sr-users March 2021

sr-users@lists.kamailio.org

61 participants
67 discussions

ERROR: connect_unix_sock: connect(/var/run/kamailio//kamailio_ctl): No such file or directory [2]
by Fadi Hamdan 06 Apr '21

06 Apr '21

Hi there, When I try to run: kazoo-kamailio status I'm getting this error: ERROR: connect_unix_sock: connect(/var/run/kamailio//kamailio_ctl): No such file or directory [2] Any solution for this error? Cheers, Hamdan, Fadi 24B Carrick Glen Avenue, Flat Bush, Auckland 2019 New Zealand fadi.h(a)msn.com<mailto:%20fadi.h@msn.com> M +64 21 066 0088

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SDP Change
by Nick Meyjes 06 Apr '21

06 Apr '21

Hi, I am trying to send calls to a provider that is switching to Nokia SBCs. They dont like seeing any IP other than our kamailio SBC in the messaging, I changed the contact but I cant find a way to change the owner line in the SDP I have try mangler and SDPOPS to change but neither seem to work on changing or removing the owner line Any ideas? Nick

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How to handle SIP-I(ISUP) with Kamailio
by Waseem Zaidi 06 Apr '21

06 Apr '21

Hi All: I am new to the list, I am struggling to find out Kamailio's capabilities with SIP-I/ISUP. I see that there is SIP-T module in kamailio and ss7ops module, but I need to get a hold on how to start. I would appreciate if someone could guide me. previously, I tried to use OpenSIPS SIP-I to extract ISUP from incoming calls, i was able to extract ISUP parameters in SDP body as xlog("Called Number: $isup_param_str(Called Party Number|Address signal)\r\n"); xlog("Calling Number: $isup_param_str(Calling Party Number|Address signal)\r\n"); xlog("Location is: $isup_param_str(Location Number)\r\n"); Location parameter is appearing as 0x04971400330268 which is wrongly interpreted. its proper interpretation should appear as 4100332086. I also tried below: xlog("Location is: $isup_param_str(Location Number|Address signal)\r\n"); but this subfield is not present. The transformation didn't make any difference: xlog("Location NUM: $(var(isup_body){isup.param, Location Number} Help pleeez ?

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Kamailio + Websocket issues
by Vinicius Kwiecien Ruoso 01 Apr '21

01 Apr '21

Hi! I'm using Kamaio in front of multiple Asterisk instances. At this moment it works as a SIP over Websocket proxy, with rtpengine, for browser clients to connect to Asterisk using WebRTC. I do not use the registration module of Kamailio, as each backend Asterisk is independent and handles its own registrations. Everything works great when making calls from the browser, and the routing is correctly executed by Kamailio based on the request SIP domain. We have an internal routing API that it calls to discover which backend Asterisk to route the calls. The issue I have is when a call initiates from that backend Asterisk, trying to reach a contact that is connected in Kamailio via the websocket. The Asterisk sends the message to the proxy, and Kamailio must route it to the corresponding websocket. I've tried a few approaches: - using add_contact_alias + handle_ruri_alias: I have the alias with alias=<ip>~<port>~ws in the contact registration, but for some reason handle_ruri_alias cannot use it - using the Path module on Asterisk, so when registering, the path is recorded and sent back from Asterisk, Kamailio is also not respecting that - Using contact_param_encode and contact_param_encode and contact_param_decode_ruri, but the encoded sip address is always the invalid websocket, like sip:58c0ktrg@5hp0nn5hqqv9.invalid;transport=ws None with success. Any hints on that can be wrong? I can share more detailed information. Greetings, Vinicius

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tcp timeout in kamailio
by Alexander Kudrevatykh 01 Apr '21

01 Apr '21

Hi. Is there an option for tcp timeout in kamailio? In my setup we have tcp peers and kamailio pings them every 5 seconds, but when tcp session stucks kamailio not close this session for 15 minutes. I tried many tcp options kamailio supports, but any of them helps. I think kamailio can use TCP_USER_TIMEOUT on linux to detect such tcp sessions and reestablish failed connections?

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dispatcher algorithm 11, no more than 25 hosts in group for proper distribution
by Ivan Matvieiev 31 Mar '21

31 Mar '21

Hello Community, There is a module dispatcher and sharing algorithm 11 that features the ability to redistribute load when the host is enabled or disabled that is quite valuable for me. In the test lab I faced unexpected behavior: - when I add to group <=25 hosts algorithm behaves as per description. - when a group includes 26 or more hosts, and 100 calls are done for instance, then about 25 go to the first host and 75 are distributed among the rest of hosts. Expected behavior: all hosts of the group having the same rweight value must receive the same amount of calls no mater what number of hosts is in the group. Could you please to help to figure out what is wrong and why when the dispatcher group consists of 26+ hosts dispatching algorithm 11 behaves differently than when group consists of 25 or less of hosts? Some configuration notes: # module configs modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_threshold", 3) modparam("dispatcher", "ds_inactive_threshold", 10) modparam("dispatcher", "ds_probing_mode", 3) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_ping_reply_codes", "501,403,404,400,200") modparam("dispatcher", "ds_ping_from",DS_PING_FROM_PARAM) modparam("dispatcher", "use_default", 0) # dispatching algorithm call if ( ds_is_from_list("101")) { sl_send_reply("100","My calls"); ds_select_dst("100", "11"); return; } # dispatcher list group config 100 sip:10.60.27.123:7000 0 10 rweight=50 100 sip:10.60.27.123:7001 0 10 rweight=50 100 sip:10.60.27.123:7002 0 10 rweight=50 100 sip:10.60.27.123:7003 0 10 rweight=50 100 sip:10.60.27.123:7004 0 10 rweight=50 100 sip:10.60.27.123:7005 0 10 rweight=50 100 sip:10.60.27.123:7006 0 10 rweight=50 100 sip:10.60.27.123:7007 0 10 rweight=50 100 sip:10.60.27.123:7008 0 10 rweight=50 100 sip:10.60.27.123:7009 0 10 rweight=50 100 sip:10.60.27.123:7010 0 10 rweight=50 100 sip:10.60.27.123:7011 0 10 rweight=50 100 sip:10.60.27.123:7012 0 10 rweight=50 100 sip:10.60.27.123:7013 0 10 rweight=50 100 sip:10.60.27.123:7014 0 10 rweight=50 100 sip:10.60.27.123:7015 0 10 rweight=50 100 sip:10.60.27.123:7016 0 10 rweight=50 100 sip:10.60.27.123:7017 0 10 rweight=50 100 sip:10.60.27.123:7018 0 10 rweight=50 100 sip:10.60.27.123:7019 0 10 rweight=50 100 sip:10.60.27.123:7020 0 10 rweight=50 100 sip:10.60.27.123:7021 0 10 rweight=50 100 sip:10.60.27.123:7022 0 10 rweight=50 100 sip:10.60.27.123:7023 0 10 rweight=50 100 sip:10.60.27.123:7024 0 10 rweight=50 Kamailio version: kamailio -v version: kamailio 5.4.4 (x86_64/linux) e16352 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: e16352 compiled on 15:56:46 Feb 15 2021 with gcc 4.8.5 System info CentOS Linux release 7.7.1908 (Core) Linux test-carrier-1.loc 3.10.0-1062.4.1.el7.x86_64 #1 SMP Fri Oct 18 17:15:30 UTC 2019 x86_64 x86_64 x86_64 GNU/Linux

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communication between two Kamailio
by Gustavo Monteiro 31 Mar '21

31 Mar '21

Hello, I need to do the simple communication between two Kamailio, ex: [Side A - K1 (10.0.0.0/24)- UE numbers 500] <----------> [Side B - K2 (192.168.0.0/24) UE numbers 400]. I already looked in the documentation, and I was surprised that I didn't find anything like that. I don't know which module or configuration to use. Thanks Gustavo (BR) Legal Disclaimer : The information contained in this message may be privileged and confidential. It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete or destroy any copy of this message!

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How to Play RBT With Media Server
by amitsharma＠coraltele.com 30 Mar '21

30 Mar '21

Dear Team, I am using SBC as proxy server and I am using Freeswitch for providing RBT/MOH. Please guide how can I play RBT from Freeswitch while using SBC as proxy server. Thanks & Regards, Amit Sharma (Sr. Team Leader) (An ISO 9001:2008 company) Mobile: <tel:9891612004> tel:9891612004 PH: +91 120 2595870 Ext.: <tel:870> tel:870 Email : <mailto:amitsharma@coraltele.com> amitsharma(a)coraltele.com Web : <blocked::http://www.coraltele.com> www.coraltele.com

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Strange behavior with $T_branch_idx in failure_route
by Victor Litvinov 25 Mar '21

25 Mar '21

Hello. We faced an issue that $T_branch_idx called from failure_route returned a strange result. Few words about our setup. We use branch indexes like key in avp when we store via-branch value for rtpengine. route[RTPMANAGE] ... if ( is_request() and is_method("INVITE") and !has_totag() and t_is_branch_route() ) { $avp(viabranch$T_branch_idx) = $branch_str; } $avp(extra_branch_id) = $avp(viabranch$T_branch_idx); $var(rtpkey) = $var(rtpkey) + " via-branch=extra"; ...... rtpengine_manage($var(rtpkey)); Call this route in branch_route and in failure_route. In branch_route T_branch_idx is 0, in failure_route is 1. In documentation: the index (starting with 1 for the first branch) of the branch for which is executed the branch_route[]. If used outside of branch_route[] block, the value is '0'. But in source code behavior is different. (modules/tmx/t_var.c:480): for BRANCH_FAILURE_ROUTE and BRANCH_ROUTE it is idx = tcx->branch_index, but for FAILURE_ROUTE it is idx = t->nr_of_outgoings + nr_branches; Can you describe why for failure_route it is different and what we can do in our case? And can you update documentation for this variable if it is wrong?

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Using sngrep for visualising encrypted SIP traffic
by Ovidiu Sas 24 Mar '21

24 Mar '21

Hello all, Here's a quick recipe to ease the troubleshooting of encrypted SIP traffic: https://voipembedded.wordpress.com/2021/03/22/troubleshooting-kamailio-encr… Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

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sr-users March 2021