- sr-users - kamailio.org

how to check path vector after lookup("location")
by Sergey Safarov 21 Aug '20

21 Aug '20

I use this construction to check the presence of path in device registration if(reg_fetch_contacts("location", "$fu", "caller") && $(ulc(caller=>path)[0]) == $null) { handle_ruri_alias(); } Is possible to check path presence after lookup("location") function call? Sergey

2 1

Bug in Postgres SQL Script?
by Noah Mehl 20 Aug '20

20 Aug '20

Hey all, I’m trying to understand the following lines in the kamdbctl.pgsql script: https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81… <https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81…> https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81… <https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81…> https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81… <https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81…> Basically, there are no tables created for Postgres that end in _id_seq. Am I missing something here? Thanks! ~Noah

2 2

kamailio+rtpengine for media not playing media file while hold the call
by Amit Pal 20 Aug '20

20 Aug '20

Dear Team, I am using kamailio with rtpengine for media. when call getting hold no media sound play (configure hold.mp3). >From rtpengine gets following error. rtpengine[624]: ERR: No suitable SDP section for media playback rtpengine[624]: WARNING: Protocol error in packet from 127.0.0.1:46333: Failed to start media playback from file [d8:supportsl10:load limite4:file38:/home/coralswitch/queuesounds/hold.mp37:call-id60:89162207232 fae50MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.13:received-froml3:IP414:19 2.168.20.254e8:from-tag32:9f5288f14b704b28a1c2dfc2112250eb6:to-tag8:1e05e757 7:command10:play mediae] rtpengine[624]: INFO: Received command 'offer' from 127.0.0.1:46333 rtpengine[624]: INFO: Replying to 'offer' from 127.0.0.1:46333 (elapsed time 0.000106 sec) /usr/local/sbin/kamailio[9094]: ERROR:rtpengine [rtpengine.c:2619]: rtpp_function_call(): proxy replied with error: Failed to start media playback from file Attached file show complete log of one call with hold /unhold actions. Please advise what I miss to configure here. Regards Amit Pal

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Module IMS_QOS does not support Re-Auth Request (RAR)
by Hamid Hashmi 20 Aug '20

20 Aug '20

Kamailio User Does module ims_qos support Re-Auth Request? [cid:282322de-506e-4b6e-a731-24841b517573] I observed that the QoS module is not responding to RAR. Kamailio version is version: kamailio 5.2.0-dev6 (x86_64/linux) 48ed62-dirty 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 48ed62 -dirty compiled with gcc 4.8.4 Regards Hamid R. Hashmi

2 1

t_load_contacts in 5.4.0
by Cindy Leung 20 Aug '20

20 Aug '20

Hello all, Just started using v5.4.0 on our system and I noticed a change in behavior when doing the append_branch, t_load_contacts, and t_next_contacts combo. Previously using v5.3.4 and it appears to be fine. Here's the call scenario: Kamailio receives a call to sip:1001@carrierB. Kamailio sees carrierB and appends 2 contacts: gateway1.carrierB.com;q=0.3 and gateway2.carrierB.com;q=0.2. After t_load_branches(0), I expect to see $rd to be changed to gateway1.carrierB.com, but it's not. This is the piece of config I'm trying to debug: xlog ("=== branch 0: $(branch(uri)[0]), $(branch(q)[0])\n"); xlog ("=== branch 1: $(branch(uri)[1]), $(branch(q)[1])\n"); t_load_contacts(0); while (t_next_contacts()) { xlog ("=== rd = $rd\n"); } This is the log I get. It appears to use the backup contact first. ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === branch 0: sip:1001@gateway2.carrierB.com;transport=udp, 200 ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === branch 1: sip:1001@gateway1.carrierB.com;transport=tcp, 300 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:522]: t_load_contacts(): load_contact mode selected: 0 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:340]: ki_t_load_contacts_mode(): nr_branches is 2 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f983cb66608 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:890]: ki_t_next_contacts(): Appending branch uri-'sip:1001@gateway1.carrierB.com;transport=tcp' dst-'' path-'' inst-'' ruid-'' location_ua-'' DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f983cb66350 ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === rd = carrierB DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f983cb66078 ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === rd = gateway2.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:627]: ki_t_next_contacts(): no contacts in contacts_avp - we are done! t_load_contacts(1) works a little better. But we don't need the probability feature. ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === branch 0: sip:1001@gateway2.carrierB.com;transport=udp, 200 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === branch 1: sip:1001@gateway1.carrierB.com;transport=tcp, 300 DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:522]: t_load_contacts(): load_contact mode selected: 1 DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:340]: ki_t_load_contacts_mode(): nr_branches is 2 DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:280]: t_load_contacts_proportional(): proportionally selected contact with uri: sip:1001@gateway1.carrierB.com;transport=tcp (q: 300, random: 287, q_index: 500, q_total: 500) DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:280]: t_load_contacts_proportional(): proportionally selected contact with uri: sip:1001@gateway2.carrierB.com;transport=udp (q: 200, random: 157, q_index: 200, q_total: 200) DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:303]: t_load_contacts_proportional(): proportionally added backup contact with uri: sip:1001@carrierB SIP/2.0#015#012Via: SIP/2.0/UDP 172.18.1.21:5060;branch=z9hG4bK-21-1-0 blah blah blah (q: -1) DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f6e457ec078 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === rd = gateway1.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f6e457ec350 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === rd = gateway2.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f6e457ec608 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === rd = carrierB DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:627]: ki_t_next_contacts(): no contacts in contacts_avp - we are done! As a comparison, this is what we've been getting in 5.3.4 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === branch 0: sip:1001@gateway2.carrierB.com;transport=udp, 200 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === branch 1: sip:1001@gateway1.carrierB.com;transport=tcp, 300 DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: tm [t_serial.c:191]: ki_t_load_contacts(): nr_branches is 2 DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: <core> [core/xavp.c:529]: xavp_destroy_list(): destroying xavp list 0x7fe13146a680 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === rd = gateway1.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: <core> [core/xavp.c:529]: xavp_destroy_list(): destroying xavp list 0x7fe13146a3a8 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === rd = gateway2.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: <core> [core/xavp.c:529]: xavp_destroy_list(): destroying xavp list 0x7fe13146a0d0 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === rd = carrierB DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: tm [t_serial.c:460]: ki_t_next_contacts(): no contacts in contacts_avp - we are done! Is there anything else that's been changed in branch building that we should pay attention to? Thanks. Cindy

2 4

Domain Name changes
by Pavithra Mohanraja 19 Aug '20

19 Aug '20

Hi, Is there any way to change the domain name at run-time Kindly help. Thanks, Pavithra

2 4

dlg_refer of callee leg issue
by Amit Pal 19 Aug '20

19 Aug '20

Dear kamailio Team, I using dlg_refer for refer callee leg on specific condition to Freeswitch conference. But I got decline message. in reverse caller leg refer to freeswitch conference. can pls tell how to achieve it. Regards Amit Pal

2 4

Force registrations on Kamailio save()
by Duarte Rocha 19 Aug '20

19 Aug '20

Greetings, I'm currently creating a Kamailio with registrations and i'm unsure how to use the save() function. I want to limit the number of contacts for the AOR to 2. I also want to force the registration of new contacts even if the maximum number is reached, removing the older ones. I saw that on opensips i could use the flags "fc2" for this but they aren't available on Kamailio's save(). I'm using the max_contacts parameter but in the docs it states the once the limit is reached, it will reject new Registers. How can i implement the force registration behaviour here? Best Regards, Duarte Rocha

2 1

Decrypting SRTP with Kamailio + rtpengine
by Ildar Mukhametzhanov 18 Aug '20

18 Aug '20

Hello! Decrypting SRTP with Kamailio + rtpengine. Hello! I have a task to decrypt SRTP for a legacy switch which doesn’t support it. Simply adding rtpengine_manage("RTP/AVP") didn’t help. SDP converted from SRTP to RTP while proxying INVITE to legacy switch, but 180 Ringing wasn’t converted back to SRTP. Little more detailed config snippet: route[RELAY] { ... if (is_method("INVITE")) { if ($rd=~$var(switch_ip_mask)) { rtpengine_manage("RTP/AVP"); } else { rtpengine_manage(); } ... } Could someone please advise typical scenarios to achieve it, I suppose the task is pretty standard. Thank you in advance!

4 3

Kamailio World Updates Online - Celebration Party - Sep 2-3, 2020
by Daniel-Constantin Mierla 18 Aug '20

18 Aug '20

Hello, we are considering to organize a small online event to compensate a bit the cancellation of Kamailio World 2020, but mainly to discuss about what is new around Kamailio project and RTC space. The event was discussed among a couple developers several weeks ago, when it was clear that the pandemic will last longer, preventing any on-site event, and the intention to do it was also announced during the recording of last ClueCon Weekly about Kamailio project at the beginning of July (https://www.kamailio.org/w/2020/07/cluecon-weekly-kamailio-updates-july-202…). The first commit of Kamailio code was done on the 3rd of September 2001, thus this is like the project's birthday, so a good opportunity to celebrate that as well. The event is planned to be 2 sessions of about 4 hours on each day, with a few presentations per day, but try focus on open discussions about. The goal is to keep in very informal, without any significant burden for organizers or participants, not to add to the daily stress with work and family in the pandemic time. It should be more like meetup style, with no registration for participants, sessions may not be recorded and can be done without slides. Likely we will use a (self-hosted) video conferencing system (Jitsi) for presentations and the IRC or Matrix channels for chatting. The time frames we look at are 15:00-19:00 Berlin time zone (13:00-17:00 UTC) on the 2nd and the 3rd of September 2020. The purpose of this email for now is to see the interest of the community for such event, reply if you think it is useful to organize it and you plan to participate. If there is interest, then I will follow up with more details about speakers and the structure of the event. 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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Do I need RTPEngine at every location onsite for webRTC
by Johnny Ritzer 18 Aug '20

18 Aug '20

We have 100 customer locations and I was hoping to no deploy a kamabox at each location if traffic is mostly just local. But we are building a WEBRTC. and hoping our architecture can use kamailio in the cloud. Each location has a PBX onsite. We are deploying webRTC so need that websocket setup jssip/sip.js. SIP SIGNALING (relay) () UA(webphone) --> Cloud Kamailio --> PBX 10.0.1.100 --> cloud.kama.com --> 10.0.1.1 But then translate uac and get RTP to pass locally without a device onsite at each location. RTP Flow (not going through kamailio) UA(webphone) --> PBX 10.0.1.100 --> 10.0.1.1

2 1

TCP/TLS CRITICAL error. send_fd(): sendmsg failed sending : Too many references: cannot splice
by Dmytro M 18 Aug '20

18 Aug '20

While executing this piece of config script, after a few days of run time, Kamailio starts experiences some issue that leads to instability and unresponsiveness of the proxy via TCP with TLS. UDP workers seems to be working fine... TCP clients can't connect. route[RELAY] { ... if (!t_relay()) { sl_reply_error(); } exit; } Please advise how we can troubleshoot this issue. Are there any immediate (or obvious) TCP/TLS/FD Kamailio/Linux parameters that should be increased to allocate more resources for the Proxy? Or may be any other parameters that may lead to a quicker resources release. Please see the errors bellow... Environment: Ubuntu 16 x86_64, Kamailio 5.3.5. Thanks, Dmytro Aug 15 04:48:19 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 164 on 16: Too many references: cannot splice (109) Aug 15 04:48:19 ERROR: <core> [core/tcp_main.c:3908]: handle_ser_child(): CONN_GET_FD: send_fd failed Aug 15 04:48:19 ERROR: tm [../../core/forward.h:292]: msg_send_buffer(): tcp_send failed Aug 15 04:48:19 WARNING: tm [t_fwd.c:1573]: t_send_branch(): sending request on branch 0 failed Aug 15 04:48:19 ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) .... Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e91c31e0 (flags 0x4018), fd 74 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 75 on 43: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e9d727b0 (flags 0x4018), fd 75 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 75 on 73: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e9d727b0 (flags 0x4018), fd 75 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 75 on 45: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e9d727b0 (flags 0x4018), fd 75 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 74 on 43: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e91c31e0 (flags 0x4018), fd 74 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 76 on 43: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e7856fa8 (flags 0x4018), fd 76 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 74 on 47: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e7856fa8 (flags 0x4018), fd 74 .....

2 2

What is a good strategy to build Kamailio <--> pbx
by Johnny Ritzer 16 Aug '20

16 Aug '20

What are your thoughts on architecture build of the following scenarios. The PBX doesn’t support websockets and we want to use sip.js / jsSip. (use Kama as wss:// SIP proxy) PBX has direct PRI/sip Trunks/ DID control/ Voicemail. PBX hosts SIP extensions. UA --> Kamailio --> PBX --------SCENARIO A.) ------- Host SIP[xxxx] extensions on Kama box but same SIP extensions are also on PBX [xxxx] 1. Build some sort of control to register handle kama to pbx …. (but reg webclients) Kama [1100] SIP-PBX [1100] Does kama act as the UA 1100 / registered to PBX What about UA to kamailio (do we do 1100a so UA-kama is 1100a) but kama is maintain REG/status/ but passing that IF client is online ? MWI (voicemail is hosted on PBX) so how does that move through can it. ------- SCENARIO B.) ------ Proxy all/ everything . use route[REGFW] Forward all REG to PBX.. Issues ---- MWI (SIP notify ?) SIP Incoming CALL incoming calls to work. (if UA holds REG to outbound proxy on PBX or ) DTMF (sipINFO) doesn’t work (I see it sending) I noticed I crashed the PBX because all the SIP scanners started pounding my Kamailio box after 2 days when I added route[REGFW] it starting eating up SIP trunk channels on PBX cause started forwarding everything. Like it starting 4241(a)xxxxx.com 4242(a)xxxxx.com . Or might need to look at fail2ban pike module to oget this but still had issues I just don’t know where to look or code to fix on SCENARIO B Snippets on my testing . route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.PBX.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.PBX.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + "kamaproxy"; # $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(); } #!endif --------------- So this led me to extra security should I host SIP extensions on kamailio as an extra security but then what about MWI/inbound then) or do I have 1100 on kama which mitel is 1100 but I pass all info from inbound.

2 1

Kamailio Query
by Amit Pal 15 Aug '20

15 Aug '20

Dear Denial/Team, I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. MY first question: When subscriber kamuser1 and kamuser2 are talking then I want to transfer these two party in freeswitch conference. I am unable to generate re-invite these call . here I was also try with rewritehost(like) but not success. how can we achieve this. My second question : Even MOH not playing when any one party hold the call. I already setup KAMUser1Kamailio(RTPProxy)KAMUser2. I have completely setup the config of rtpproxy and mohqueues. log shows all signal handle by kamailio and RTPProxy but not getting moh. I am attaching my config file for reference. can we do this by mediaproxy with freeswitch ,all thing are configured as document but no success. My third question: Kamailio relay dtmf on sipinfo. How can we get dtmf in between the call and taking actions on it. I am highly thankful for your great support . Thanks Regards

1 0

Kamailio Team Attentions
by Amit Pal 15 Aug '20

15 Aug '20

Dear Team, “I am unable to reach you, my previous mail get bounce. I received bounce mail from sr-users. Below is my previous mail contains.” Last Sent Mail Contents I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. MY first question: When subscriber kamuser1 and kamuser2 are talking then I want to transfer these two party in freeswitch conference. I am unable to generate re-invite these call . here I was also try with rewritehost(like) but not success. how can we achieve this. My second question : Even MOH not playing when any one party hold the call. I already setup KAMUser1Kamailio(RTPProxy)KAMUser2. I have completely setup the config of rtpproxy and mohqueues. log shows all signal handle by kamailio and RTPProxy but not getting moh. I am attaching my config file for reference. can we do this by mediaproxy with freeswitch ,all thing are configured as document but no success. My third question: Kamailio relay dtmf on sipinfo. How can we get dtmf in between the call and taking actions on it. I am highly thankful for your great support . Thanks Regards Neol

1 0

Required Kamailio clarification!!!
by Amit Pal 15 Aug '20

15 Aug '20

Dear Team, I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. MY first question: When subscriber kamuser1 and kamuser2 are talking then I want to transfer these two party in freeswitch conference. I am unable to generate re-invite these call . here I was also try with rewritehost(like) but not success. how can we achieve this. My second question : Even MOH not playing when any one party hold the call. I already setup KAMUser1==>Kamailio(RTPProxy)==>KAMUser2. I have completely setup the config of rtpproxy and mohqueues. log shows all signal handle by kamailio and RTPProxy but not getting moh. I am attaching my config file for reference. can we do this by mediaproxy with freeswitch ,all thing are configured as document but no success. My third question: Kamailio relay dtmf on sipinfo. How can we get dtmf in between the call and taking actions on it. I am highly thankful for your great support . Thanks Regards Neol

1 0

Can the callid be modified/mapped when branching a call?
by Magnus Kelly 14 Aug '20

14 Aug '20

Good evening all, Can I ask what might be a basic question, but I would like to confirm if the CallID field can be modified/mapped when branching a call ? Thanks Magnus

2 2

How to check TLS versions available
by David Cunningham 14 Aug '20

14 Aug '20

Hello, Does anyone know of a method to check what TLS versions are available from Kamailio for clients to use? For example, is TLS 1.0 available, TLS 1.1, etc. Thanks in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

4 5

Segfault starting kamailio, function "registered"
by Jon Bonilla (Manwe) 14 Aug '20

14 Aug '20

Hi when I start kamailio in my dev system it fails to start and segfaults. Version: 5.4.0 CFG Line: if (registered("kam_location",$tu)) { Log: Aug 14 10:20:29 pekedev2 kamailio: CRITICAL: <core> [core/cfg.y:3588]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 382, column 34-36: function registered: parameter 2 is not constant#012 Aug 14 10:20:29 pekedev2 kernel: [1501045.504630] kamailio[21276]: segfault at 6e6f6974 ip 00005566607d8887 sp 00007ffd6761b240 error 4 in kamailio[55666079d000+431000] Aug 14 10:20:29 pekedev2 kernel: [1501045.504655] Code: 89 bd 68 ff ff ff 64 48 8b 04 25 28 00 00 00 48 89 45 c8 31 c0 48 83 bd 68 ff ff ff 00 0f 84 f5 09 00 00 48 8b 85 68 ff ff ff <8b> 00 83 f8 01 0f 85 65 09 00 00 48 8b 85 68 ff ff ff 8b 40 0c 85 The coredump is not clear to me. I extracted using coredumpctl (fuck you systemd) and running gdb. Is there a new method now that systemd has taken control of my systems? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

conditional on secfilter seems does not work or how to do it
by PICCORO McKAY Lenz 14 Aug '20

14 Aug '20

I implemented secfilter in a simple way, in first step routing put that conditional for black list check that already works: ``` secf_check_ip(); if ($? == -2) { xlog("L_ALERT", "$rm from $si is blacklisted"); drop(); } ``` BUT NOW i want to change to whitelick checks, so i reviewed the docs and "2" is resulting for whitelist, so then i said "if not whitelisted so block" using "!= 2" as on https://www.kamailio.org/wiki/cookbooks/5.3.x/core#operators BUT SEEMS DOES NOT WORK: that is the code: ``` secf_check_ip(); if ($? != 2) { xlog("L_ALERT", "$rm from $si is not in whitelist, block"); drop(); } ``` when i tested all the calls passed not matter if are in the table or not as whitelist! I want able to call only if are present and whitelisted the ip address.. so i also tested with: ``` secf_check_ip(); if ($? == 0) { xlog("L_ALERT", "$rm from $si is not present, so block"); drop(); } ``` to check if the ip address are presented in the table but call also passed and must not cos i not put never a entry in the table! Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 3

Kamailio drop call with Teams
by sip user 13 Aug '20

13 Aug '20

Hi, I'm connecting Teams with kamailio server. From Kamailio to teams I have no problems, but from teams to Kamailio yes. Drop the call.. With ngrep I see that: INVITE: [image: imagen.png] 200 OK: [image: imagen.png] I never received ACK.. In my configuration: Kamailio.cfg: #!KAMAILIO #!define WITH_TLS event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:FQND:5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } request_route{ remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","$fU is trying to call to $rU con valores $tu\n"); $ru = "sip:" + $rU + "@ip asterisk:port"; route(RELAY); } } What I'm doing wrong? I don't understand why not received ACK.. Could anyone help me? Thanks

1 0

Kamailio lookup via Rest Interface
by krishdinesh＠yahoo.com 13 Aug '20

13 Aug '20

Hi, I am trying to do the following and wanted to seek help on how this can be done in Kamailio and if it feasible to do this. If not, whether there are other ways to do this. I would like to do the following .. If there is an incoming call, kamailio should be able to send the incoming calling number to an external application via Rest Interface, the external application will return a response on what to do with this call, it can be simple as sending a call to the called extension or a media server like asterisk for further treatment. I have a kamailio version 5.3.5 running on CentOS 7. I have a few end points registered to kamailio and they are able to call each other Any help would be appreciated. Thank you. Dinesh

1 0

Kamailio v5.4.0 RPM Availability
by Asgaroth 13 Aug '20

13 Aug '20

Hi all, I would just like to check in and find out if/when the Kamailio v5.4.0 rpm's will be available on the OpenSUSE build service. Thanks to all involved for all the hard work in this new release! Thanks There are two major products that came out of Berkeley: LSD and UNIX. We don't believe this to be a coincidence. -- Jeremy S. Anderson 00asgaroth00(a)gmail.com (https://link.getmailspring.com/link/146DE51A-E8C2-4126-B8D5-EA45E62CBA11@ge…)

4 14

smsops module - status report?
by Pafel 13 Aug '20

13 Aug '20

Hello, I am trying to generate sms status report using smsops module but without success. I couldn't find any example on the net. Can somebody help me? According to https://en.wikipedia.org/wiki/GSM_03.40 TP-MTI (tpdu(type) in kamailio) should be "1 0" for status report. Delivery is "0 0" and submit "0 1". According to the provided examples in smsops module: tpdu(type) = 4 is delivery tpdu(type) = 1 is submit So what's the number for status? I could't find the mapping in the code. Only this: // Types of the PDU-Message typedef enum _pdu_message_type { DELIVER = 0x00, SUBMIT = 0x01, COMMAND = 0x02, ANY = 0x03, } pdu_message_type_t; So any hint or example is appreciated. Thanks, Pavel Siderov

1 0

Log-level Kamailio IMS
by Pavithra Mohanraja 13 Aug '20

13 Aug '20

Hi, Is there any way to have global log level changes without restart of kamailio. If there kindly let me know. Thanks, Pavithra

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