Kamailio is brought up, and dialog state is synced >> due to the DMQ module. >> The Keepalived will now attach the VIP to the Primary Instance. If the >> caller hangs up the phone at this point, the BYE sip message traverses >> through the Primary Kamailio instance to the callee and the call is >> cleared, but there are two issues here. >> >> 1. The Primary Kamailio throws an error in *ds_load_remove() *saying >> that it cannot find the load for the specific call id. This is apparently >> due to the fact that the dispatcher data is not synced between the two >> modules but dialog data is. So dialog wise, things are fine. Can this be >> fixed somehow? >> >> 2. The above is still as grave a problem as the dialogs are cleared. >> BUT, if we check the '*kamcmd dispatcher.list*' on the *SECONDARY* >> kamailio even well after the call is cleared, the *DLGLOAD* shows 1. >> Since we are using the *Call Load based distribution* for the >> dispatching, this is effectively one call stuck on the dispatcher, which >> leads to resource leak. >> >> Is this a known issue, and if so, do we have a way to circumvent this? >> >> Regards, >> Harneet Singh >> -- >> "Once you eliminate the impossible, whatever remains, no matter how >> improbable, must be the truth" - Sir Arthur Conan Doyle >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> > > -- > "Once you eliminate the impossible, whatever remains, no matter how > improbable, must be the truth" - Sir Arthur Conan Doyle > -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

1 0

Kamailio v5.3.8 released
by Henning Westerholt 21 Dec '20

21 Dec '20

Hello, Kamailio SIP Server v5.3.8 stable has been released. This is a maintenance release of the previous stable branch (5.3), that includes fixes since the release of v5.3.7. There are no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are recommended to be upgraded to v5.3.8. For more details about version 5.3.8 (including links and guidelines to download the tar file or from GIT repository), visit: https://www.kamailio.org/w/2020/12/kamailio-v5-3-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

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MSTeams Kamailio with PBX Hold Refer issue
by me.projects＠yahoo.com 21 Dec '20

21 Dec '20

Hi, I have been trying to integrate MSTeams with PBX Through Kamailio. Calls are working now. Hold has Issues. MSTeams uses REFER to manage this all with NOTIFY. I am sending t_send_reply("202", "Accepted"); when receive Refer now I need to send NOTIFY I think I can use t_uac_send() NOTIFY I am unable to send t_uac_send properly on when recertifications refer from MSTeams. Any one with the experience on this ? Thanks,

2 1

Trying to use $sel(contact.uri.params)
by Chaigneau, Nicolas 21 Dec '20

21 Dec '20

Hello, I'd like to use a < select > as described here: http://www.kamailio.org/wiki/cookbooks/5.4.x/selects#contactparams_s I'm trying to extract a parameter from the Contact URI. What I've tried to do: xlog("L_INFO", "contact.uri: $sel(contact.uri)\n"); # OK xlog("L_INFO", "contact.uri.params: $sel(contact.uri.params)\n"); # null ?? This does not seem to work... see output here : 3(10316) INFO: {1 INVITE r8wEHsAgxNIS5OJL} <script>: contact.uri: sip:+33145678901;tgrp=YC031;trunk-context=YC90@10.31.22.2:5099 3(10316) INFO: {1 INVITE r8wEHsAgxNIS5OJL} <script>: contact.uri.params: <null> What am I doing wrong ? Maybe this is the wrong approach for what I'm trying to do ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 2

Releasing Kamailio 5.3.8
by Henning Westerholt 21 Dec '20

21 Dec '20

Hello, I am planning to release next week (likely on Monday 21st December) the next minor version out of branch 5.3, version 5.3.8. Backports of fixed issues from git master will be integrated as usual. If you are aware of new issues which were not yet reported, let us know (by creating an issue on the tracker). Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

send MESSAGE to other sessions of same user
by Syed Shahryar 21 Dec '20

21 Dec '20

What's the best way of sending a message to all other sessions (and the receiver) of the same sender? For example: If user1(a)example.com is logged in to Kamailio from multiple devices and from one of the devices user1 sends a SIP message to user2(a)example.com. I want the message forwarded not only to user2(a)example.com but also to all other user1(a)example.com sessions. What's the easiest way to achieve this? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

1 0

Kamailio not updating CSeq after uac_auth() for ACK
by Terry Tailor 18 Dec '20

18 Dec '20

Hi, Here is setup details Kamailio Proxy a <-----> Kamailio Proxy b <-----> Asterisk Kamailio proxy b auth based on IP Kamailio Proxy a 1. Kamailio proxy sends an Invite request to Kamailio proxy b with CSeq: 102 INVITE 2. Kamailio proxy b check again IP auth and sends invite request to asterisk with CSeq: 102 INVITE 4 asterisk response with 401 Unauthorized with CSeq: 102 5. Kamailio proxy b sends ACK request with CSeq: 102 ACK 6. Kamailio proxy b sends INVITE request with CSeq: 103 INVITE 7. Asterisk sends 100 trying and 180 ringing with CSeq: 103 INVITE 8. Kamailio proxy b relay 180 ringing with CSeq: 102 INVITE 9. Asterisk send 200 Ok with CSeq: 103 INVITE and proxy b relay it to proxy a with CSeq: 102 INVITE 10. Now Proxy a send an ACK with CSeq: 102 ACK and proxy b relay ACK to asterisk with CSeq: 102 ACK Now asterisk is retransmitting 200 ok with CSeq: 103 INVITE and proxy a is sending ACK with Cseq: 102 ACK Same happing in case of 487 Request Terminated Here is my config modparam("dialog", "default_timeout", 10800) # 3h modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 0) modparam("dialog", "send_bye", 1) modparam("dialog", "track_cseq_updates", 1) modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 3 sec modparam("tm", "fr_timer", 2000) modparam("tm", "retr_timer1", 250) modparam("tm", "retr_timer2", 500) # default invite retransmission timeout after 1xx: 60 sec modparam("tm", "fr_inv_timer", 60000) modparam("tm", "reparse_on_dns_failover", 0) if (is_method("INVITE") && !has_totag()) { dlg_setflag(FLD_RTPPROXY); dlg_manage(); } route(RELAY); route[RELAY] { # At this point all checks are passed t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); t_on_failure("MANAGE_FAILURE"); if(is_method("INVITE") && isflagset(FLT_FROMDID)) { #$fd = "sip.comms.kuflink.com"; t_on_failure("RELAY_AUTH"); } if (!t_relay()) { send_reply("500", "Cannot relay to $du/$ru"); } exit; } failure_route[RELAY_AUTH] { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { #$fd = "sip.comms.kuflink.com"; $avp(auser) = "DID_UA_USERNAME"; $avp(apass) = "DID_UA_PASSWORD"; if (uac_auth()) { t_relay(); } exit; } } route[WITHINDLG] { if (!has_totag()) { return; } if (is_method("BYE")) { rtpengine_delete(); } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessly route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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; } } send_reply("404","Not here"); exit; } kamailio -v version: kamailio 5.3.7 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, 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 with gcc 6.3.0 uname -a Linux example.com 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 GNU/Linux SIP logs 2020/12/17 10:08:18.483907 proxy-a.com:5060 -> 10.0.14.118:5060 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 102 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 2020/12/17 10:08:18.485063 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1;rport=5060 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: kamailio (5.3.7 (x86_64/linux)) Content-Length: 0 2020/12/17 10:08:18.485576 10.0.14.118:5060 -> 10.0.14.118:6000 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 102 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 2020/12/17 10:08:18.622867 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as4107645c Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sip1.comms.kuflink.com", nonce="11139d33" Content-Length: 0 2020/12/17 10:08:18.624034 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@proxy-a.com SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as4107645c Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK Content-Length: 0 2020/12/17 10:08:18.625557 10.0.14.118:5060 -> 10.0.14.118:6000 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 103 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 Authorization: Digest username="xxxxx", realm="xxxx", nonce="11139d33", uri="sip:123456@proxy-a.com", response="qwewqeqwe", algorithm=MD5 2020/12/17 10:08:18.779483 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Length: 0 2020/12/17 10:08:20.937756 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Length: 0 2020/12/17 10:08:20.938593 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Length: 0 2020/12/17 10:08:22.075814 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:22.077049 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316 2020/12/17 10:08:22.103365 proxy-a.com:5060 -> 10.0.14.118:5060 ACK sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.0fc1940efc770391762d7d67094c24f8.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK2a5fae72;rport=5060 Route: <sip:proxy-b.com;lr;did=d18.7a9> Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 2020/12/17 10:08:22.103877 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@10.0.14.118:6000;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.79f351d031912aae585207d2f26d5650.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.0fc1940efc770391762d7d67094c24f8.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK2a5fae72;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 P-K-CSeq-Refresh: 103 2020/12/17 10:08:22.575533 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:22.576777 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316 2020/12/17 10:08:22.581649 proxy-a.com:5060 -> 10.0.14.118:5060 ACK sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.b9c7eeccd4cd7b44f05b173308dbaf0a.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK219933f6;rport=5060 Route: <sip:proxy-b.com;lr;did=d18.7a9> Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 2020/12/17 10:08:22.582313 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@10.0.14.118:6000;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.33626b5695ad4648ba393bc62bed6a8c.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.b9c7eeccd4cd7b44f05b173308dbaf0a.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK219933f6;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 P-K-CSeq-Refresh: 103 2020/12/17 10:08:23.575828 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:23.577863 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316

4 7

Kamailio RTP proxy require?
by Mihai Cezar 17 Dec '20

17 Dec '20

Hi, Can't figure it out when to use RTPproxy, i am forwarding everything to asterisk in seem to work without RTPproxy. The scenario is like this: Softphone -> INTERNET -> Kamailio - > Asterisk (private ip) Now it seems to work without any rtpproxy but is it ok?

5 10

DLGLOAD stuck in Dispatcher Module(HA Mode)
by harneet singh 16 Dec '20

16 Dec '20

Hi All, I am using Kamailio in HA mode with Keepalived providing the VIP(Virtual IP) functionality, and have run into a rather peculiar issue. *Setup:* Caller ------------ KamailioVIP(*Primary *and *Secondary* Kamailio)------- Callee HA provided by Keepalived DMQ is used between the Primary and Secondary Kamailio for dialog sync *Issue Seen:* *Suppose the Primary Kamailio has been brought down* and the Secondary one is actually active and tied to the VIP. When a call is fired from the Caller, it traverses through the Secondary Kamailio and lands onto the callee. The dialogs are updated properly. At this point, the Primary Kamailio is brought up, and dialog state is synced due to the DMQ module. The Keepalived will now attach the VIP to the Primary Instance. If the caller hangs up the phone at this point, the BYE sip message traverses through the Primary Kamailio instance to the callee and the call is cleared, but there are two issues here. 1. The Primary Kamailio throws an error in *ds_load_remove() *saying that it cannot find the load for the specific call id. This is apparently due to the fact that the dispatcher data is not synced between the two modules but dialog data is. So dialog wise, things are fine. Can this be fixed somehow? 2. The above is still as grave a problem as the dialogs are cleared. BUT, if we check the '*kamcmd dispatcher.list*' on the *SECONDARY* kamailio even well after the call is cleared, the *DLGLOAD* shows 1. Since we are using the *Call Load based distribution* for the dispatching, this is effectively one call stuck on the dispatcher, which leads to resource leak. Is this a known issue, and if so, do we have a way to circumvent this? Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 1

INFO: RTC Devroom Fosdem 2021 - Online Event
by Daniel-Constantin Mierla 16 Dec '20

16 Dec '20

Hello, (cross-posting to kamailio public groups, a few may get this message duplicated). Fosdem 2021 is going to be online, meaning that it opens the possibility for everyone to do a presentation on the RTC Devroom, no longer being required to travel to Brussels. Therefore I am sending here the link to CFP in case someone would be interested to present about Kamailio or something else RTC-related: * https://github.com/FreeRTC/fosdem/blob/master/2021/cfp.md New speakers are always welcome! Only have in mind that the event is targeting developers and source code development, therefore your proposals have to be made from this perspective. Note also that submission deadline is 20th of December, very short time frame from now. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

ERROR: dmq [notification_peer.c:337]: add_server_and_notify(): error adding notification node
by Syed Shahryar 16 Dec '20

16 Dec '20

Here is my setting: modparam("dmq", "server_address", "sip:0.0.0.0:5090" ) modparam("dmq", "notification_address", "sip:0.0.0.0:5090") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 2) modparam("dmq", "worker_usleep", 0) modparam("dmq", "ping_interval", 90) I have tried localhost and I believe other options but to no avail. I also get the following message: ERROR: dmq [dmq_funcs.c:516]: ping_servers(): cannot retrieve initial nodelist from sip:0.0.0.0:5090 How do I set up DMQ for adding and removing any Kamailio instance to/from the cluster and any Kamailio instance could serve any request coming from any client? ie. all Kamailio instances keep the current state in DMQ. I also have dmq_usrloc enabled. -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Kamailio dtmf signaling
by Amit Pal 16 Dec '20

16 Dec '20

Dear Team, I am using kamailio + rtpengine, how can I processed dtmf from cfg scripts. even thought I can able to see dtmf over syslog. but can not able to capture in kamailio.cfg. how can I prepared ivr using kamalio? Please find the attached syslog. Regards Amit Pal

3 2

ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri </sip?jwt=..
by Syed Shahryar 15 Dec '20

15 Dec '20

I am sending JWT tokens via SIP URL through JSSIP for authentication before websocket connection is made. However, I am getting errors like: ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri < */sip?jwt=..* I have the following setting in kamailio config: modparam("xhttp", "url_match", "^/sip") // only match this path Is this a bug or some configuration issue? How do I resolve it? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

X-Forwarded-for IP to asterisk
by Mihai Cezar 15 Dec '20

15 Dec '20

Hello, Is it something like the XFF for Kamailio so I can see in asterisk (sip show peers) the ip of the registered softphone and not Kamailio's IP ? Thanks!

4 5

How to use port number using $env in listen
by Syed Shahryar 15 Dec '20

15 Dec '20

I tried many combinations of the following but I keep getting syntax errors: listen=tls:TLS_IP:$env(TLS_PORT) advertise PUBLIC_IP:$env(TLS_PORT) TLS_IP and PUBLIC_IP are constants defined at the top using #!define. And, TLS_PORT is supposed to be the env variable. -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Losing MySQL connections
by Duarte Rocha 15 Dec '20

15 Dec '20

Greetings all, I'm having some dropped MySQL connections in my Kamailio in times of low or no activity in the proxy. These are the errors i'm getting : db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013) <core> [db_query.c:348]: db_do_update(): error while submitting query usrloc [ucontact.c:1147]: db_update_ucontact_ruid(): updating database failed usrloc [ucontact.c:1663]: update_contact_db(): failed to update database registrar [save.c:784]: update_contacts(): failed to update contact sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) In order to try to fix this i've added timeout_interval and ping_interval from the db_mysql module. Is this the best approach? I've restarted my Kamailio but i don't see any ping going to my BD. Is there any other config that I need to add so that the pings are sent? Is it possible to do a ping to verify connection and reconnect mannualy on the config file? Best Regards,

2 1

Problems debugging memory issues
by Sheldon L 15 Dec '20

15 Dec '20

Hi, I'm trying to investigate a possible memory leak issue in Kamailio and I'm following the instructions indicated in https://www.kamailio.org/wiki/tutorials/troubleshooting/memory I'm using Kamailo 4.4 with the following kamailio internals: Print out of kamailio internals Version: kamailio 4.4.2 (x86_64/linux) 892ad6 Default config: /apollo/env/TachyonPlatformKamailio/etc/kamailio/kamailio.cfg Default paths to modules: /apollo/env/TachyonPlatformKamailio/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_LISTEN=16 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: 892ad6 Compiled with: x86_64-unknown-linux-gnu-gcc 4.9.4 Compiled on: 23:36:48 Dec 8 2020 Thank you for flying kamailio! I have attached to one of the kamailio tcp receiver processes via GDB and I am trying to dump memory blocks from the memory manager. However, mem_block is not in the current_context. The following is the output from my gdb session: Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-amazon-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /opt/codedeploy-agent/deployment-root/95bb1634-e653-4cec-b327-99b18b9227a8/d-7AJ1XZP58/deployment-archive/sbin/kamailio...done. Attaching to program: /apollo/env/TachyonPlatformKamailio/sbin/kamailio, process 8946 ptrace: Operation not permitted. (gdb) set $i=0 (gdb) set $a = mem_block->first_frag No symbol "mem_block" in current context. (gdb) info list Undefined info command: "list". Try "help info". (gdb) info threads No threads. Is there something I need to do before accessing the mem_block symbol? Also do you have any other suggestions for debugging memory leaks that may not be documented? Note: i can't upgrade to the latest version of Kamailio as we have custom modules and would require considerable effort to upgrade. Thanks, Sheldon.

2 1

sngrep TLS connection
by Agiftel 15 Dec '20

15 Dec '20

Hi all, does anyone was able to use sngrep (with -k option) to decrypt TLS connections? I'am doing what help says; used private key is the correct one; but no traffic is seen on sngrep. If i use not ecrypted call i can see everything, so sngrep is working good. Any hint? regards -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

5 8

Kamilio+asterisk = deregister after 2-3 minute
by Mihai 15 Dec '20

15 Dec '20

Hi, I am forwarding all the traffic to my asterisk server ( including registers) everything seems to work when i first register but if stay idle 2-3 minute it gets unregistered and then i can't register back unless i restart kamailio see attached print screen

6 6

voltdb support
by Ali Taher 14 Dec '20

14 Dec '20

Hello, I just need to know if kamailio can communicate with Voltdb for sqlops operations for example. Regards, Ali Taher

2 8

Kamailio v5.4.3 Released
by Daniel-Constantin Mierla 14 Dec '20

14 Dec '20

Hello, Kamailio SIP Server v5.4.3 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.3. For more details about version 5.4.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/12/kamailio-v5-4-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Wrong To tag in t_reply
by Ivan Ribakov 14 Dec '20

14 Dec '20

Hi all, I’m trying to implement following routing behaviour: - ingress INVITE is sequentially forked to 1 or more endpoints - under certain conditions sequential forking may be stopped and pre-defined error returned to the caller Here’s is part of the config responsible for handling errors: failure_route[ROUTE_ELEMENT_FINAL_FAILURE_HELPER] { route(ROUTE_ELEMENT_FINAL_FAILURE); } route[ROUTE_ELEMENT_FINAL_FAILURE] { # We need to check if the error happens before the first egress, because T-state is not defined yet so we can not use t_reply if ($var(before_first_egress) == 0) { t_reply("$var(sipCode)", "$var(sipMessage)"); } else { sl_send_reply("$var(sipCode)", "$var(sipMessage)"); } } Here is a simple scenario where delivering call to first endpoint times out after receiving 180 Ringing (using custom fr_inv_timer value) where: - .33 is caller - .208 is Kamailio - .138 is callee Notice that 504 is retransmitted by Kamailio. Here is a comparison of 180 and 504 returned by Kamailio: As can be seen, 180 was proxied from the callee and has callee’s to-tag (tag=CALLEE1), however 504 that was generated by Kamailio using t_reply() has a different to-tag (presumably generated by Kamailio). How can I tell Kamailio not to generate new to-tag but to use the last one that proxied back to the caller? Thanks in advance for any help! Regards, Ivan

2 2

Releasing Kamailio v5.4.3
by Daniel-Constantin Mierla 14 Dec '20

14 Dec '20

Hello, I am considering to release Kamailio v5.4.3 sometime next week, likely on Monday, December 14, 2020. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

how to use environment variables to set listen address
by Syed Shahryar 13 Dec '20

13 Dec '20

This is a generic question about using environment variables in a config file. If I want to use below where TLS_IP itself is a constant defined somewhere else. TLS_IP might be set via an env variable as well. #!define TLS_TO_LISTEN tls:TLS_IP:$env(TLS_PORT) Will that work? I see syntax errors and other weird issues. What's the right syntax? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

DMQ using Consul
by Syed Shahryar 12 Dec '20

12 Dec '20

Can someone share their configuration file explaining how they used consul to achieve high availability? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

empty body on first NOTIFY on REGINFO subscription - bug ?
by Adrian Tabacioiu 11 Dec '20

11 Dec '20

Hello, If the user never registered (or few hours passed from its unregister) the subscription to REGINFO will return an empty body NOTIFY (no xml). I don't see any flag in module to force a body for this case, as in the presence module is force_dummy_presence who can force a body on a first NOTIFY. The RFC3680 says : o Notifications triggered from a SUBSCRIBE contain full state (the list of all contacts bound to the address-of-record). It appears that in this case the response is not conform to RFC. Regards, Adrian

1 0

porting keepalive module new functions on 5.4 on 5.3?
by Benoit Chesneau 11 Dec '20

11 Dec '20

2 new functions have been added to the keepalive module: - [ka_add_destination(sip_uri, owner)](https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keep…) - [ka_del_destination(sip_uri, owner)](https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keep…) Is there any possibility to do the same on 5.3? It seems that on 5.3 adding a destination can only be done statically: https://kamailio.org/docs/modules/5.3.x/modules/keepalive.html#idm53 or can it also be done dynamically when receiving an event using the kazoo module for example? Benoît Chesneau, Enki Multimedia — t. +33608655490 Sent with [ProtonMail](https://protonmail.com) Secure Email.

2 1

Dispatcher algorithm 11 possible priority drifting
by Володимир Іванець 10 Dec '20

10 Dec '20

Hello all! We are running Kamailio version 5.3.3 with 16 small Asterisk servers. Kamailio uses a dispatcher module to distribute calls. Algorithm #11 is selected. All destinations are configured with priority set to 50 and attribute to rweight=50. We noticed that one server constantly receives more calls than others. I run a few tests. I was sending 1000 calls to the system. 4 per second. All servers except one were getting around 60 calls while the last one - around 100. I then noticed that the server which receives most calls is always last in the "kamcmd dispatcher.list" command output. I then changed the order in the dispatcher DB table and repeat the test. The other server that now was last was getting the most calls. Does anyone else use algorithm #11 and finds the same thing? Is there something additional that I can provide to help with the investigation? Thanks!

3 3

kamailio.cfg fomatter
by Juha Heinanen 10 Dec '20

10 Dec '20

After noticing that emacs mode is missing, below is my first cut of kamailio-mode.el. -- Juha ----------------------------------------------------------------------- ;; kamailio mode for emacs (function list is not complete) ;; activate by starting your kamailio.cfg with this kind of line: ;; #-*-mode: kamailio; tab-width: 4; c-basic-offset: 4; (setq kamailio-font-lock-keywords (let* ( ;; define several category of keywords (x-keywords '("branch_route" "break" "case" "default" "else" "event_route" "exit" "failure_route" "for" "if" "loadmodule" "onreply_route" "onsend_route" "reply_route" "request_route" "return" "route" "switch" "while")) (x-types '()) (x-constants '()) (x-events '()) (x-functions '("acc_db_request" "add_path_received" "add_rr_param" "allow_routing" "allow_trusted" "append_hf" "append_to_reply" "avp_check" "avp_delete" "change_reply_status" "check_route_param" "defined" "dp_translate" "ERROR" "fix_nated_sdp" "force_send_socket" "forward" "get_redirects" "handle_ruri_alias" "inactivate_gw" "INFO" "in_list" "isflagset" "is_avp_set" "isbflagset" "is_direction" "is_domain_local" "is_e164" "is_gflag" "is_gruu" "is_method" "is_numeric" "is_present_hf" "jsonrpc_dispatch" "load_gws" "lookup_domain" "lookup" "loose_route" "modparam" "mt_match" "NOTICE" "proxy_challenge" "record_route" "redirecting" "remove_hf" "resetflag" "rtpengine_answer" "rtpengine_delete" "rtpengine_offer" "save" "sdp_content" "sdp_transport" "send_reply_mode" "send_reply" "setbflag" "setflag" "setxflag" "set_reply_close" "set_reply_no_connect" "set_rtpengine_set" "set_uri_host" "set_uri_user" "set_forward_no_connect" "sql_query" "sql_result_free" "starts_with" "subst_uri" "tel2sip" "t_check_status" "t_is_canceled" "t_load_contacts" "t_next_contacts" "t_on_branch" "t_on_failure" "t_on_reply" "t_relay_cancel" "t_relay" "t_set_fr" "unregister" "WARN" "ws_handle_handshake" "www_challenge" "xhttp_reply")) (x-keywords-regexp (regexp-opt x-keywords 'words)) (x-types-regexp (regexp-opt x-types 'words)) (x-constants-regexp (regexp-opt x-constants 'words)) (x-events-regexp (regexp-opt x-events 'words)) (x-functions-regexp (regexp-opt x-functions 'words))) `( (,x-types-regexp . font-lock-type-face) (,x-constants-regexp . font-lock-constant-face) (,x-events-regexp . font-lock-builtin-face) (,x-functions-regexp . font-lock-function-name-face) (,x-keywords-regexp . font-lock-keyword-face) ))) (defvar kamailio-mode-syntax-table nil "Syntax table for `kamailio-mode'.") (setq kamailio-mode-syntax-table (let ( (synTable (make-syntax-table))) ;; python style comment: “# …” (modify-syntax-entry ?# "<" synTable) (modify-syntax-entry ?\n ">" synTable) synTable)) (define-derived-mode kamailio-mode c-mode "kamailio mode" "Major mode for editing Kamailio configuration files" (setq font-lock-defaults '((kamailio-font-lock-keywords))) (set-syntax-table kamailio-mode-syntax-table)) (provide 'kamailio-mode)

2 5

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 10 Dec '20

10 Dec '20

Hello Kamailions, Well, I think I found the cause of my issue where the Dialog profile was not defined.. Well it was. I had not enabled antiflood, by not enabling antiflood, the htable module was also not loaded... I guess the us of profiles_with_value parameter needs htable? Is that correct? If so, that means the htable is a dependency for the dialog module...., then it would be sometihng that could be added to the documentation of the dialog module. When this is not the case, then I have to dig deeper, as I do not get an error anymore about the profile not being known... Thank you. Gertjan

2 2

REGINFO events not sending
by Adrian Tabacioiu 10 Dec '20

10 Dec '20

Hello, Does anybody has used with success the reginfo modules ? I added lines bellow in configuration, and tried different variations, Even if my subscribe for "reginfo" events is accepted (with 202 OK), I receive no notification when respective user registers. Nor do I get the xml body in the first notification. Added in my config: loadmodule "presence_reginfo.so" loadmodule "pua_reginfo.so" modparam("pua_reginfo", "default_domain", "192.168.60.65") modparam("pua_reginfo", "server_address", "sip:reginfo@192.168.60.65") modparam("pua_reginfo", "publish_reginfo", 1) Thank you in advance, Adrian

2 3

SLA presence not working proper with (SCA module)
by Aatif Shaikh 10 Dec '20

10 Dec '20

Aatif Shaikh <aatif.shaikh(a)ecosmob.com> Mon, Dec 7, 11:00 AM (3 days ago) to sr-users Hello All I configure my kamailio.cfg to handle SLA using (SCA module) and shared presence is working but still facing some issue in case of hold invite and BYE and the reason behind of that issue is following error Nov 26 07:38:40 SBC-4-1 /usr/local/sbin/kamailio[104107]: ERROR: sca [sca_call_info.c:1008]: sca_call_info_invite_request_handler(): Failed to update sip:4569@test.sip.abcd.com appearance-index 0 to active actually, in Kamailio appearances, I can see the entry for this call but the index is 1 instead of 0, and Kamailio going to update with a 0 index [root@SBC-4-1 ntcarfte-kamailio]# kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 active 1606376289 sip:4569@10.xx.xx.xx :5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120MTQ5OTcyZjFhMjMyNmI1ZGE1MWY4ODc2M2RkN2VmZmQ 45dc18d8 H6cS9vecZ248B if I manually update appearances using RPC command with index 1 then the phone started blinking as I change state to held # kamcmd sca.update_appearance sip:4569@sc4test.sip.teledge.com 1 held # kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 held 1606376516 sip:4569@10.xx.xx.xx :5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120NzE5MDgyZDc1NjJiYjcwMWFlYmI3NzM3NjE2OTRhZjU 08f005cb eK9ratm83g92e kamailio.cfg route[SCA] { if(is_method("SUBSCRIBE")) { if ($hdr(Event) == "call-info" || $hdr(Event) == "line-seize") { xlog("L_INFO","(MAIN) :HELLO ($avp(uuid)) : $avp(rsi) $sp $hdr(Event)"); xdbg("SCA: $hdr(Event) SUBSCRIBE $ru from $si:$sp"); sca_handle_subscribe(); exit; } return; } if (!is_method("BYE|CANCEL|INVITE|PRACK|REFER")) { return; } sca_call_info_update(); } route[RELAY_OUTBOUND_FS] { xlog("L_INFO","(RELAY_OUTBOUND_FS) : ($avp(uuid)) : INSIDE ROUTE "); if (!has_totag()){ $avp(set_h) = 1; } t_on_reply("REPLY_OUTBOUND_FS"); route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } route[RELAY_INBOUND_FS] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(s:puburis_caller) = $fu; setflag(8); dlg_manage(); } if(is_method("REGISTER")) { t_on_reply("REPLY_REGISTER"); } else { t_on_reply("REPLY_INBOUND_FS"); } route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[REPLY_INBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } onreply_route[REPLY_OUTBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } can someone help me with this Did I missed something to handle SLA?

1 0

NAT problem with recvonly calls
by David Cunningham 10 Dec '20

10 Dec '20

Hello, We have a problem with a SIP doorbell device which sends media one way only, and NAT at the receiving device. When the doorbell button is pressed it makes a call to a configured destination. Since the doorbell only sends and doesn't receive it sends the INVITE with sendonly in the SDP, and the destination then replies with a 200 OK with recvonly in the SDP. The problem is that the destination is behind NAT, and its reply contains a private network IP in the SDP. Normally Asterisk when nat=yes works around that by adjusting the destination for RTP to be the address it actually receives audio from, however because this device is recvonly Asterisk never receives audio from it. This means Asterisk keeps trying to send the doorbell's RTP to the private network IP which of course fails, and the destination never gets the RTP from the doorbell. We haven't found a solution in Asterisk to this, so are now looking to Kamailio which acts as a load-balancing proxy in front of Asterisk for one. For example, maybe we could use fix_nated_sdp, but only on 200 OK's with recvonly. Has anyone else encountered this, and are there any recommended solutions? Thank you in advance! -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

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Re: [SR-Users] NAT problem with recvonly calls
by Ovidiu Sas 09 Dec '20

09 Dec '20

The doorbell is the caller. ICE support would be required on the callee side (between NATed callee and asterisk). There's no need for ICE support on the doorbell/caller side. -ovidiu On Wed, Dec 9, 2020 at 2:42 AM Alex Balashov <abalashov(a)evaristesys.com> wrote: > > ICE support just doesn't seem to me to be something a SIP doorbell would > have. > > On 12/9/20 2:39 AM, Olle E. Johansson wrote: > > > > > >> On 8 Dec 2020, at 18:55, Richard Fuchs <rfuchs(a)sipwise.com > >> <mailto:rfuchs@sipwise.com>> wrote: > >> > >> Use IPv6 > > > > While I like that proposal, assuming that IPv6 will not have a stateful > > firewall is propably not a good assumption. > > So I suspect that an IPv6 firewall would not let the packets in if there > > is not outbound traffic first. I do like Ovidius > > ICE-based solution and would like to add RTCP - if muxed it would open > > the port, but the likelyhood that it’s > > implemented is propably low. > > > > IPv6 greetings! > > /O :-) > >> > >> Cheers > >> > >> On 07/12/2020 23.01, David Cunningham wrote: > >>> Hello, > >>> > >>> We have a problem with a SIP doorbell device which sends media one > >>> way only, and NAT at the receiving device. > >>> > >>> When the doorbell button is pressed it makes a call to a configured > >>> destination. Since the doorbell only sends and doesn't receive it > >>> sends the INVITE with sendonly in the SDP, and the destination then > >>> replies with a 200 OK with recvonly in the SDP. The problem is that > >>> the destination is behind NAT, and its reply contains a private > >>> network IP in the SDP. > >>> > >>> Normally Asterisk when nat=yes works around that by adjusting the > >>> destination for RTP to be the address it actually receives audio > >>> from, however because this device is recvonly Asterisk never receives > >>> audio from it. This means Asterisk keeps trying to send the > >>> doorbell's RTP to the private network IP which of course fails, and > >>> the destination never gets the RTP from the doorbell. > >>> > >>> We haven't found a solution in Asterisk to this, so are now looking > >>> to Kamailio which acts as a load-balancing proxy in front of Asterisk > >>> for one. For example, maybe we could use fix_nated_sdp, but only on > >>> 200 OK's with recvonly. > >>> > >>> Has anyone else encountered this, and are there any recommended > >>> solutions? > >>> > >>> Thank you in advance! > >>> > >>> -- > >>> David Cunningham, Voisonics Limited > >>> http://voisonics.com/ <http://voisonics.com/> > >>> USA: +1 213 221 1092 > >>> New Zealand: +64 (0)28 2558 3782 > >>> > >>> _______________________________________________ > >>> Kamailio (SER) - Users Mailing List > >>> sr-users(a)lists.kamailio.org > >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >> _______________________________________________ > >> Kamailio (SER) - Users Mailing List > >> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> > >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- VoIP Embedded, Inc. http://www.voipembedded.com

1 0

GitHub discussion
by Sergey Safarov 09 Dec '20

09 Dec '20

GitHub release new feature - discussion. Example https://github.com/vercel/vercel/discussions

2 1

Kamailio as an LCR based on prices but also signaling quality indicators
by Abdoul Osséni 08 Dec '20

08 Dec '20

Hello, I am trying to configure kamailio as an LCR based on prices but also signaling quality indicators like ASR and ACDR. Can you please help me and tell me where to start? Regards Abdoul

4 3

kamailio-5.2.0 Crash - udp_send failed
by Hamid Hashmi 07 Dec '20

07 Dec '20

I am using "kamailio 5.2.0-dev6 (x86_64/linux) 48ed62-dirty" what could be the reason of the crash its usually crash with the following logs Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36eee8c560,418,0,10.100.227.202:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: WARNING: tm [t_reply.c:1581]: t_retransmit_reply(): send pr buffer failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8eb948,358,0,10.100.177.91:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8b4a80,358,0,10.100.72.230:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8ac740,416,0,10.100.72.230:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8b5098,356,0,10.100.56.38:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8e8f18,416,0,10.100.177.91:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: <script>: Re-Transmit: REGISTER (sip:410060269353579@ims.mnc006.mcc410.3gppnetwork.org (10.100.236.57:5060) to sip:410060269353579@ims.mnc006.mcc410.3gppnetwork.org, 1SDq5cuY3iDYFKLATTp) Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8e8a08,354,0,10.100.2.76:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8ad100,360,0,10.100.144.137:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8eb860,360,0,10.100.140.226:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:43 pcscf [Proxy-CSCF][20861]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 77 Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.64.49]:32888 -> Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f36a739fe78 r: 0x7f36a739fef8 Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: ALERT: <core> [main.c:748]: handle_sigs(): child process 20815 exited by a signal 11 Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: ALERT: <core> [main.c:751]: handle_sigs(): core was generated Sep 25 16:58:44 pcscf kernel: [52607827.581845] net_ratelimit: 16 callbacks suppressed Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: CRITICAL: cdp [diameter_peer.c:423]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Sep 25 16:59:01 pcscf CRON[24805]: (root) CMD (/bin/bash /root/volumes/monitoring/src/services_status.sh -p http://10.161.224.25:9091 -s kamailio_pcscf,sems -n pcscf) Sep 25 16:59:01 pcscf CRON[24804]: (CRON) info (No MTA installed, discarding output) Sep 25 16:59:44 pcscf [Proxy-CSCF][20742]: CRITICAL: <core> [main.c:654]: sig_alarm_abort(): shutdown timeout triggered, dying... but last time its also crashed with the following logs as well Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.183.34]:39022 -> Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f92ca8d0150 r: 0x7f92ca8d01d0 Sep 25 18:39:59 pcscf [Proxy-CSCF][28010]: ERROR: cdp [authstatemachine.c:404]: auth_client_statefull_sm_process(): auth_client_statefull_sm_process(): Received invalid event 6 while in state Open! Sep 25 18:39:59 pcscf [Proxy-CSCF][28010]: ERROR: cdp [authstatemachine.c:404]: auth_client_statefull_sm_process(): auth_client_statefull_sm_process(): Received invalid event 6 while in state Open! Sep 25 18:40:00 pcscf [Proxy-CSCF][28034]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 77 Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.100.218]:1337 -> Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f92ca87f070 r: 0x7f92ca87f0f0 Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: ALERT: <core> [main.c:748]: handle_sigs(): child process 27989 exited by a signal 11 Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: ALERT: <core> [main.c:751]: handle_sigs(): core was generated Sep 25 18:40:01 pcscf kernel: [52613904.372973] net_ratelimit: 29 callbacks suppressed Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: CRITICAL: cdp [diameter_peer.c:423]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Sep 25 18:40:01 pcscf CRON[30581]: (root) CMD (/bin/bash /root/volumes/monitoring/src/services_status.sh -p http://10.161.224.25:9091 -s kamailio_pcscf,sems -n pcscf) Sep 25 18:40:01 pcscf CRON[30580]: (CRON) info (No MTA installed, discarding output) Sep 25 18:40:02 pcscf sems[2858]: [#7f2080044700] [update_uac_request, trans_layer.cpp:2080] WARNING: While sending 200 ACK: no matching transaction Sep 25 18:41:01 pcscf [Proxy-CSCF][27919]: CRITICAL: <core> [main.c:654]: sig_alarm_abort(): shutdown timeout triggered, dying... PS: Kamailio is not listening on localhost. Regards Hamid R. Hashmi

3 4

Kamilio dispatcher not registering with asterisk
by Mihai 07 Dec '20

07 Dec '20

Hi all, i have a problem with kamailio 5.3.x and asterisk 13, i basically forward everything, but i cant manage to make it register. The register arrives on asterisk see the capture bellow: Any help is much appreciated! ---------------- REGISTER sip:oro.sips.nl;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 104.201.100.15;branch=z9hG4bK4d5e.997621d931a2ad9f6423e2ac7782a319.1 Via: SIP/2.0/UDP 34.12.55.102:34626 ;received=34.12.55.102;branch=z9hG4bK-524287-1---0a0bffd64bbd9941;rport=34626 Max-Forwards: 69 Contact: <sip:test-user1@34.12.55.102:34626 ;transport=UDP;rinstance=85c33e976be3dad1> To: <sip:test-user1@oro.sips.nl;transport=UDP> From: <sip:test-user1@oro.sips.nl;transport=UDP>;tag=6d8fd914 Call-ID: h6lkViFgxurpVRkw4Ly-GQ.. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.4.9 rv2.10.11.7 Allow-Events: presence, kpml, talk Content-Length: 0 Path: <sip:104.201.100.15;lr;received=sip:34.12.55.102:34626> -------------------the config ------- # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_FS 10 ####### Global Parameters ######### debug=4 log_stderror=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:127.0.0.1:5060 sip_warning=no ####### Modules Section ######## # set module path #mpath="/usr/local/lib/kamailio/modules/" #loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "path.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- #modparam("dispatcher", "db_url", DBURL) #modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_from", "sip:proxy@mykamailio") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshold", 10) modparam("dispatcher", "ds_inactive_threshold", 5) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=488;class=3") modparam("dispatcher", "ds_probing_mode", 1) modparam("path", "use_received", 1) ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); if(ds_is_from_list()) { setflag(FLT_FS); } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); if (isflagset(FLT_FS)) { route(FROM_FS); exit; } ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); } route[FROM_FS] { record_route(); route(RELAY); exit; } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' switch ($fd){ case "oro.sips.nl": send_reply(483, "Ok, Skipper!"); if (!ds_select_dst("1", "4")) { log("ERROR: no active destinations found!\n"); send_reply(503, "Service Unavailable"); exit; } ds_select_dst("1", "4"); log("---- flag value is $dd -- branch $fd \n"); xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); break; case "oro2.sips.nl": send_reply(483, "Ok, Skipper!"); if (!ds_select_dst("2", "4")) { log("ERROR: no active destinations found!\n"); send_reply(503, "Service Unavailable"); exit; } ds_select_dst("2", "4"); log("---- flag value is $dd -- branch $fd \n"); xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); break; default: log("unknow destination?"); send_reply(503, "No service defined"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }

3 4

SLA presence not working proper with (SCA module)
by Aatif Shaikh 07 Dec '20

07 Dec '20

Hello All I configure my kamailio.cfg to handle SLA using (SCA module) and shared presence is working but still facing some issue in case of hold invite and BYE and the reason behind of that issue is following error Nov 26 07:38:40 SBC-4-1 /usr/local/sbin/kamailio[104107]: ERROR: sca [sca_call_info.c:1008]: sca_call_info_invite_request_handler(): Failed to update sip:4569@test.sip.abcd.com appearance-index 0 to active actually, in Kamailio appearances, I can see the entry for this call but the index is 1 instead of 0, and Kamailio going to update with a 0 index [root@SBC-4-1 ntcarfte-kamailio]# kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 active 1606376289 sip:4569@10.xx.xx.xx:5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120MTQ5OTcyZjFhMjMyNmI1ZGE1MWY4ODc2M2RkN2VmZmQ 45dc18d8 H6cS9vecZ248B if I manually update appearances using RPC command with index 1 then the phone started blinking as I change state to held # kamcmd sca.update_appearance sip:4569@sc4test.sip.teledge.com 1 held # kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 held 1606376516 sip:4569@10.xx.xx.xx:5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120NzE5MDgyZDc1NjJiYjcwMWFlYmI3NzM3NjE2OTRhZjU 08f005cb eK9ratm83g92e kamailio.cfg route[SCA] { if(is_method("SUBSCRIBE")) { if ($hdr(Event) == "call-info" || $hdr(Event) == "line-seize") { xlog("L_INFO","(MAIN) :HELLO ($avp(uuid)) : $avp(rsi) $sp $hdr(Event)"); xdbg("SCA: $hdr(Event) SUBSCRIBE $ru from $si:$sp"); sca_handle_subscribe(); exit; } return; } if (!is_method("BYE|CANCEL|INVITE|PRACK|REFER")) { return; } sca_call_info_update(); } route[RELAY_OUTBOUND_FS] { xlog("L_INFO","(RELAY_OUTBOUND_FS) : ($avp(uuid)) : INSIDE ROUTE "); if (!has_totag()){ $avp(set_h) = 1; } t_on_reply("REPLY_OUTBOUND_FS"); route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } route[RELAY_INBOUND_FS] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(s:puburis_caller) = $fu; setflag(8); dlg_manage(); } if(is_method("REGISTER")) { t_on_reply("REPLY_REGISTER"); } else { t_on_reply("REPLY_INBOUND_FS"); } route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[REPLY_INBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } onreply_route[REPLY_OUTBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } can someone help me with this Did I missed something to handle SLA? -- -- Regards, *Aatif Shaikh *| Software Developer +91 9033976268Hangout: aatif.shaikh(a)ecosmob.comSkype : aatif.shaikh(a)ecosmob.com *Ecosmob Technologies Pvt. Ltd. * https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data <https://twitter.com/ecosmob> <https://www.facebook.com/Ecosmob> <https://plus.google.com/+Ecosmob> <https://www.linkedin.com/company/ecosmob> <https://in.pinterest.com/ecosmob/> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

1 0

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 06 Dec '20

06 Dec '20

Thank you Joel for the tip. But that is also not it..... I tried of course, but when you look at the Dialog module documentation is says: modparam("dialog", "profiles_with_value", "caller ; my_profile") So there are spaces in the modparam config for the dialog profiles. Its just so strange that when starting Kamailio I get the error that the Profile outbound is not configured, while it is... Log line: Dec 6 15:04:04 proxy01 /usr/local/sbin/kamailio[76664]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <inbound> not defined kamailo.cfg # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "inbound") Im starting to think that I am missing some configuration in the kamailio.cfg file... Rgds, Gertjan

1 0

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 05 Dec '20

05 Dec '20

Thanks again Sergiu, /Hmmm... if dialog params are wrapped in a ifdef condition, then do you have something like #!define WITH_DIALOG at the top?/ But the #!define WITH_DIALOG is to choose between loading module parameters or not loading module parameters... When you do not add #!ifdef WITH_XXX before the parameters and #!endif after, then those parameters will be loaded when starting Kamailio. So, I still have the issue that the profile is not being loaded. I have tried renaming the profiles, in the route and the parameters, but that does not make a difference.. # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "inbound ; outbound") route[CONCURRENT_IN] { #Add call to customer profile #Check if customer has not reached inbound call limit #Get max concurrent calls sql_query("cc", "select max_calls_in from calllimit where cust_id='$avp(custid)'", "ra"); $avp(maxcalls)=$dbr(ra=>[0,0]); sql_result_free("ra"); $avp(concurrent) = 0; get_profile_size("inbound", "$avp(custid)", "$avp(concurrent)"); if( $avp(concurrent) >= $var(max) ) { xlog("L_INFO"," Call limit reached for customer $avp(custid)\r\n "); sl_send_reply("503", "No Lines Available"); exit; } set_dlg_profile("inbound", "$avp(custid)"); return; } But this still results in: Dec 5 09:34:33 proxy01 /usr/local/sbin/kamailio[75755]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <inbound> not defined Dec 5 09:34:33 proxy01 /usr/local/sbin/kamailio[75755]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1578

2 1

question regarding rtpengine and ICE candidate selection
by Andrew Chen 04 Dec '20

04 Dec '20

Hi all, I was wondering if someone can help me understand how the ICE parameter works in the rtpengine module works. So basically our client does an ICE candidate lookup and grabs a list of them and applies it to the INVITE that gets sent to the Kamailio. The list looks like this: 2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]: a=candidate:3373280875 1 udp 2122262783 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0 network-id 2 network-cost 10 2020-12-03T18:05:46.456057+00:00 ashmainkama51 kamailio[22147]: a=candidate:3040609428 1 udp 2122197247 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 59828 typ host generation 0 network-id 3 network-cost 10 2020-12-03T18:05:46.456081+00:00 ashmainkama51 kamailio[22147]: a=candidate:1681997092 1 udp 2122129151 192.168.1.66 59829 typ host generation 0 network-id 1 network-cost 10 2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]: a=candidate:2274611867 1 tcp 1518283007 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active generation 0 network-id 2 network-cost 10 2020-12-03T18:05:46.456131+00:00 ashmainkama51 kamailio[22147]: a=candidate:4223662180 1 tcp 1518217471 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 9 typ host tcptype active generation 0 network-id 3 network-cost 10 2020-12-03T18:05:46.456155+00:00 ashmainkama51 kamailio[22147]: a=candidate:717406676 1 tcp 1518149375 192.168.1.66 9 typ host tcptype active generation 0 network-id 1 network-cost 10 2020-12-03T18:05:46.456180+00:00 ashmainkama51 kamailio[22147]: a=candidate:2130547417 1 udp 8199935 206.81.191.27 61165 typ relay raddr 85.247.0.121 rport 53091 generation 0 network-id 1 network-cost 10 In the rtpengine_offer, I have this: rtpengine_offer("ICE=force DTLS=passive replace-session-connection replace-origin external internal"); What i am trying to do is to tell the freeswitch endpoint to use rtpengine as ICE candidate and I see in the SDP this is happening: a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host generation 0 Btw..206.81.191.52 is a separate AWS instance running rtpengine binary. On the rtpengine_answer, I tell the remote client side the same thing..use rtpengine as your ICE candidate: rtpengine_answer("ICE=force DTLS=passive replace-session-connection replace-origin internal external"); and this is snippet from 200 OK: a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host generation 0 Question: Why does the rtpengine logs still show that it's trying to use 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 as RTP candidate in this scenario? Dec 3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG: [ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint: 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0) I thought ICE=force will handle such adjustments so that all rtp is handled by the rtpengine to our client and bypass the STUN server? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 22

REFER from MS teams to Kamailio.
by Yuriy Nasida 04 Dec '20

04 Dec '20

Well, I saw similar questions in the list already but looks like nobody has answer. Please look at REFER below. Kamilio get REFER from MS and sends it to FS node. Next, FS node try to make 3th call for some reason.I expect that FS will not do 3th call and just will connect Alice and Bob itself. 2020/05/14 12:32:00.637027 KAM_IP:5060 -> FS_IP:5060 REFER sip:Alice_number@FS_IP:5060;transport=udp SIP/2.0 FROM: Customer1<sip:MS_TRUNK_NUMBER@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=a860c50a3fb54d08b4e5740fa2dfb3d6 TO: <sip:Alice_number@FQDN_OF_TRUNK:5061>;user=phone;tag=e8ct9S6ty13va CSEQ: 4 REFER CALL-ID: 2c71b2a6669b5343a231e1244b19c945 MAX-FORWARDS: 50 Via: SIP/2.0/UDP FQDN_OF_TRUNK:5060;branch=z9hG4bK10ae.2c42897feca117121a23bf0c8d54cd19.0;i=c VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK7e3e8998 CONTACT: <sip:api-du-a-euwe.pstnhub.microsoft.com:443 ;x-i=6b68e7aa-f5e2-44ec-9edf-0bacbabfce07;x-c=2c71b2a6669b5343a231e1244b19c945/d/8/b68f86794a8e44d19543f8edbee6b2fc CONTENT-LENGTH: 0 REFER-TO: <sip:Bob_number@sip.pstnhub.microsoft.com:5061 ;user=phone;transport=tls> REFERRED-BY: <sip:sip.pstnhub.microsoft.com:5061 ;x-m=8:orgid:21bc47d3-c050-4292-8234-46f7005b97aa;x-t=fb788ef8-3c4c-455a-8d62-f3c20832c0d3;x-ti=6b68e7aa-f5e2-44ec-9edf- acbabfce07;x-tt=aHR0cHM6Ly9hcGktZHUtYS1ldXdlLnBzdG5odWIubWljcm9zb2Z0LmNvbS92MS9uZ2MvY2FsbG5vdGlmaWNhdGlvbj9kY2k9YzIxMjE3MzEyNTQ2NDk1ZjlhYTcwODliYTkwNGIxZGQ%3D> USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.5.6.2 i.EUWE.4 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY P-ASSERTED-IDENTITY: <tel:MS_TRUNK_NUMBER>,< sip:customer1@m365x587912.onmicrosoft.com> PRIVACY: id X-AUTH-IP: 52.114.75.24 X-AUTH-PORT: 3136 Any advice?

4 11

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 04 Dec '20

04 Dec '20

Hello Sergiu, Thank you for the pointer, the line 1576 is: get_profile_size("callee", "$avp(custid)", "$avp(concurrent)"); That is where Kamailio refuses to start.... And as you mentioned my error in the IF statement after correction you pointed out, Kamailio still refuses on the fact that the profile <callee> was not defined... Dec 4 13:42:50 proxy01 /usr/local/sbin/kamailio[75008]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <callee> not defined Dec 4 13:42:50 proxy01 /usr/local/sbin/kamailio[75008]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1576 Dec 4 13:42:50 proxy01 systemd[1]: kamailio.service: Main process exited, code=exited, status=255/n/a Dec 4 13:42:50 proxy01 systemd[1]: kamailio.service: Failed with result 'exit-code'. Does anyone else maybe have an idea on where I go wrong? Rgds, Gertjan

2 1

Fix Sdp Error
by Duarte Rocha 04 Dec '20

04 Dec '20

Hello, I noticed that when i do calls behind NAT i'm getting this error when i call fix_nated_sdp() : "nathelper [nathelper.c:1659]: replace_sdp_ip(): can't extract 'a=rtcp' IP from the SDP" As far as i can see, all the ips on SDP are correctly replaced to a public IP, so i don't understand the origin of this error. This is the SDP received by Kamailio : Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3421914885 1 IN IP4 192.168.24.89 Session Name (s): SIPPER for PhonerLite Connection Information (c): IN IP4 192.168.24.89 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101 Media Attribute (a): rtpmap:107 opus/48000/2 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:110 speex/8000 Media Attribute (a): rtpmap:111 speex/16000 Media Attribute (a): rtpmap:9 G722/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:11 L16/44100 Media Attribute (a): rtpmap:118 L16/16000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ssrc:2716178988 Media Attribute (a): sendrecv And this is the SDP sent by Kamailio after NAT being treated : Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3421914885 1 IN IP4 1.1.1.1 (Public IP) Session Name (s): SIPPER for PhonerLite Connection Information (c): IN IP4 1.1.1.1 (Public IP) Time Description, active time (t): 0 0 Media Description, name and address (m): audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101 Media Attribute (a): rtpmap:107 opus/48000/2 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:110 speex/8000 Media Attribute (a): rtpmap:111 speex/16000 Media Attribute (a): rtpmap:9 G722/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:11 L16/44100 Media Attribute (a): rtpmap:118 L16/16000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ssrc:2716178988 Media Attribute (a): sendrecv Media Attribute (a): oldmediaip:192.168.24.89 Media Attribute (a): oldmediaip:192.168.24.89 Is there something failing in the SDP fix? Best Regards,

1 0

TOPOS replication with KDMQ
by James Browne 04 Dec '20

04 Dec '20

Has anyone ever implemented DMQ replication of TOPOS data between kamailio servers? I'm trying to find the easiest (to maintain) way to implement this replication, possibly without using an external database. James

3 3

presence not work proper with subs_db_mode = 2
by Aatif Shaikh 04 Dec '20

04 Dec '20

Hello, all have a problem with Kamailio presence, I have configured my kamailio.cfg to handle dialog presence and facing issue in case of subs_db_mode = 2 in case of mode 2 facing the following presence issue. 1. after hangup presence is still up 2. sometimes stuck in a ringing state 3. sometimes presence not came..etc but in the case of subs_db_mode = 3 (DB ONLY) mode presence working fine not facing any issue with this mode. I noticed that sometimes Kamailio will throw the following message during PUBLISH processing and it will not generate appropriate NOTIFY I also verify inactive watcher subscription is there for that user and also a proper entry in the dialog table. MSG during PUBLISH processesing and it will not generate appropriate NOTIFY Oct 30 07:42:10 NC-HOSTED-1 /usr/local/sbin/kamailio[2409]: DEBUG: presence [notify.c:1444]: publ_notify(): Could not find subs_dialog kamailio.cfg param modparam("pua", "db_url", DBURL_KAMAILIO) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 100) modparam("pua", "dlginfo_increase_version", 1) modparam("pua", "default_expires", 3600) modparam("pua", "fetch_rows", 1000) modparam("pua", "outbound_proxy", SERVER_IP) modparam("dialog", "db_url", DBURL_KAMAILIO ) modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 260) modparam("dialog", "db_fetch_rows", 500) modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", 8) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "use_pubruri_avps", 1) modparam("pua_dialoginfo", "include_localremote", 1) modparam("pua_dialoginfo", "override_lifetime", 300) modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") modparam("pua_dialoginfo", "pubruri_caller_dlg_var", "pubruri_caller") modparam("pua_dialoginfo", "pubruri_callee_dlg_var", "pubruri_callee") modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence", "db_table_lock_type", 0) modparam("presence", "db_url", DBURL_KAMAILIO) modparam("presence", "db_update_period", 5) modparam("presence", "send_fast_notify", 1) modparam("presence", "clean_period", 100) modparam("presence", "subs_db_mode", 2) //modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 1) //new/ //modparam("presence", "notifier_processes", 2) //new// modparam("presence", "notifier_poll_rate", 20) //new// modparam("presence", "waitn_time",1) modparam("presence", "fetch_rows", 1000) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "max_expires", 3600) modparam("presence_xml", "db_url", DBURL_KAMAILIO) modparam("presence_xml", "force_active", 1) logic for presence request_route { if(is_method("PUBLISH") || is_method("SUBSCRIBE")) { route(HANDLE_PRESENCE); } } route[HANDLE_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; } route[RELAY] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(t_ext) = $tU ; $avp(domain) = $fd ; $avp(et) = $_s(sip:$avp(t_ext)@$avp(domain)) ; $avp(s:puburis_callee) = $avp(et); setflag(8); dlg_manage(); } } I already report this issue in Kamailio git https://github.com/kamailio/kamailio/issues/2540 -- -- Regards, *Aatif Shaikh *| Software Developer +91 9033976268Hangout: aatif.shaikh(a)ecosmob.comSkype : aatif.shaikh(a)ecosmob.com *Ecosmob Technologies Pvt. Ltd. * https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data <https://twitter.com/ecosmob> <https://www.facebook.com/Ecosmob> <https://plus.google.com/+Ecosmob> <https://www.linkedin.com/company/ecosmob> <https://in.pinterest.com/ecosmob/> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

1 0

Dialog profile not defined
by Gertjan Wolzak 03 Dec '20

03 Dec '20

Hello Kamailions, I am running into the following issue. The same setup I had working on 5.0, now I am using a 5.4 version of Kamailio, but I do not believe it has to do with the version, more with my ability to make errors... I am trying to limit incoming calls by using the dialog module. The max concurrent calls value I retrieve from a database. But when I try to start Kamailio it will not because the dialog profile is not defined... These are my configured Dialog parameters: # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "callee") My check on the concurrent inbound calls: route[CONCURRENT_IN] { #Add call to callee profile #Check if callee has not reached inbound call limit #Get max concurrent calls sql_query("cc", "select max_calls_in from calllimit where cust_id='$avp(custid)'", "ra"); $avp(maxcalls)=$dbr(ra=>[0,0]); sql_result_free("ra"); $avp(concurrent) = 0; get_profile_size("callee", "$avp(custid)", "$avp(concurrent)"); if( $avp(concurrent) = $var(max) ) { xlog("L_INFO"," Call limit reached for customer $avp(custid)\r\n "); sl_send_reply("503", "No Lines Available"); exit; } set_dlg_profile("callee", "$avp(custid)"); return; } And the kamailio log error: Dec 3 09:12:08 proxy01 kamailio[74102]: INFO: <core> [main.c:2833]: main(): processes (at least): 21 - shm size: 67108864 - pkg size: 8388608 Dec 3 09:12:08 proxy01 kamailio[74102]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <callee> not defined Dec 3 09:12:08 proxy01 kamailio[74102]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1576 Is someone able to see where I go wrong and point me in the right direction? Thank you. Rgds, Gertjan Wolzak

2 2

$TF pseudo variable can not be used
by Володимир Іванець 03 Dec '20

03 Dec '20

Hello! I'm using a $TF variable in my config for some logging purposes. It was providing the correct string with Kamailio version 5.3.8 on CentOS 6 and 8. After updating to Kamailio version 5.4.1, the variable started to show something else. I also tried versions 5.4.2 and development 5.5.0 on CentOS 6 and the problem still persists. With version 5.5.0_dev3 this variable shows "pG▒" for me. I was wondering if someone else is using it too with Kamailio version 5.4.1 or newer on different platforms and has this issue. Thanks!

2 2

kamailio51 installs 5.2.1?
by David Villasmil 03 Dec '20

03 Dec '20

Hello everyone, On a debian buster i added the repo deb http://deb.kamailio.org/kamailio51 buster main deb-src http://deb.kamailio.org/kamailio51 buster main But when i install i get 5.2, is this right? additional info: # apt-cache showpkg kamailio Package: kamailio Versions: 5.2.1-1 (/var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_binary-i386_Packages) (/var/lib/dpkg/status) Description Language: File: /var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_binary-i386_Packages MD5: df3e15f422439e08c305782f5650a98c Description Language: en File: /var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_i18n_Translation-en MD5: df3e15f422439e08c305782f5650a98c 5.1.10.1+bpo10 (/var/lib/apt/lists/deb.kamailio.org_kamailio51_dists_buster_main_binary-i386_Packages) Description Language: File: /var/lib/apt/lists/deb.kamailio.org_kamailio51_dists_buster_main_binary-i386_Packages MD5: ebddf40d0dfbfde1c479419970f978c7 Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

kamailio not absorbing ACK to CANCEL when behind a Path proxy
by Anthony Alba 02 Dec '20

02 Dec '20

Folks, I have a weird situation, when I enable Path header, kamailio behind a Path proxy, that kamailio doesn't recognise the ACK to CANCEL and tries to forward it. Test scenario: - voice.example.com is a stateless proxy with TLS/UDP bridging to kamailio. - on the UDP leg to kamailio it will add a Path: header to REGISTER UA1 = david is calling UA2 = charles and UA2 is sending 603 1 . UA1: preloaded route set (outbound proxy) # this is the Path proxy Route: sip:voice.example.com;transport=tls;lr 2. UA2: sends CANCEL 603 - kamailio immediately sends its (per-hop?) ACK - kamailio forwards CANCEL to UA1 kamailio sent this (per-hop?) ACK - the Route header is the internal UDP interface of the Path proxy ACK sip:charles@192.168.1.7:37309;transport=TCP;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.122.99:5064;branch=z9hG4bK735d.6b27c8d09e40462ef47eccf90eb10823.0 Max-Forwards: 70 From: sip:david@voice.example.com;tag=i1acAHsitf2gcN9uHH-ZSyJ5OdT8O5M0 To: <sip:charles@voice.example.com>;tag=LmUY79rIdBQiiSakML4F0lHl271VdUn8 Call-ID: qM3W-fMrzPsvcjRSD1okEp5nbgHCF5Hu CSeq: 16667 ACK Route: <sip:192.168.122.100;lr> Content-Length: 0 3. UA1 sends ACK kamailio sees this - top Via is path proxy - 2nd Via is UA (caller) - CSeq correctly matches the INVITE - kamailio does not recognise this ACK from UA1(caller), tries to forward it, and also resends 603 to UA1(caller) 3 times ACK sip:charles@voice.example.com SIP/2.0 Via: SIP/2.0/UDP 192.168.122.100:5060;branch=z9hG4bKb3c9a7a3b6ec9e78d0144cda709f7047 Via: SIP/2.0/TLS 192.168.1.17:44925;rport=44925;branch=z9hG4bKPjFM-9gpaJEghJvN19TjnvScCwCoAGzwEc;alias;received=192.168.1.17 Max-Forwards: 70 From: sip:david@voice.example.com;tag=i1acAHsitf2gcN9uHH-ZSyJ5OdT8O5M0 To: sip:charles@voice.example.com;tag=LmUY79rIdBQiiSakML4F0lHl271VdUn8 Call-ID: qM3W-fMrzPsvcjRSD1okEp5nbgHCF5Hu CSeq: 16667 ACK Route: <sip:voice.example.com;transport=tls;lr> Content-Length: 0 Now kamailio should absorb this ACK, as it has already sent its per-hop ACK. Instead I see ERROR: <core> [core/forward.c:541]: forward_request(): cannot forward to af 2, proto 3 no corresponding listening socket I don't expect kamailio to forward anything at this point. The route block is route[WITHINDLG] and it calls t_relay() normally. route[WITHINDLG] { if (!has_totag()) return; # 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 statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); # we should absorb this ACK, no? exit; } 4. If I remove the Path proxy and kamailio is the TLS proxy at voice.example.com the call flow works perfectly. The ACK from caller (david) is absorbed. 5. Regular calls, when UA2 accepts the call, work fine. In-dialog requests for BYE, from both UAs, work correctly with this Path proxy. Each UA has the complete 5 element route set( 2 x Path proxy + kamailio + 2 x Path proxy). The working route set looks like this: BYE sip:david@192.168.1.17:44925;transport=TCP;ob SIP/2.0 Via: SIP/2.0/TLS 192.168.1.7:37309;rport;branch=z9hG4bKPjHdUtaUL5AfRcV-SX1Puya4niJKOOQ4VQ;alias Max-Forwards: 70 From: <sip:charles@voice.example.com>;tag=6zOt.H7w8NL5njV7klwGLlSJy.U5yBC9 To: <sip:david@voice.example.com>;tag=Pa0ORUxA-7lK0BnR4pLJ5n8rn2OvQrJy Call-ID: 3gWy4Q7O.-zQ-I07lRKHyFyX6azyCSwG CSeq: 12855 BYE Route: <sip:voice.example.com;transport=tls;lr;r2=on> Route: <sip:192.168.122.100;lr;r2=on> Route: <sip:192.168.122.99:5064;lr> Route: <sip:192.168.122.100;lr;r2=on> Route: <sip:voice.example.com;transport=tls;lr;r2=on> User-Agent: basesip 1.0.0 Content-Length: 0 Any ideas? Anthony Alba

1 0

android client SIP TLS client autentication + kamailio
by beer Ll 02 Dec '20

02 Dec '20

Hello Has anybody use a android SIP client with TLS client authentication enabled with kamailio? Do you have suggestion ? Many thanks Luca

3 2

Kamailio 5.4 ndb_redis, selecting a database
by Daniel Donoghue 02 Dec '20

02 Dec '20

Hello, I note that I can use the ndb_redis server config parameter to specify connection details for my redis servers; I also note that the connection string can specify a “db” parameter which can be set from 0 to 9, with the default being 0. I am looking for a way to dynamically set the db parameter. I do not which to add the same server with different names N times just to select a different db and having N times more connections to it than necessary. It looks as though I can call redis_cmd(“svrN”, “SELECT dbN”, “init”); but will that propagate properly and set the correct db for all connections to my redis server for the lifetime of the request? Ideally, I’d like to be able to specify an avp for the db, or have a function to set the db per request. With every blessing, — Daniel Donoghue

1 0

Kamailio Remove Remove "Allow" features from header
by me.projects＠yahoo.com 02 Dec '20

02 Dec '20

Hi, I want to remove some "Allow" features from my Kamailio SBC like I want to keep following only Allow: OPTIONS, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, How can I achieve that? Thanks,

2 1

How to DIY/Setup An Open Source IP PBX Appliance/Server?
by Turritopsis Dohrnii Teo En Ming 01 Dec '20

01 Dec '20

Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server? Good day from Singapore, After reading recent reviews, I gather that Asterisk is the gold standard when it comes to open source VoIP systems and it is the most famous open source PBX out there. Article: Compare the Top 10 Best Open Source PBX Software of 2020 Link: https://www.voipreview.org/business-voip/best-open-source-pbx-software Article: Top 10 Free Open Source PBX Software Solutions Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/ The following is an excerpt from Wikipedia: "Asterisk is a core component in many commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software with an open-source distribution model. AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software to realize all telephony functions. AstLinux is a "Network Appliance for Communications" open-source software distribution.[15] FreePBX, an open-source graphical user interface, bundles Asterisk as the core of its FreePBX Distro[16] LinuxMCE bundles Asterisk to provide telephony; there is also an embedded version of Asterisk for OpenWrt routers. PBX in a Flash/Incredible PBX and trixbox are software PBXes based on Asterisk. Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer PBX, fax, instant messaging and email functions, respectively, before switching to 3CX. Issabel is an open-source Unified Communications software which uses Asterisk for telephony functions. It was forked from the open-source versions of Elastix when 3CX acquired it. *astTECS uses Asterisk in its VoIP and mobile gateways." Link: https://en.wikipedia.org/wiki/Asterisk_(PBX) I would like to DIY/setup an IP PBX appliance/server using free open source projects. Which free open source project, mentioned in the list and links above, would you recommend to DIY my IP PBX appliance/server? Should I buy a desktop computer or get one of those appliances listed in the link below to serve as my IP PBX appliance/server? Link: https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celero… Please also refer me to very good, detailed and well explained guides/tutorials/manuals on setting up open source IP PBX appliances/servers. Lastly, please recommend a cheap and affordable IP phone (suggest brand and model) to go along with my DIY open source IP PBX appliance/server. Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 Tuesday, is a TARGETED INDIVIDUAL (TI) living in Singapore. Thank you very much. -----BEGIN EMAIL SIGNATURE----- The Gospel for all Targeted Individuals (TIs): [The New York Times] Microwave Weapons Are Prime Suspect in Ills of U.S. Embassy Workers Link: https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html ******************************************************************************************** Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and Australia (25 Dec 2019 to 9 Jan 2020): [1] https://tdtemcerts.wordpress.com/ [2] https://tdtemcerts.blogspot.sg/ [3] https://www.scribd.com/user/270125049/Teo-En-Ming -----END EMAIL SIGNATURE-----

3 2

How to msg_apply_changes in a branch route?
by Anthony Alba 01 Dec '20

01 Dec '20

Is there a way to obtain the effect of msg_apply_changes in a branch route? I want to do: route[BRANCHMANAGE] { rtpengine_manage() msg_apply_changes() // further mangle the SDP with textops for // obtuse UAs } The reason for this, is that after forking, I have some UAs that are extremely picky about SDP, and I need to mangle the body to make them happy. (4000 char limit, 68 attribute limit on SDP body - ever seen that? Compared with the absolutely gigantic OFFERs from WebRTC signalling libraries...) Now rtpengine_manage() and textops are working on separate copies of the msg body and the results don't stack correctly. Cheers Anthony Alba

2 1

vscode-kamailio-syntax - issue with empty comment line
by Chaigneau, Nicolas 01 Dec '20

01 Dec '20

Thanks for the link. One issue I've noticed: If you have an empty comment line (just # on a single line), then the next line is wrongly highlighted as comments. For example: # loadmodule "db_postgres.so" > i'm using https://github.com/miconda/vscode-kamailio-syntax in VScode. > its great! This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 4

Re: [SR-Users] [sr-dev] kamailio SIP and RTP proxy
by Henning Westerholt 01 Dec '20

01 Dec '20

Hi Patrick, better to contact our sr-users list with the usage related questions, added to CC. Have a look to the SDP of the SIP packets to see if it contains the correct IP would be one idea to debug this further. Feel free to ask again on sr-users after you have got more details. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-dev <sr-dev-bounces(a)lists.kamailio.org> On Behalf Of Patrick Leybag Sent: Wednesday, November 25, 2020 6:26 AM To: sr-dev(a)lists.kamailio.org Subject: [sr-dev] kamailio SIP and RTP proxy Hi, Can someone help me? I self host a kamailio using my raspberry pi as a load balancer for my two asterisk servers and get a did number. when I call to my DID number it points to my kamailio and kamailio will distribute to asterisk server but the call has no audio. I tried port forwarding ports 5060 for SIP and 10000-20000 for RTP but it still does not work. Any help is much appreciated. Thank you in advance

1 0

Kamailio Failover
by Rolly Casuga Jr. 01 Dec '20

01 Dec '20

Hi, I am having a problem on fail overing my master kamailio server and slave kamailio server. I use kamailio as a load balancer for my asterisk servers. And it is working with calls, but when I added failover with keepalived to the slave, there is no audio during calls. Thank you, Rolly

2 1

Kamailio behind HAProxy
by Joey Golan 01 Dec '20

01 Dec '20

Hello, I have a kamailio server running behind HAProxy with proxy protocol v2 enabled. In Kamailio I have set the parameter tcp_accept_haproxy=yes and loaded tcpops module. UEs are registered using TLS and kamailio sees that the message has received from their real ip address + port and not HAProxy ip + port. When UE A calls UE B, kamailio is trying to reach UE B using his real ip address and port instead of HAProxy IP address + port. I know I can get the tcp ip and port of HAProxy using $tcp(c_si) and $tcp(c_sp) but I can’t make it work. What is the right way to do this? How should I use these variables properly in order to establish the call successfully? Thanks, Joey.

6 14

core dump error when running under docker
by Carlos Vicente 30 Nov '20

30 Nov '20

Greetings, I'm having a "fun" time trying to run Kamailio inside a Docker container. It keeps failing with: 0(238) DEBUG: <core> [core/mem/pkg.c:108]: pkg_print_manager(): pkg - using memory manager: q_malloc 0(238) DEBUG: <core> [core/mem/shm.c:300]: shm_print_manager(): shm - using memory manager: q_malloc Illegal instruction (core dumped) Unfortunately the core dump is nowhere to be found. I have tried to follow the instructions at: https://www.kamailio.org/wiki/tutorials/troubleshooting/coredumpfile but that hasn't helped. The environment is as follows: * Host hardware: Raspberry Pi 4 (4G memory) * Host OS: Ubuntu 20.04.1 LTS * Docker version 19.03.13 * Docker base image: arm32v7/ubuntu root@2ef8a21534fa:/# kamailio -v version: kamailio 5.3.2 (arm/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT-NOSMP, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, 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 with gcc 9.2.1 The config file passes the syntax check: kamailio -c -f /etc/kamailio/kamailio.cfg and, the most interesting part, if I run Kamailio on the host machine, _with the exact same config file_, it runs fine. I'd appreciate any clues you may have. Thanks. cv

4 10

What is the lifecycle of Branch Flag (setbflag)
by me.projects＠yahoo.com 30 Nov '20

30 Nov '20

Hi, What is the lifecycle of Branch Flag? If its set in Request Route will it be available till end of Dialog or It will be available during Transaction only ? Thanks

3 5

[Mod: CDP] TCP Zero Window Error
by Hamid Hashmi 27 Nov '20

27 Nov '20

Dear Users I have been facing a diameter peer disconnection issue between I-CSCF and HSS. after taking traces on both sides, I saw a lot of TCP Zero Window packets from HSS towards I-CSCF and after that HSS disconnects with the following errors in the log. Nov 26 20:30:25 PROD-HSS-STANDBY [HSS][16401]: ERROR: cdp [peerstatemachine.c:157]: sm_process(): sm_process(): In state Closed invalid event Timeout Nov 26 20:31:25 PROD-HSS-STANDBY [HSS][16401]: ERROR: cdp [peerstatemachine.c:157]: sm_process(): sm_process(): In state Closed invalid event Timeout Nov 26 20:32:25 PROD-HSS-STANDBY [HSS][16401]: ERROR: cdp [peerstatemachine.c:157]: sm_process(): sm_process(): In state Closed invalid event Timeout Nov 26 20:33:04 PROD-HSS-STANDBY systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Nov 26 20:33:04 PROD-HSS-STANDBY kamailio_hss[18098]: Stopping Kamailio SIP Server: kamailio_hss:. Nov 26 20:33:04 PROD-HSS-STANDBY systemd[1]: kamailio_hss.service: Succeeded. Nov 26 20:33:04 PROD-HSS-STANDBY systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. Nov 26 20:33:04 PROD-HSS-STANDBY [HSS][16274]: CRITICAL: cdp [diameter_peer.c:423]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Nov 26 20:33:07 PROD-HSS-STANDBY systemd[1]: Starting LSB: Start the Kamailio SIP proxy server... Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: Starting Kamailio SIP Server: kamailio_hss:loading modules under config path: /usr/local/lib64/kamailio/modules/:/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/local/lib/kamailio/modules/ Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: Listening on Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: udp: 10.161.224.81 [10.161.224.81]:8080 Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: tcp: 127.0.0.1 [127.0.0.1]:5090 Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: tcp: 10.161.224.81 [10.161.224.81]:8080 Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: Aliases: Nov 26 20:33:07 PROD-HSS-STANDBY [HSS][18155]: WARNING: sl [../../modules/tm/tm_load.h:149]: load_tm_api(): Cannot import load_tm function from tm module Nov 26 20:33:07 PROD-HSS-STANDBY [HSS][18155]: WARNING: sl [../../modules/tm/tm_load.h:149]: load_tm_api(): Cannot import load_tm function from tm module Nov 26 20:33:07 PROD-HSS-STANDBY [HSS][18281]: WARNING: cdp [tcp_accept.c:120]: create_socket(): create_socket(): Trying to open/bind/listen on 10.161.224.81 port 3868 Nov 26 20:33:07 PROD-HSS-STANDBY [HSS][18281]: WARNING: cdp [tcp_accept.c:145]: create_socket(): create_socket(): Successful socket open/bind/listen on 10.161.224.81 port 3868 Nov 26 20:33:07 PROD-HSS-STANDBY kamailio_hss[18138]: . Nov 26 20:33:07 PROD-HSS-STANDBY systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Nov 26 20:38:44 PROD-HSS-STANDBY [HSS][18282]: WARNING: cdp [peermanager.c:310]: peer_timer(): Inactivity on peer [icscfcdp2.ims.mnc006.mcc410.3gppnetwork.org] and no DWA, Closing peer... Nov 26 20:39:01 PROD-HSS-STANDBY systemd[1]: Starting Clean php session files... Nov 26 20:39:01 PROD-HSS-STANDBY systemd[1]: phpsessionclean.service: Succeeded. Nov 26 20:39:01 PROD-HSS-STANDBY systemd[1]: Started Clean php session files. Nov 26 20:39:20 PROD-HSS-STANDBY [HSS][18282]: ERROR: cdp [peerstatemachine.c:157]: sm_process(): sm_process(): In state Closed invalid event Timeout I am using the following Kamailio version Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster Version: kamailio 5.2.8 (x86_64/linux) 5a7f5b Default config: /usr/local/etc/kamailio/kamailio.cfg Default paths to modules: /usr/local/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, 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 MAX_RECV_BUFFER_SIZE=262144 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: 5a7f5b Compiled with: gcc 8.3.0 Compiled on: 04:34:39 Nov 19 2020 How I can resolve TCP Zero Window Error? I have already tried increasing CDP worker threads, TCP read and write buffers etc. Regards Hamid R. Hashmi

2 1

handling of locally generated 478 errors
by Henning Westerholt 27 Nov '20

27 Nov '20

Hello, I want to ask for your opinion on the best approach regarding the handling of locally generated 478 errors. To give an example, like the ones generated from TM during t_relay() on an unresolvable destination. Nov 25 17:40:13 kamailio[19345]: ERROR: {28607414 INVITE bba500ac-a9df-1239-6693-00505682c04d} tm [ut.h:286]: uri2dst2(): failed to resolve "invalid.skalatan.de" :unresolvable A or AAAA request (-7) Nov 25 17:40:13 kamailio[19345]: ERROR: {28607414 INVITE bba500ac-a9df-1239-6693-00505682c04d} tm [t_fwd.c:1738]: t_forward_nonack(): failure to add branches Nov 25 17:40:13 kamailio[19345]: CRITICAL: {28607414 INVITE bba500ac-a9df-1239-6693-00505682c04d} rtpengine [../../core/parser/../ip_addr.h:658]: ip_addr2sbuf(): unknown address family 0 These errors will not show up in onreply or failure_route. A long time ago this was discussed on the list [1], as some functionality were phased out out that support these scenarios. Kamailio will try to generate a 478 with TM, this will obviously fail as well, and then generate a 478 with SL. Question 1) Is this intentional that the internally generated 478 is not showing up in the failure_route, like for for 408? This has been tested several times, but it is a complicated configuration. Question 2) Are there any other (better) ideas how to handle that besides using a "event_route[sl:local-response]" to catch this, e.g. to tear down otherwise stale rtpengine sessions etc..? As a side note, event_route[tm:local-response] seems not to work as well because of the tm failure. Thanks, Henning [1] https://lists.kamailio.org/pipermail/sr-users/2011-June/069020.html -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

2 4

Setup IMS solution with Kamailio
by Abidi Adnen 27 Nov '20

27 Nov '20

Greetings , i am around to setup an IMS solution based on your Kamailio solution however i cant find a clear guide to dit properly. I am seeking by this mail you guide to could achieve my setup Thanks and BR, Adnen

2 1

issues using evapi module
by David Escartin 26 Nov '20

26 Nov '20

Dear all We are trying to use the evapi module to send some data to an external application but I'm having problems getting the clients connected. I have the kamailio (version 5.3) running with a tcp socket 127.0.0.1:8228, and the evapi params are just modparam("evapi", "workers", 4) modparam("evapi", "netstring_format", 0) modparam("evapi", "bind_addr", "127.0.0.1:8448") modparam("evapi", "max_clients", 32) I tried a different number of workers and netstring_format 1 too. When I start the kamailio i added some debug to the code, and seems when doing the mod init of the evapi dispatcher 38(4779) DEBUG: <core> [core/sr_module.c:779]: init_mod_child(): idx 38 rank -2: evapi [EvAPI Dispatcher] it reaches to while(1) { ev_loop (loop, 0); } at evapi_run_dispatcher function. I guess if I connected to the tcp socket and sent some event, I would see the client accepted and the event route evapi:connection-new would be triggered. But i'm not able to do that. I tried to use the prime option, a tcp input client connection from logstash, so i could relay the data to the logstash using the evapi relay, but i only see the tcp socket being created but no client accepted. I also tried to connect with an erlang gen_tcp client, but it's the same i only see 47(4798) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 127.0.0.1 47(4798) DEBUG: <core> [core/tcp_main.c:1174]: tcpconn_new(): on port 54537, type 2, socket 105 47(4798) DEBUG: <core> [core/tcp_main.c:1497]: tcpconn_add(): hashes: 1117:1187:1505, 1 47(4798) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xad0880, 105, 2, 0x7fc211712d58), fd_no=53 and if i try to send any data 47(4798) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xad0880, 105, -1, 0x0) fd_no=54 called 47(4798) DEBUG: <core> [core/tcp_main.c:4456]: handle_tcpconn_ev(): sending to child, events 1 47(4798) DEBUG: <core> [core/tcp_main.c:4129]: send2child(): selected tcp worker idx:0 proc:43 pid:4791 for activity on [tcp:127.0.0.1:8448], 0x7fc211712d58 43(4791) DEBUG: <core> [core/tcp_read.c:1749]: handle_io(): received n=8 con=0x7fc211712d58, fd=39 43(4791) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xb3c720, 39, 2, 0x7fc211712d58), fd_no=1 43(4791) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xb3c720, 39, -1, 0x10) fd_no=2 called 43(4791) DEBUG: <core> [core/tcp_read.c:1671]: release_tcpconn(): releasing con 0x7fc211712d58, state 1, fd=39, id=1 ([127.0.0.1]:54537 -> [127.0.0.1]:8448) 43(4791) DEBUG: <core> [core/tcp_read.c:1672]: release_tcpconn(): extra_data (nil) 47(4798) DEBUG: <core> [core/tcp_main.c:3559]: handle_tcp_child(): reader response= 7fc211712d58, 1 from 0 47(4798) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xad0880, 105, 2, 0x7fc211712d58), fd_no=53 47(4798) DEBUG: <core> [core/tcp_main.c:3686]: handle_tcp_child(): CONN_RELEASE 0x7fc211712d58 refcnt= 1 and when i try to send any data 38(10867) DEBUG: evapi [evapi_dispatch.c:610]: evapi_recv_notify(): received [0x7f17d23fc628] [{"test" : "1.1.1.1", "uuid" : "1-31629(a)3.3.3.3" , "pdd" : "4"}] (75) 38(10867) DEBUG: evapi [evapi_dispatch.c:316]: evapi_dispatch_notify(): the message was sent to 0 clients I don't know what i'm missing, or if i'm understanding the use of the module correctly could you please take a look? thanks a lot David -- [image: Logo] David Escartín Almudévar VoIP/Switch Engineer descartin(a)sonoc.io *SONOC* C/ Josefa Amar y Borbón, 10, 4ª · 50001 Zaragoza, España Tlf: +34 917019888 · www.sonoc.io

2 5

Location Fields Address vs Received
by Daniel Greenwald 25 Nov '20

25 Nov '20

On a usrloc record there is an Address field and Received field. The Received field is sometimes not set. What causes it to be not set and what is the difference between the two fields. Sorry if this is a dumb question, I looked at documentation but couldn't find it spelled out. Here is are 2 examples from kamcmd ul.dump: { AoR: utkcf9b6h3(a)ata.phaxio.com Contacts: { Contact: { Address: sip:utkc4ff9b6h3@69.78.34.56:4615;transport=tcp Expires: 42 Q: -1.000000 Call-ID: d1c314305df3d342(a)192.168.0.3 CSeq: 129173 User-Agent: OBIHAI/OBi200-3.2.2.5898 Received: [not set] Path: [not set] State: CS_NEW Flags: 0 CFlags: 0 Socket: tcp:10.0.22.254:5060 Methods: 7839 Ruid: uloc-5fbdd369-491e-4 Instance: <urn:uuid:00000000-0000-0000-0000-9cadef62b3da> Reg-Id: 0 Server-Id: 0 Tcpconn-Id: 58 Keepalive: 0 Last-Keepalive: 1606317577 Last-Modified: 1606317577 } } } Contact: { Address: sip:mfjngfd272av@192.168.0.106:5060 Expires: 30 Q: -1.000000 Call-ID: 67413de375b2e76c(a)192.168.0.106 CSeq: 146320 User-Agent: OBIHAI/OBi300-3.2.2.5921 Received: sip:70.34.56.78:5060 Path: [not set] State: CS_NEW Flags: 0 CFlags: 64 Socket: udp:10.0.22.254:5060 Methods: 7839 Ruid: uloc-5fbdd369-490e-efa3 Instance: <urn:uuid:00000000-0000-0000-0000-9cadef810877> Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1606314719 Last-Modified: 1606314719 }

2 1

Next online devel meeting - Wed, Nov 25, 2020, 15:00UTC
by Daniel-Constantin Mierla 25 Nov '20

25 Nov '20

Hello, in order to plan the roadmap to next major release, it is time to organize another online devel meeting, so the developers and community members can syncronize and plan what should be done till Kamailio v5.5 will be out. Because freenode.net service is requiring user registration to join IRC channels on their servers, the meeting will be done on a matrix chat room. More details are in the wiki page of the event, available at: * https://www.kamailio.org/wiki/devel/irc-meetings/2020b Feel free to add there topics that you want to be discussed. The proposed date is next Wednesday, November 25, 2020, at 15:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 5

tsilo: ts_append() cannot find the ts_store() data
by Anthony Alba 25 Nov '20

25 Nov '20

Hi, having tsilo + PUSH problem. My probIem is no matter how I ts_store() the data, ts_append() cannot find any matching uri ERROR: tsilo [ts_append.c:64]: ts_append(): failed to retrieve record for sip:david@voice.example.com In route[LOCATION] { ts_store("sip:$tU@voice.example.com") } in route[PUSHJOIN] { ts_append("location", "sip:$tU@voice.example.com") } ts.dump looks like this { "jsonrpc": "2.0", "result": { "Size": 2048, "R-URIs": { "R-URI": "sip:david@192.168.122.7:5060;ob", "Hash": 57095964, "Transactions": { "Transaction": { "Tindex": 15146, "Tlabel": 805874386 } } }, "Stats": { "RURIs": 1, "Max-Slots": 1, "Transactions": 1 } }, "id": 43582 } Thanks Anthony Alba

3 7

kamailio.cfg fomatter
by marek 25 Nov '20

25 Nov '20

hi, i'm using https://github.com/miconda/vscode-kamailio-syntax in VScode. its great! But i need format code. Are there some formatters for kamailio.cfg? It can be for other editors than VScode. Marek

6 9

ACK Relay Issue Transaction Not Found
by me.projects＠yahoo.com 24 Nov '20

24 Nov '20

Hi When ACK is received on Kamailio from PBX t_check_trans() fails. 200 OK sent PBX from Teams and ACK from PBX is provided here. What could be the issue. SIP/2.0 200 OK FROM: "Ext1"<sip:1102@PBX_IP>;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f TO: <sip:1101@KAM_IP>;tag=e667397cb47c457a860222931e320b41 CSEQ: 25772 INVITE CALL-ID: a6685754-b402-4a15-82b2-5fb2551e5b58 VIA: SIP/2.0/UDP PBX_IP:7790;received=PBX_IP;branch=z9hG4bKPj170912fc-0908-40bb-8248-9f6cb560332d;rport=7790 RECORD-ROUTE: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>,<sip:sbc.callsprout.com;transport=tls;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f;lr>,<sip:KAM_IP:7790;transport=udp;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f;lr> CONTACT: <sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;ias=52.114.132.46~5061~3> CONTENT-LENGTH: 353 SUPPORTED: timer CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY REQUIRE: timer SESSION-EXPIRES: 1800;refresher=uac SERVER: Microsoft.PSTNHub.SIPProxy v.2020.11.16.5 i.USEA.5 X-Siptrace-Fromip: tls:KAM_IP:7790 X-Siptrace-Toip: udp:PBX_IP:7790 X-Siptrace-Time: 1606205334 607029 X-Siptrace-Method: INVITE X-Siptrace-Dir: out ACK sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;alias=.114.132.46~5061~3 SIP/2.0 Via: SIP/2.0/UDP PBX_IP:7790;rport;branch=z9hG4bKPjaa30c14c-c965-473d-bc49-9f223f9f634c From: "Ext1" <sip:1102@PBX_IP>;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f To: <sip:1101@KAM_IP>;tag=e667397cb47c457a860222931e320b41 Call-ID: a6685754-b402-4a15-82b2-5fb2551e5b58 CSeq: 25772 ACK Route: <sip:KAM_IP:7790;lr;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f> Route: <sip:sbc.callsprout.com;transport=tls;lr;ftag=e08f0a03-1120-4e13-b0e2-90d1a027c05f> Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 X-Siptrace-Fromip: udp:PBX_IP:7790 X-Siptrace-Toip: udp:KAM_IP:7790 X-Siptrace-Time: 1606205334 615495 X-Siptrace-Method: ACK X-Siptrace-Dir: in

5 10

How to route to calls to asterisk
by Mihai Cezar 23 Nov '20

23 Nov '20

Hi all, I need some help on getting to work my scenario: I have 3 asterisk servers, and every agents (soft phones like zoiper) is connecting to which server they need by contacting the PublicIP of each server. (the problem is that I need to reuse the IP address on others servers) . I want if is possible to use a proxy like "haproxy" but for sip to redirect the agents (softphones), to their coresponding server. For example: - james(a)ast.example.com -> must register to asterisk server 1, - sarah(a)ast2.exampe.com -> must register to asterisk server 2 - andy(a)ast3.example.com-> must register to asterisk server 3 ast.example.com, ast2.example.com, ast3.example.com = will share all the same ip, when the request arrives in the proxy it should check the vhost and redirect to the coresponding server for registering. Any hints and how to do this scenario?

2 3

ACK Not Relayed to MS Teams
by me.projects＠yahoo.com 23 Nov '20

23 Nov '20

Hi, I am integrating PBX with Teams via Kamailio Call is working but ACK delivery has an issue This is the ACK Kamailio receives from PBX I process the ACK with handle_ruri_alias(); and record_route_preset(AS per GUIDE LINES) But this change is not reflected in RELAY ACK from PBX ACK sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=0c344c6a-cd75-4095-9891-bf9445c28b82;x-c=5691aa770226562db7d3e16ab090e196/s/1/e857255ef2344e03bd1d12e270ef5e6c;ias=52.114.132.46~5061~3 SIP/2.0 Via: SIP/2.0/UDP PBX_IP:7790;rport;branch=z9hG4bKPjc50440d0-77db-4467-94cb-2f93c4f88538 From: "USER" <sip:13300@PBX_IP>;tag=ab87b9b5-eee3-420c-8472-2eec82a769ac To: <sip:1508@KAMAILIO_IP>;tag=a2ce378b106949cf8bac8f978bb573fb Call-ID: 3dbc6680-fec3-4ba5-8232-6dda8c669bca CSeq: 23811 ACK Route: <sip:KAMAILIO_IP:7790;lr;ftag=ab87b9b5-eee3-420c-8472-2eec82a769ac> Route: <sip:sbc.KAMAILIO_FQDN.com;transport=tls;lr;ftag=ab87b9b5-eee3-420c-8472-2eec82a769ac> Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 X-Siptrace-Fromip: udp:PBX_IP:7790 X-Siptrace-Toip: udp:KAMAILIO_IP:7790 X-Siptrace-Time: 1606135546 516717 X-Siptrace-Method: ACK X-Siptrace-Dir: in

2 3

textops.subst_uri does not change all branches
by Anthony Alba 23 Nov '20

23 Nov '20

Hello kamailioers, Version 5.4.2. I have a R-URI that I need to change TLS -> TCP in all branches. I do this in request route before t_relay: // AoR has multiple contacts, all with transport=TLS route[REQUEST] { textops.subst_uri("/transport=TLS/transport=TCP/i") //parallel forking //why doesn't this change all branches? t_relay() } If I print out the R-URI in the branch route, I see that only the first R-URI is changed, in the other branches the R-URI transport=TLS is not changed to transport=TCP. Any suggestions on how to get this to work across all Contact addresses? Thanks Anthony Alba

2 1

Issue with ca-list
by George Goglidze 20 Nov '20

20 Nov '20

Hi Folks, I was wondering if somebody could help me with an issue. I’m a newbie here, just installing Kamailio sip server. I’ve enabled TLS, and am trying create a SIP Trunk to external SIP Service which is TLS enabled port 5061. I’ve configured the following in tls.cfg: [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certs/sbc-private.pem certificate = /etc/kamailio/certs/godaddy.pem ca_list = /etc/kamailio/certs/calist.pem In the section above – ca_list = calist.pem contains all the CA’s and Subordinates of the destination server. Private_key and certificate are of my own server (public godaddy signed) [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certs/sbc-private.pem certificate = /etc/kamailio/certs/godaddy.pem ca_list = /etc/kamailio/certs/godaddyca.pem In the section above the ca_list is godaddy’s ca and subordinate. In the wireshark I can see that I’m sending out SIP OPTIONS PING (I’m using dispatcher module). Then the server replies with tls SERVER HELLO which includes it’s certificate But for some reason we are rejecting it: Alert (level: fatal, Description: Unknown CA) How should I set this up to make sure the remote server CA’s are verified? Thank you,

3 9

t_uac_send function with From tag
by João Vitor Arruda 20 Nov '20

20 Nov '20

Hello, I'm trying to use the function t_uac_send inside a failure_route as described in https://kamailio.org/docs/modules/5.3.x/modules/tm.html#tm.f.t_uac_send to send a CANCEL out. Although the documentation says "it can include From/To tags" i was not able to get the from tag provided to be used. Another random generated one is used. I've tried the following format without success: t_uac_send("CANCEL", "$ru", "", "", "From: <$fu>;tag=$ft\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") t_uac_send("CANCEL", "$ru", "", "", "From: $fu;tag=$ft\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") t_uac_send("CANCEL", "$ru", "", "", "From: bob(a)kamailio.org;tag=2w3e\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") In the last one which is copied from the documentation that is how the >From header going out looks like: From: <bob(a)kamailio.org;tag=2w3e>;tag=3393f0703fb0ccaca74109ff37de39f5-36d71ef0 Any tips on how to get the From tag passed to the function to be used? Thanks, Joao Arruda

2 6

Kamailio suddenly stopes responding to sip messages
by Uttam Giri 19 Nov '20

19 Nov '20

Hi, We are using Kamailio with Jitsi client where we observed sometimes Kamailio stops responding to sip messages. As a result, my Jitsi client goes offline and never back online. Also in the Kamailio syslog there are no error logs(debug enabled) related to this issue. As a temporary workaround, we had to restart the Kamailio server all the time. Does anyone observed a similar issue and can suggest any resolution. Kamailio version:- version: kamailio 5.3.5 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, 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 with gcc 7.5.0 UDP process when Kamailio stops responding:-. root@ip-10-68-16-14:/home/ugiri# kamctl ps { "jsonrpc": "2.0", "result": [ { "IDX": 0, "PID": 15436, "DSC": "main process - attendant" }, { "IDX": 1, "PID": 15442, "DSC": "udp receiver child=0 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 2, "PID": 15443, "DSC": "udp receiver child=1 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 3, "PID": 15444, "DSC": "udp receiver child=2 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 4, "PID": 15445, "DSC": "udp receiver child=3 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 5, "PID": 15446, "DSC": "udp receiver child=4 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 6, "PID": 15447, "DSC": "udp receiver child=5 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 7, "PID": 15448, "DSC": "udp receiver child=6 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 8, "PID": 15449, "DSC": "udp receiver child=7 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 9, "PID": 15450, "DSC": "slow timer" }, { "IDX": 10, "PID": 15451, "DSC": "timer" }, { "IDX": 11, "PID": 15452, "DSC": "secondary timer" }, { "IDX": 12, "PID": 15453, "DSC": "JSONRPCS FIFO" }, { "IDX": 13, "PID": 15454, "DSC": "JSONRPCS DATAGRAM" }, { "IDX": 14, "PID": 15455, "DSC": "ctl handler" }, { "IDX": 15, "PID": 15456, "DSC": "Http Async Worker" }, { "IDX": 16, "PID": 15457, "DSC": "Http Async Worker" }, { "IDX": 17, "PID": 15458, "DSC": "TIMER NH" }, { "IDX": 18, "PID": 15459, "DSC": "tcp main process" } ], "id": 17709 } Thanks, Uttam

2 3

Re-Invite from callee With RTPEngine between RTP--SRTP
by me.projects＠yahoo.com 19 Nov '20

19 Nov '20

Hi, I have have Kamailio as SBC one-end is SRTP and other end is RTP So I m using rtpengine_manage("replace-origin replace-session-connection DTLS=passive OSRTP-offer ICE=force SRTP")rtpengine_manage("replace-origin replace-session-connection DTLS=off SDES-off ICE=remove RTP") 1-Asterisk(RTP) -----2-Kamailio/RTPEngine ------ 3- Carrier with SRTP I detect source IPs and to manage this all. When call is made from 1 to 3 all works and during call 3 send re-Invite to Hold Call .. We need to call rtpengine_manage and also when on 200OK reply of re-invite from 1. How shall I detect that 1- if this is Re-Invite (has_totag() )and 2- its from Callee. Please advise.

1 0

Presence of plain text username and password in kamailio.cfg
by Ahmed Marsou 19 Nov '20

19 Nov '20

Hi; I want to remove all plain text usernames an passwords from kamailio.cfg file. Like modparam("auth_db", "db_url", "dbdriver://username:password @dbhost/dbname") or this modparam("sqlops","sqlcon","ca=>dbdriver://username:password @dbhost/dbname") Can you help me with some ideas of how can I handle that? Thank you.

5 10

Add Reason Header to kamailio generated CANCEL
by Alfonso Pinto 18 Nov '20

18 Nov '20

Hi everyone, I've got a specific case: when the inv_fr times out, I need to add a Reason header to the CANCEL generated by kamailio. I've tried to see if I could do it in the onsend_route, but that one is not triggered for the generated CANCEL. I also checked event_route[tm:local-request], but that one isn't triggered either for the generated CANCEL. Is there any way to do it? Or maybe to have any pointer about where to look in the code so I may try to trigger event_route[tm:local-request] for these generated CANCELs? Regards, Alfonso

6 17

record-route and INFO
by David Villasmil 18 Nov '20

18 Nov '20

Hello guys, I have a setup like this: FS(private)-->(private)Kamailio1(Public)-->Kamailio2(public)-->provider. I have mhomed=0, enable_double_rr=1, I'm listening on 2 different ports, 1 private, 1 public (with advertise) and on Kamailio1 the DISPATCH ROUTE I force the socket $fs = "LISTEN_PRIVATE_PROTO:LISTEN_PRIVATE_IF:5080"; To the private ip when the destination IP is a private IP. This all works fine. But When calling OUT (i.e. public) kamailio adds itself only for the private IP: *INVITE sip:12345678@mydmain.com <sip%3A12345678(a)mydmain.com> SIP/2.0Record-Route: <sip:172.31.25.124:5080;lr;did=9da.3a61;nat=yes>Via: SIP/2.0/UDP 172.31.25.124:5080;branch=z9hG4bKf05e.979fd257ac7f288804ef146d12c1e0fb.0Via: SIP/2.0/UDP 172.31.6.1:5080;received=172.31.6.1;rport=5080;branch=z9hG4bKQ3H8mgHaUjaBm* So when INFO messages come from the provider to kamailio2, kamailio 2 tries to send the INFO straight to "172.21.25.124". Should i manually add a record-route when calling OUT with record_route_advertised_address("1.2.3.4:5090"); Or is there a better way of getting this working? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

Info: URL expander browser extension for Kamailio resources
by Daniel-Constantin Mierla 18 Nov '20

18 Nov '20

Hello, short note to the community forum in case someone finds it useful -- I wrote a tiny browser extension to facilitate navigation to various web resources related to Kamailio project, helping to jump quicker to web pages by typing some keywords on the address bar in the modern browsers. Installation and usage instructions are in the readme of the project, available at: * https://github.com/miconda/KSR-URL-Expander A few other details are in a news article: * https://www.kamailio.org/w/2020/11/browser-url-expander-for-kamailio-resour… Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 1

UAC Module
by Javier Valencia 18 Nov '20

18 Nov '20

Hi there! Configuring the UAC module to use "remote registers" in a Kamailio 4.4.3. Everything it's working except for the flags database field to which I always put a value of 0. But when Kamailio starts to launch the REGISTRATION that field does not persist in the DB. In the kamailio.cfg file I have not made any changes, I only use the remote registers DB feature. Is it normal behavior? Do I have any parameters not set or incorrectly set? Regards, JV

2 2

utility function to check if ip address is private?
by Juha Heinanen 18 Nov '20

18 Nov '20

I looked in the wiki for a utility function that would check if given ip address is private, but didn't find one. Does it exist? -- Juha

3 5

how to add URI parameter
by Sergiu Pojoga 18 Nov '20

18 Nov '20

Hi there, Does Kamailio have a function to add URI parameters? I looked around but couldn't find one. I refuse to believe Kamailio doesn't have one. There's a *add_uri_param(param)*, but that's strictly for R-URI. I need it for something else, Contact. Thanks, --Sergiu

2 1

help: can anybody give me a configuration template of SBC?
by 陈理军 18 Nov '20

18 Nov '20

Hi I want to configure Kamailio SIP server to act as a SBC. I had read the article of Kamailio working as SBC to connect MS Team project:   https://skalatan.de/en/blog/kamailio-sbc-teams But I can not find the kamailio.cfg file for this scenario. Can anybody give me a configuration template of SBC? BRs Albert

3 2

Multiple Registrations via WebSocket & Cisco phones
by Melek Oktay 17 Nov '20

17 Nov '20

Hi, I am using FreeSwitches behind the Kamailio proxy server and I am trying to allow multiple registration to my extensions. So, following configuration is sample of my Kamailio modparam("registrar", "xavp_cfg", "reg") ...... $xavp(reg=>max_contacts) = 10; save("location"); .... I saw my phones could register with the same account credentials via several phones such as Cisco, Zoiper, Yealing etc. When the call is forward to this extension, all of them are ringing. Very Nice. But, when I am trying to REGISTER WebRTC supports soft-phones to my system and with the same account credentials, my extensions are not ringing like in the previous scenario. WebRTC uses Websocket (WS) technology and clients register to Kamailio via usrloc module. When the call is forward to this extension, Kamailio try to replicate WebRTC'S INVITE packet to other phones (Cisco, Yealing, zoiper etc) and none of them understand incoming INVITE request because of WebRTC supported protocols (ICE ,a=candidate) , in a brief, phones could not recognize/understand incoming WebRTC request. This is a really tough issue for me, how can I send appropriate INVITEs for each of them.

2 2

The Algorithm “13” - latency optimized dispatching
by Julien Chavanton 14 Nov '20

14 Nov '20

About The Algorithm “13” - latency optimized dispatching, Is now reviewed once and tested, it will most likely be ready to merge soon. I want to share my thoughts on it one more time as it is not too late to get more feedback before we merge. I think it is the best algorithm in most use cases, here is why : It is providing round-robin and fail-over with automatic de-prioritization of slow/unresponsive gateways. You probably asked yourself the following questions in the past : "How do I set the thresholds to put a gateway out of service ?" *ds_probing_threshold*, *ds_inactive_threshold* and timers ... - If your thresholds are too strict, you may end up running out of gateway. - If your thresholds are too tolerant, you may end up adding excessive delays to call establishment and using degraded gateways. The automatic de-prioritization can help to address this concern more efficiently by providing more flexibility. - it can react faster than lets say 2 consecutive timeouts. - it will not disable gateways but simply de-prioritize / reorder them if needed. The only main drawback I can imagine is when you always need to evenly distribute calls using round-robin. It may be needed sometimes but in this case it means you are willing accept to send calls to a degraded gateway or trough degraded network paths. Even if you may select to preset a mixture of round-robin sets, thanks to *ds_select_routes* however it will stay static, needs to be configured precisely, and will not react to degradation automatically. I hope this will help use to protect QoS and lower latency of calls routed by Kamailio. Feel free to let me know what you think Julien

3 6

rport parameter in VIA header in uac remote registration
by Jetse Verschuren 12 Nov '20

12 Nov '20

Hi, I'm currently trying to connect Kamailio, running in docker, to my FritzBox. The FritzBox is the modem provided by my ISP, with a builtin voip server which I can use to dial out (basically a PSTN gateway). In my setup I have a custom router/firewall/NAT between the FritzBox and the server. When I connect LinPhone to the FritzBox, from behind the NAT, it works fine. But when I use the Kamailio uac remote registration, I don't get a response. After some investigation with wireshark, I found that the the SIP response from the FritzBox comes on the wrong port, and gets rejected by the NAT. The request comes from port 1025, but the response is sent to port 5060. I compared the requests from LinPhone and Kamailio, and combined with some research, found it probably was the rport option in the VIA header. >From what I read, it basically signals the uas to use the source port to reply to, instead of the port indicated in the request, which would be necessary to effectively go through a NAT. After reading a lot of documentation (and even parts of the source), I wasn't able to find an option to enable this option. Is there some option I completely missed, or is it not possible? Or am I completely wrong in my "diagnosis"? Thanks in advance, Jetse

3 4

Having trouble Setup Kamailio as a load balancer for Asterisk servers using module dispatcher
by Usama Shaikh 11 Nov '20

11 Nov '20

Hello Everyone. I am having trouble to setting up kamailio for load balancing for asterisk servers. I am using module dispatcher for load balancing AND two different servers for kamailio and asterisk. I am doing it first time and have not exact idea about its all configs… but i have researched and done so far, now need help from you guys. Currently, trying to setup with the one kamailio and one asterisk then will go ahead if I get any success… using MariaDB MySQL for the dispatcher data. Simply I want to do is : 1. Register my asterisk devices with kamailioIP (and I am able to do this using below configs) Devices are getting register with the asterisk from with kamailioIP 2. I want to receive calls on the registered devices with KamailioIP (and i am not able to do this) WORK FLOW: KAMAILIO IP = XX.XX.XX.95 on port 5060 and Asterisk IP = XX.XX.XX.164 on port 5060 SIP Device(ZOIPER) -------->>>>KAMAILIO SERVER(device register/load balancer)--------->>> ASTERISK SERVER Please have a look into my configuration files and let me know if anything is missing or wrong in these. Or anything to add or remove from the confs… my kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ASTERISK #define WITH_NAT #!define WITH_DEBUG #!define WITH_USRLOCDB # # Kamailio (OpenSER) SIP Server v3.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL7 log_name="kamailio" children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:0.0.0.0:5060 #listen=udp: KamailioIP:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = " ASTERISKIP.164" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = " KamailioIP .95" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="/usr/local/kamailio-devel/lib64/kamailio/modules/" #!else mpath="/usr/local/kamailio-devel/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "sctp.so" loadmodule "dialog.so" loadmodule "siptrace.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" loadmodule "dispatcher.so" loadmodule "jsonrpcs.so" loadmodule "htable.so" # ---------------------- mod dispatcher params ------------------------------ modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_ping_from", "sip:dispathcer@ KamailioIP ") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "ds_ping_method", "INFO") #modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) #modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;code=488;code=481;class=3") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") #modparam("dispatcher", "priority_col", "dstpriority") #modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") # do failover modparam("dispatcher", "flags", 2) #modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") #modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") #modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") #!endif # ----------------- setting module-specific parameters --------------- modparam("siptrace", "db_url", DBURL) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ---------------------- mod uac params ------------------------------ modparam("uac", "reg_contact_addr", "ASTERISKIP:5060") modparam("uac", "reg_db_url", "mysql://kamailio:kamailiorw@localhost /kamailio") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac","restore_mode","auto") modparam("uac", "restore_dlg", 1) modparam("uac","auth_realm_avp","$avp(s:auth_realm_avp)") modparam("uac","auth_username_avp","$avp(s:auth_username_avp)") modparam("uac","auth_password_avp","$avp(s:auth_password_avp)") # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "ruri_matching", 0) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- 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", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "matching_mode", 1) modparam("usrloc", "db_insert_update", 1) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "secret") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes #modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "stats=>size=6;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- #modparam("debugger", "cfgtrace", 1) #modparam("debugger", "breakpoint", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { route(DISPATCH); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) if(is_method("OPTIONS")) { sl_send_reply("200","Keepalive"); exit; } if(is_method("NOTIFY")) { sl_send_reply("200","OK"); exit; } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } 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 t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio append_hf("FromIp: $src_ip\r\n", "Call-ID"); route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server 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; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } if (is_method("REGISTER") || from_uri==myself){ if (!allow_trusted()) { xlog('wokred here for ipauth check'); sl_send_reply("403", "Forbidden"); } } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sip_users", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } if(t_check_status("401")) { uac_auth(); } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } t_relay(); #!endif } # Test if coming from Asterisk route[FROMASTERISK] { # if($si==$sel(cfg_get.asterisk.bindip) # && $sp==$sel(cfg_get.asterisk.bindport)) # return 1; # return -1; # dispatch: is this asterisk ip ? if (ds_is_from_list("1")) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { # $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" # + $sel(cfg_get.asterisk.bindport); # append_hf("X-Asterisk-Context: dexter-phones\r\n"); # route(RELAY); # exit; # select asterisk port append_hf("X-Asterisk-Context: dexter-phones\r\n"); route(DISPATCH); } route[DISPATCH] { # round robin dispatching on gateways group '1' #xlog("<<<<<<<<<<<<<<<<trying for dispatch>>>>>>>>>>>>>>>>>>>"); #if (is_method("INVITE")) { # dst_select( "GROUP", "HASH METHOD") #ds_select_dst("1","4"); #sl_send_reply("100","Trying"); # forward(); # exit(); #} if (!ds_select_domain("1", "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } “kamctl dispatcher dump” looks like this: { "jsonrpc": "2.0", "result": { "NRSETS": 1, "RECORDS": [{ "SET": { "ID": 1, "TARGETS": [{ "DEST": { "URI": "sip:ASTERISKIP:5060", "FLAGS": "AP", "PRIORITY": 1, "ATTRS": { "BODY": "rweight=50;weight=50;cc=1", "DUID": "", "MAXLOAD": 0, "WEIGHT": 50, "RWEIGHT": 50, "SOCKET": "", "SOCKNAME": "", "OBPROXY": "" } } }] } }] }, "id": 13403 } And also have set trunk in ASTERISK sip.conf register => kamailioSIP:XXXX@KamailioIP:5060 [kamailio] type=friend host= KamailioIP port=5060 transport=udp allow=all allow=gsm allow=alaw,ulaw insecure=invite,port Permit = 0.0.0.0/0.0.0.0 sipdebug=yes username=kamailioSIP canreinvite=no secret=XXXX qualify=yes dtmfmode=auto when i Register my asterisk devices with kamailioIP (and I am able to do this) Devices are getting register with the asterisk from with kamailioIP asterisk -r : (here 8101 and 8100 is registered with Kamailio IP ) [Nov 9 10:36:02] -- Registered SIP '8101' at XX.XX.XX.95:5060 [Nov 9 10:36:02] -- Unregistered SIP '8101' [Nov 9 10:36:03] -- Registered SIP '8101' at XX.XX.XX..95:5060 [Nov 9 10:36:03] == Manager 'listencron' logged on from 127.0.0.1 [Nov 9 10:36:04] == Manager 'sendcron' logged off from 127.0.0.1 [Nov 9 10:36:07] -- Unregistered SIP '8100' [Nov 9 10:36:09] -- Registered SIP '8100' at XX.XX.XX..95:5060 [Nov 9 10:36:09] > Saved useragent "Z 3.15.40006 rv2.8.20" for peer 8100 [Nov 9 10:36:09] NOTICE[8453]: chan_sip.c:23869 handle_response_peerpoke: Peer '8100' is now Reachable. (51ms / 2000ms) when i dial 8100 to 8101 i get in my asteris -r NOTICE[8453][C-0000000b]: chan_sip.c:26002 handle_request_invite: Call from '8101' (95.217.223.95:5060) to extension '8100' rejected because extension not found in context 'none-dial'. and not able to call extension to extension, PLEASE HELP to do this, I have search a lot and tried so much of stuff in configs but no success still kindly guide me if anything I am missing to do in the process or anything I am doing wrong. I will be eagerly waiting for any help or pointer… Thanks in advance

3 3

Re: [SR-Users] Checking user registration
by ahmed moghazy 11 Nov '20

11 Nov '20

When i used lockup location it is done but if the user registered one time on server the location will be saved on kamailio so i want a method to check registered users live every second for example and i want to delete locations of un registered users ? Thank u. > On 10/11/2020, at 12:41 PM, ahmed moghazy <ahmed.m.moghazi(a)gmail.com> wrote: >

2 2

Checking callee registration
by ahmed moghazy 10 Nov '20

10 Nov '20

I want to check if callee is registered or not before routing call because this made error and make caller unregistered untill i restarted the linphone

3 3

Checking user registration
by ahmed moghazy 10 Nov '20

10 Nov '20

I want kamailio to reject call and send message user is not registered when a caller is trying to call unregistered user (callee) , Thank you.

2 1

Expire timeout for notify message answer
by Victor Litvinov 10 Nov '20

10 Nov '20

Hello dear community. We use a presence module with the enabled option "timeout_rm_subs". And we face issues that subscriptions remove too early. We expect that it will be removed after the fr_timer, but it is removed after 1 second. We I can see in log (i'm add additional logs): when we send NOTIFY: tm [timer.c:491]: retr_buf_handler(): timer retr_buf_handler @1436309038 (timer_ln 0x7f4708f45bd0 -> rbuf 0x7f4708f45bb0 -> [0x7f4708f458a0 <-> 11592:145228 527]) tm [timer.c:492]: retr_buf_handler(): fr_expire 1436309054 T1 4000 T2 4000 fr_timeout 80 tm [timer.c:536]: retr_buf_handler(): new interval 4000 ms / 64 ticks (max 4000 ms) when timer expire: tm [timer.c:491]: retr_buf_handler(): timer retr_buf_handler @1436309054 (timer_ln 0x7f4708f45bd0 -> rbuf 0x7f4708f45bb0 -> [0x7f4708f458a0 <-> 11592:145228 527]) tm [timer.c:492]: retr_buf_handler(): fr_expire 1436309054 T1 4000 T2 4000 fr_timeout 80 tm [timer.c:367]: final_response_handler(): transaction [0x7f4708f458a0 <-> 11592:145228527] scheduled for deletion tm [timer.c:424]: final_response_handler(): stop retr. and send CANCEL (0x7f4708f458a0) and then subscription remove. As we can see the fr_expire variable was set to ticks+16 ( 1 second). And another strange thing that T1 and T2 timers are the same. So the question is if this behaviour is correct, how can we configure the presence module fr_timer as an expiration value for the answer to notify? And can we use it without retransmissions?

2 1

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