sr-users November 2018

sr-users@lists.kamailio.org

68 participants
89 discussions

by Andrey Deykunov

Hi guys, I've implemented an rpc call that closes all WS connections: void ws_rpc_close_all(rpc_t *rpc, void *ctx) { ws_connection_t **list = NULL, **list_head = NULL; ws_connection_t *wsc = NULL; int ret; list_head = wsconn_get_list(); if(!list_head) return; list = list_head; wsc = *list_head; while(wsc) { LM_WARN("Closing connection\n"); ret = close_connection(&wsc, LOCAL_CLOSE, 1000, str_status_normal_closure); wsc = *(++list); } wsconn_put_list(list_head); } Please take a look. Can I call close_connection() and perform list iterations in this function not using WSCONN_LOCK? Thanks, Andrey

6 years

from Asterisk to Kamailio

by Valter Nogueira

Today, I use Asterisk as a SIP/RTP PROXY I proxy from customers Asterisks to a VOIP provider, in a multi-homed server. Now, I want to move to Kamailio without any rupture in customer's configuration. As anyone can imagine I am kind of lost. USER ACCOUNTS In Asterisk, I create a dynamic host account named ACCOUNT1 and I receive in *FROM HEADER sip:ACCOUNT1@customer_ip_address* In Kamailio, I have to define the account's domain like *kamctl add ACCOUNT1(a)mydomain.com <ACCOUNT1(a)mydomain.com> password. *Kamailio just accepts a REGISTER/INVITE from *ACCOUNT1(a)mydomain.com <ACCOUNT1(a)mydomain.com>* SIP/RTP PROXY In Asterisk, I just dialout to the VOIP PROVIDER like *dial (SIP/VOIP_ACCOUNT/${EXTENSION})* Asterisk does all the magic (it is a B2BUA). It bridges the new call and media to the original call. Moreover, user don't know anything about how call are completed, nor how credentials are setup and soon. In Kamailio, I guess that I have to use nat, tm and rtpproxy modules and maybe uac. I am not sure how to setup it. Can someone send me a clue? Thank you, Valter

6 years

Kamailio-5.1.6 Crash

by Hamid Hashmi

I have been using the following version of kamailio Version: kamailio 5.1.6 (x86_64/linux) 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: unknown Compiled with: gcc 4.8.5 Compiled on: 11:48:33 Oct 11 2018 and I noticed today that the Kamailio service is crashing. [root@kamailiostandby /]# gdb core.17805 GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-100.el7_4.1 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-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... [New LWP 17805] Core was generated by `/usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/SIP'. Program terminated with signal 11, Segmentation fault. #0 0x00007f636559e54b in ?? () "/core.17805" is a core file. Please specify an executable to debug. (gdb) list No symbol table is loaded. Use the "file" command. (gdb) bt #0 0x00007f636559e54b in ?? () #1 0x00007f63a9571810 in ?? () #2 0x0000000000000000 in ?? () Regards Hamid R. Hashmi

6 years

registrar::save() does not save 'received' field of the 'location' table

by Andrey Deykunov

Hi, We deploy kamailio in the cluster containing two nodes (Active and Passive). When Active node becomes Passive, our websocket clients continue sending REGISTER to this node. Passive node routes these requests to the Active node through the UDP, that causes the following warning message: 2018-11-14T04:43:31.070005-08:00 qapbx-ca1-1 kamailio[25597]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:64.78.52.88:5060, to ws:10.9.3.116:20033) Every registered client should be stored in the 'location': $var(save_val) = save("location", 0, "sip:$au@$td"); if ($var(save_val) <= 0) { xlog("L_ERR", "[$fU,R,$ci]: Registration saving failed, contact $ct"); sl_reply_error(); } else if ($var(save_val) == 3) { xlog("L_INFO", "[$fU,R,$ci]: Unregistration accepted. Login: $fU; ua: $ua; s-ip: $si; s-port: $sp; 3d parm: sip:$au@$td"); } else { xlog("L_INFO", "[$fU,R,$ci]: Registration accepted. Login: $fU; ua: $ua; s-ip: $si; s-port: $sp; 3d parm: sip:$au@$td"); } Log output of the Avtive node: 2018-11-14T04:43:31.063376-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: REGISTER received with R-URI: sip:64.78.52.90, ct: <sip:vu2o757j@si7ofk7i8cs7.invalid;transport=ws>;+sip.ice; reg-id=1;+sip.instance="<urn:uuid:00ff4c8d-2a5e-4614-b80a-d7c14c0e41e3>";expires=600, cseq=1, s-ip: 64.78.52.88, s-port: 5060 2018-11-14T04:43:31.063947-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: Our node is active 2018-11-14T04:43:31.064323-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: Authorization headers are empty in REGISTER command. Send 401 Unauthorized to 64.78.52.88 2018-11-14T04:43:31.285967-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: REGISTER received with R-URI: sip:64.78.52.90, ct: <sip:vu2o757j@si7ofk7i8cs7.invalid;transport=ws>;+sip.ice; reg-id=1;+sip.instance="<urn:uuid:00ff4c8d-2a5e-4614-b80a-d7c14c0e41e3>";expires=600, cseq=2, s-ip: 64.78.52.88, s-port: 5060 2018-11-14T04:43:31.286564-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: Our node is active 2018-11-14T04:43:31.286943-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: Registration accepted. Login: 54004221; ua: JsSIP 3.2.11; s-ip: 64.78.52.88; s-port: 5060; 3d parm: sip:54004221@qapbx-ca1.qaserverdata.net but in this scenario the save() method stores null value in the 'received' row. How can I fix it? Thank you, Andrey

6 years

RTPengine: shouldn't SDP <sess-version> be incremented when SDP is updated in-dialog?

by George Diamantopoulos

Hello all, I have the following issue (I think) with RTP engine. An INVITE comes in, and rtpengine will rewrite the SDP accordingly, as configured in kamailio.cfg. After some time a reINVITE is sent out in the opposite direction, for session refresh purposes. As I use rtcp-mux-offer in kamailio.cfg for this direction, RTPengine will inject the rtcp-mux parameter, and this reINVITE is forwarded to the UAC that sent the original INVITE. However, the SDP in the reINVITE is exactly the same as the SDP in the 200 OK to the original INVITE, with the exception of the rtcp-mux parameter. Since the SDP offered from the same end has changed, shouldn't session version be incremented as well? Does this sound like something that should be reported as a bug to rtpengine? Or am I missing something here? Thanks! Best regards, George Diamantopoulos

6 years

Number of sql connections

by Duarte Rocha

Greetings, I have my core parameter listen like this : listen=udp:eth0 children=12 Like this and the other settings, my kamailio has 21 threads and 20 Sql connections. If I add more listens, listen=udp:eth0 listen=udp:eth0:5072 listen=udp:eth0:5074 Now I have 45 threads and 45 Sql connections. Why does this happen and can i set a a number of threads and/or sql connections that doesn't change with the number of "listen" parameters that i have ? Best Regards, Duarte Rocha

6 years

recording audio stream

by bill＠telopar.net

Hi All: I have a project that requires that I record the audio streams from an ongoing phone call. It seems that rtp_proxy module should do that, but I can't seem to get it to work. My config file: *if(nat_uac_test("8")) { //rtpproxy_manage("co"); xlog("L_INFO", "rtpproxy Call is going to private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("crwie","10.1.1.2"); } else { //rtpproxy_manage("cor"); xlog("L_INFO", "rtpproxy Call is coming from a private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("crwei","10.1.1.2"); } *There appears to be conversations going on with rtpproxy on both legs of the call. But I seem to be missing the step that records the audio*. *Anybody have an example of how to do this. TIA, Bill **

6 years

Which algorithm to use as load-balance

by David Villasmil

Hello, Is it possible to set an algorithm like the following? fs1 - Priority 10 - maxload 20 calls fs2 - Priority 10 - maxload 20 calls fs3 - Priority 9 - maxload 20 calls fs4 - Priority 8 - maxload 20 calls fs5 - Priority 7 - maxload 20 calls So that, whenever any new call comes in, fs1 and fs2 would be used, meaning the fs with the highest priority would be use UNLESS there's 20 calls on that server, in which case, the next server with the highest priority would be used. Thanks!, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

6 years

Freeradiusclient Interim-Update Support

by Hamid Hashmi

Dear All Is module acc_radius supports interim updates? If yes then how to configure it. Regards Hamid R. Hashmi

6 years

textops module get_body_part function gives boundary error even if not multipart

by YAS0 CANER

Hello, i try to get content from Invite request but it gives boundary error. it is not multipart or mixed content but it checks boundary and gives error. Thanks. My message INVITE Content-Type: application/resource-lists+xml <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list> <entry uri="100?Call-ID=$calli&From=$fromtag&To=$toTag&Session-ID=1"/> <entry uri="103?Call-ID=$calli2&From=$fromtag2&To=$toTag2&Session-ID=2"/> </list> </resource-lists> static int ki_get_body_part_helper(sip_msg_t* msg, str* ctype, pv_spec_t *dst, int mode) { ... if(get_boundary(msg, &boundary)!=0) { LM_ERR("Cannot get boundary. Is body multipart?\n"); return -1; } .... int get_boundary(struct sip_msg* msg, str* boundary) params.s = memchr(msg->content_type->body.s, ';',msg->content_type->body.len); if (params.s == NULL) { LM_INFO("Content-Type hdr has no params <%.*s>\n", msg->content_type->body.len, msg->content_type->body.s); return -1; }

6 years

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sr-users November 2018