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37519 discussions

cnxcc doesn't terminate calls
by Igor Potjevlesch 26 Oct '16

26 Oct '16

Hi, I'm testing cnxcc module in order to make a simple call duration limitation. My understanding is to use the function like this : $dlg_var(customer_id) = "customer-" + $fU; $var(max_time) = 5; if (!cnxcc_set_max_time("$dlg_var(customer_id)", "$var(max_time)")) Unfortunately, after 5 seconds, nothing happen. I don't even see an error from dialog module. Anyone has a better understanding of the way the module runs? Thank you! Regards, Igor.

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app_perl modifies Via header?
by Igor Potjevlesch 25 Oct '16

25 Oct '16

Hello, I try to use app_perl and cnxcc module. As soon as I load the module like this: # --------- app_perl params ---- modparam("app_perl", "filename", "/usr/local/etc/kamailio/AAA/aaa-script.pl") modparam("app_perl", "modpath", "/usr/local/lib64/kamailio/perl/") The Via header is modified for each calls: /usr/local/sbin/kamailio[21064]: INFO: <core> [parser/msg_parser.c:355]: parse_headers(): ERROR: bad header field [Via:ASIP/2.0/UDP 91.] /usr/local/sbin/kamailio[21064]: ERROR: <core> [parser/parse_via.c:2048]: parse_via(): ERROR: parse_via: bad char <A> on state 100 What could explain this behaviour? Regards, Igor.

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Kamailio SIP TLS issue
by Sergey Basov 25 Oct '16

25 Oct '16

Hi All I have some strange behavior of kamailio with TLS. I have configured second IP addres on server, added it to tls listener, and tls.cfg file. But when I try to connect using openssl s_client -showcerts -connect 10.1.23.33:5061 -tls1 -state and openssl s_client -showcerts -connect 10.1.23.23:5061 -tls1 -state I see same certificates (sip2 my config samples are bellow) if I make changes in port number (for ip 10.1.23.33 set port 5091 in both config parts) - I see correct certificates. Does anyone have this problem? Thanks in advance. ----- listen section ---- listen=tls:10.1.23.23:5061 listen=tls:10.1.23.33:5061 ----- tls.cfg ------ [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip1.key certificate = /etc/kamailio/keys/sip1.crt [server:10.1.23.33:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip1.key certificate = /etc/kamailio/keys/sip1.crt [server:10.1.23.23:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip2.key certificate = /etc/kamailio/keys/sip2.crt -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

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Sending INVITE’s on REGISTER
by Igor Olhovskiy 25 Oct '16

25 Oct '16

Hi! I have this kind of scenario Receive an INVITE -> send it to user, but if possible, if I’ll receive REGISTER with same credentials I’m calling to - drop an INVITE there also, if call was not answered on first location. Means create other INVITE branch async on the fly, if possible. Where at least to look at? Thanks. -- Best regards, Igor

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How compile Kamailio in freebsd : add carrierroute module
by AFRICALLSHOP 25 Oct '16

25 Oct '16

Hello, I try to compile kamailio in freebsd. I want to add carrierroute module. I do : 1) cd /usr/ports/net/kamailio 2) make install but carrierroute module is not compiled. Could you help me? Regards Abdoul.

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How compile Kamailio in freebsd : add carrierroute module
by Abdoul Osséni 24 Oct '16

24 Oct '16

Hello, I try to compile kamailio in freebsd. I want to add carrierroute module. I did : 1) cd /usr/ports/net/kamailio/ 2) make install but carrierroute module is not compiled. Could you help me? Regards Abdoul.

1 0

Rewriting To on BYE
by Charles Phillips 24 Oct '16

24 Oct '16

I need a sanity check if you don’t mind… I am setting up a stateless proxy for several (separate, not load balanced) Asterisk servers behind Kamailio 4.2 (with domain and dispatcher). All works as expected except when called party hangs up, as Asterisk seems to address the BYE “To” (that is sent to the calling party) to the originating domain, and not the IP of the UA (as it does in other cases). When this happens, stateless Kamailio does not seem to know how to route them, which makes sense. I assume I would need to rewrite the “To” field on BYE message so that it contains the UA’s IP (to match the BYE’s that are routed correctly). The RURI header contains the IP/port of the UA, so I am attempting to use it and here is where I am stuck… Here is the subst from onreply_route: if (is_method("BYE")) { xlog("L_NOTICE", "BYE Detected \n"); subst('/^To:(.*)<sip:(.*)@(.*)>(.*)$/To: <sip:\2@$rd:$rp>\4/ig'); } Here is the “To”: To: "Test User 2" <sip:102@zxcv.asdf.qwerty.net>;tag=8AB53900B9646F1DF05330E7FBF846C8 Any obvious glaring mistakes here? Would I then need additional logic to forward the BYE on? My apologies if this is wordy, or nonsensical...

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PCSCF cannot create via header for sipml5 ACK package
by Serhat Guler 24 Oct '16

24 Oct '16

Hello, I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says: 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5. The ACK package: ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB To: <sip:alice@net1.test>;tag=18823 Contact: "Bob"<sip:bob@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> Max-Forwards: 69 Route: <sip:mo@192.168.0.11:880 ;transport=ws;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:4060 ;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:4060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.1c3> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached. Thanks in advance, Serhat

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configuration kamailio 4.4.2 with Asterisk 13.10.0
by Serge Gabriel Ahnoux 24 Oct '16

24 Oct '16

Hello I am trying to configure kamailio 4.4.2 with asterisk 13.10.0. My Os is Debian 7.4 I followed the tutorial : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb… The two softwares are installed on my server. So they have the same IP adress but the port is different like they said in the tutorial. I use too Mysql to stock users informations I have 2 databases Kamailio and Asterisk. I use ODBC for communicate beetwen asterisk server and his database. However when my sip client (linephone) connects to kamalio, kamailio doesn't forward information to asterisk. I don't kow the reason. So i can send message but not call If someone knows how to configure, please help me. I join my kamailio configuration file. Regards. ahnouxs(a)gmail.com

2 1

Dropping non-provisional replies in onreply_route
by Alex Balashov 21 Oct '16

21 Oct '16

The core documentation says that in a named onreply_route[], only provisional replies can be drop()'d. To drop any reply, it is necessary to use a global onreply_route. Is there any workaround for this, i.e. so I can drop a 2xx reply from a specific TM transaction? The reason is, to know whether to drop it, I need access to either AVPs or, ideally, dialog variables. Since the global onreply_route is executed by the core, I presume I won't have access to anything that persists through TM there. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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