sr-users April 2015

sr-users@lists.kamailio.org

89 participants
143 discussions

by renuka ghate

I have configured the kamailio server for TLS.. but my soft phone supports only DTLS, how do i configure kamailio to support DTLS. regards, Renuka

9 years, 7 months

Re: [SR-Users] Avoiding "script writer didn't release transaction" warnings

by Mickael Marrache

I think understand in which case it happens. When I receive a REGISTER, I create a transaction since I need to use the t_continue feature to wake up a suspended transaction. However, the save() command of the REGISTRAR module doesn't seem to call t_reply for replying the 200 (it looks like the SL module is used instead). Therefore, I get the warning every time it happens since the transaction is not released explicitly. Should I call t_release just before exiting the block handling the REGISTER? From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Thursday, April 09, 2015 1:43 PM To: sr-users(a)lists.sip-router.org Subject: Avoiding "script writer didn't release transaction" warnings Hi, In my logs, I get a lot of the following warnings: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction I force transaction creation from my script to absorb retransmissions as soon as possible. I understand that I need to explicitly call t_release in order to avoid this warning. What is the most appropriate place in the script to call t_release? Thanks, Mickael

9 years, 7 months

Avoiding "script writer didn't release transaction" warnings

by Mickael Marrache

Hi, In my logs, I get a lot of the following warnings: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction I force transaction creation from my script to absorb retransmissions as soon as possible. I understand that I need to explicitly call t_release in order to avoid this warning. What is the most appropriate place in the script to call t_release? Thanks, Mickael

9 years, 7 months

Work with radius

by Никитенко Виталий

Hi. I have scheme: User -> Kamailio -> Asterisk -> PSTN | V Radius Server At the beginning dialog (INVITE) kamailio send to radius server START packet which indicates Called, Calling, Accounting Session ID, connect_time, setup_time. When the terminating call (BYE) kamailio forms STOP package. How can I get on kamailio Called, Calling, connect_time that I have in kamailio for START packet if I only match Accounting Session ID? Thank.

9 years, 7 months

missing BYE when 2 redundant kamailio servers share the same database

by Ding Ma

Hi, all I'm trying to set up 2 kamailio servers for active-active redundancy. The two kamailio severs share the the same database with db_mode=3, and no registration replication. Use pjsua2 as SIP client for testing. The test setup is as follows: kamailio server 1(k1): 10.0.1.30:5061 <http://10.0.1.30:5061/> kamailio server 2(k2): 10.0.1.32:5061 <http://10.0.1.32:5061/> sip client c1: sip:16317@10.0.1.30 <mailto:sip%3A16317@10.0.1.30>, client ip: 10.0.1.254 sip client c2: sip:72316@10.0.1.30 <mailto:sip%3A72316@10.0.1.30>, client ip: 10.0.1.254 c1 is registered with k1. c2 is registered with k2. When c1 calls c2, the call flow is as follows: INVITE: c1------>k1------>k2------>c2 200 OK: c1<------k1<------k2<------c2 So far, the INVITE works. c1 and c2 can establish a call. However, when c2 hangs up and sends BYE, the BYE is forwarded all the way to k1, and then k1 gives a 500 server error. The log on k1 seems to indicate k1 was attempting to create a TLS connection to itself instead of using the existing TLS connection to c1 to forward the BYE. BYE: c1 X k1<------k2<------c2 | |------->Connection refused/500 Here is the kamailio log on k1: Mar 16 10:34:21 kamailio.sip.com <http://kamailio.sip.com/> /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 10.0.1.30:55999 <http://10.0.1.30:55999/> failed (RST) Connection refused Mar 16 10:34:21 kamailio.sip.com <http://kamailio.sip.com/> /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 10.0.1.30:55999 <http://10.0.1.30:55999/>: connect & send for 0x7fc96a68a1a0 failed: Connection refused (111) This is the routing logic for BYE in kamailio.cfg: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } else {...... If I add location based routing for BYE from peer kamailio, then the BYE seems to be forwarded correctly: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails # If BYE coming from Kamailio peer, route BYE by location $var(peerlist)=$sel(cfg_get.trusted.peers); $var(i) = 0; while($var(i)<$(var(peerlist){param.count})) { xlog("L_DBG", "=====$(var(peerlist){param.count})=====$(var(peerlist){param.valueat,$var(i)})=====\n"); if(src_ip==$(var(peerlist){param.valueat,$var(i)})) { lookup("location"); xlog("L_DBG", "=====BYE from $fu $si:$sp to $du=====\n"); break; } $var(i) = $var(i) + 1; } } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } else {...... My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces will be sent later to avoid exceeding email size limit. Thanks

9 years, 7 months

rtpstat counters

by Andres

In the documentation at: http://kamailio.org/docs/modules/4.2.x/modules/rtpproxy.html I see a variable called $rtpstat which "Returns the RTP-Statistics from the RTP-Proxy. The RTP-Statistics from the RTP-Proxy are provided as a string and it does contain several packet-counters." But I cannot find out where exactly it describes the packet counters. For example the xlog entry: log("L_ALERT", "STATS-rtpproxy,$rtpstat\n"); Will yield an output like: STATS-rtpproxy,60 596 660 1256 0#012 I need to find the documentation for those counters. Any ideas? Thanks, -- Technical Support http://www.cellroute.net

9 years, 7 months

Variable interpolation in xavp identifiers

by Alex Balashov

Hello, There seems to be an implementational difference in interpolation of PVs in AVP and XAVP identifiers and/or field names. The pseudovariables documentation[1] on classical AVPs says: The 'id' can be: ... pseudo variable - if value of pv is integer, id is integer, if string, id is string So, for example, this composition of variables works: $var(f) = 'donkey'; $avp($var(f) = 'eehaw'; xlog("L_INFO", "Donkey says: $avp(donkey)\n"); It correctly prints: Donkey says: eehaw Making some assumptions, I tried this, and it did not work similarly as expected: $var(f) = 'donkey'; $xavp(r=>$var(f)) = 'eehaw'; xlog("L_INFO", "Donkey says: $xavp(r=>donkey)\n"); This prints: Donkey says: <null> And a PV dump confirms that the PV value is not interpolated: Mar 31 18:26:15 localhost /usr/local/sbin/kamailio[2399]: INFO: debugger [debugger_api.c:1799]: dbg_dump_json(): {"$xavp(r)":[{"$var(f)":["eehaw"]}]} Interpolation of variables does not work for either field or stem names. Is this a design principle of XAVPs, or an oversight? Thanks, -- Alex [1] http://www.kamailio.org/wiki/cookbooks/4.2.x/pseudovariables#avp_id_-_avps -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

9 years, 7 months

BYE not forwarded

by Grant Bagdasarian

Hello, I have the following setup: UAC (Bria) ------ Kamailio -------- Freeswitch Kamailio is used to authenticate the client using basic authentication. Upon authentication the INVITE is sent to Freeswitch for further handling. Kamailio stays in the path using record-route. At this point, the authentication and calling FS works fine. For now, FS just answers, plays a file and hangsup the call. Answering the call and playing a file works fine. When FS hangs up the call, the BYE message is sent back to Kamailio. The problem is the BYE message from FS contains a Route header which points to Kamailio again. Which causes Kamailio to send the BYE message back to itself. Kamailio then replies with a 404. The BYE is never sent to the UAC. When I remove the record_route() for INVITEs, the call works fine, but Kamailio isn't kept in the signaling path. The BYE is sent directly to the UAC. T 10.0.5.52:54822 -> 10.0.5.50:5060 [AP] BYE sip:grant@10.0.0.40:5060;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 10.0.5.52;rport;branch=z9hG4bK99X8a4Q8BFy1a. Route: <sip:10.0.5.50;transport=tcp;lr=on>. Max-Forwards: 70. From: <sip:123456789@10.0.0.40>;tag=aey7rKe7DmQte. To: <sip:grant@10.0.0.40>;tag=dbc9c27b. Call-ID: NmQ4YjYyMDFkNmMwZjg0ZmU5NjQ3ODhjYzU0MDBmNGI. CSeq: 73909300 BYE. User-Agent: CM SBC. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY. Supported: timer, path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. Here is the WITHINDLG route: route[WITHINDLG] { xlog("L_INFO", "[R-ROUTE-WITHINDLG:$ci] !>" "Route:WITHINDLG;UnixTime:$TV(un);RequestURI:$rU;RequestMethod:$rm;FromTag:$ft;ToTag:$tt;CallID:$ci;From:$fU;To:$rU\r\n"); if (!has_totag()) { return; } # Sequential request within a dialog should take the path determined by record-routing if (loose_route()) { route(RELAY); exit; } else { if (is_method("ACK")) { if (t_check_trans()) { # No Loose-Route, but stateful ACK. Must be ACK after 487 route(RELAY); exit; } else { # ACK without matching transaction exit; } } sl_send_reply("404","Not here"); } exit; } I'm using TCP as the transport. Kamailio: 4.2.3 Freeswitch: 1.5.15 I've attached a SIP trace from the point of view of Kamailio. UAC: 10.0.0.40 Kamailio: 10.0.5.50 Freeswitch: 10.0.5.52 What is going on here? The whole Record-Route thing confuses me a lot. Regards, Grant

9 years, 7 months

Fritzbox issues kamailio 4.0.7 regarding stale nonce?

by Daniel Tryba

Did anyone notice issues with Fritzbox 7390 devices? When upgrading from 4.0.1 to 4.0.7 some of these devices fail to register after some time. They keep trying to reuse an old nonce and ignore the new one in the 401 response to the REGISTER. It will try 11 times to register with the same cseq, nonce and nc, then it will try to register (11 times) with cseq+1, nc+1 and still the old nonce. So www_auth fails with -4: stale nonce. I could find anything in kamailio auth module to accept this behavior. -- Telefoon: 088 0100 700 Sales: sales(a)pocos.nl | Service: servicedesk(a)pocos.nl http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel 17097024

9 years, 7 months

Kamailio V 2.4.2 Dialog_ng crash and IMS_USRLOC_SCSCF doc

by Yasin CANER

9 years, 7 months

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sr-users April 2015