- sr-users - kamailio.org

sip
by Nar 01 Aug '14

01 Aug '14

Hi, I am developing a social application where I want to enable voice & video calling function and still I don't know what to use in server side, do you have any product that I can use to develop the server side(for 10.000+ users)? Thank you in advance.

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[Kamailio]list members
by Djamel Bahamid 01 Aug '14

01 Aug '14

Hi, I would like to subscribe to the list. Best Regards, Djamel BAHAMID.

2 1

SIGUSR1 for memory status not working as documented - only one process reports back
by Alex Villacís Lasso 01 Aug '14

01 Aug '14

I am trying to track down a memory leak that was triggered by a patch I wrote for my local copy of kamailio 4.1.4 . For this, I am following the documentation at http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory . This page claims that once memlog is set in the configuration file, a kamailio process will dump a report of the allocation map when shutting down, or when receiving a SIGUSR1. I have configured my kamailio.cfg with memlog=1 and no other change, and I see the memory report on shutdown for all processes. However, when I send a SIGUSR1 to a kamailio process, the process does absolutely nothing, with the exception of the first kamailio process (the one reported as Type=attendant by "kamctl ps"). All of the other processes just ignore SIGUSR1. What is going on? It is inconvenient to force a shutdown of all the kamailio processes just to get the memory report.

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Unable to SUBSCRIBE for presence using sip.js through WSS
by Alex Villacís Lasso 31 Jul '14

31 Jul '14

My kamailio.cfg configuration file is attached. I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP), presence seems to work correctly. However, when using SIP.js via a websocket, Kamailio is unable to send the NOTIFY for the presence event. However, MESSAGE packets are routed correctly between peers, even through websockets. The location table after registration looks like this: +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ | id | ruid | username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | cflags | user_agent | socket | methods | instance | reg_id | +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ | 192 | uloc-53d96398-1afc-1 | admin | pbx.villacis.com | sip:nfinhooe@192.0.2.90;transport=ws | sip:192.168.3.2:41478;transport=WS | NULL | 2014-07-30 17:18:57 | -1.00 | 6a2dot9ihoobhng9ackqmq | 90 | 2014-07-30 17:08:57 | 0 | 64 | SIP.js/0.5.0 | tls:192.168.2.18:5061 | 782 | <urn:uuid:b8888e95-7833-46a4-a21e-8d2e809a66b8> | 1 | +-----+----------------------+----------+------------------+--------------------------------------+------------------------------------+------+---------------------+-------+------------------------+------+---------------------+-------+--------+--------------+-----------------------+---------+-------------------------------------------------+--------+ When running the following javascript code: var subs = ua.subscribe('admin(a)pbx.villacis.com', 'presence'); subs.on('notify', function (n) { alert(n.request.body); }); ... I see the following traffic in the browser console area: SUBSCRIBE sip:admin@pbx.villacis.com SIP/2.0 Via: SIP/2.0/WSS 192.0.2.90;branch=z9hG4bK3654279 Max-Forwards: 70 To: <sip:admin@pbx.villacis.com> From: "Administrator" <sip:admin@pbx.villacis.com>;tag=rt0men9qve Call-ID: pa7jno8q3op5pu0qitad CSeq: 322 SUBSCRIBE Proxy-Authorization: Digest algorithm=MD5, username="admin", realm="pbx.villacis.com", nonce="U9lswVPZa5WSu5jz1P/jIOJTvthKXw6v", uri="sip:admin@pbx.villacis.com", response="f7246e64573b5888d69896a70d4204aa" Event: presence Expires: 3600 Contact: <sip:nfinhooe@192.0.2.90;transport=ws> Allow: ACK,CANCEL,BYE,OPTIONS,MESSAGE Supported: outbound User-Agent: SIP.js/0.5.0 Content-Length: 0 SIP/2.0 202 OK Via: SIP/2.0/WSS 192.0.2.90;branch=z9hG4bK3654279;rport=41478;received=192.168.3.2 To: <sip:admin@pbx.villacis.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-84c3 From: "Administrator" <sip:admin@pbx.villacis.com>;tag=rt0men9qve Call-ID: pa7jno8q3op5pu0qitad CSeq: 322 SUBSCRIBE Expires: 3600 Contact: <sip:192.168.2.18:5061;transport=ws> Server: kamailio (4.1.4 (x86_64/linux)) Content-Length: 0 However, I also get the following messages in the kamailio log: Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [t_msgbuilder.c:1544]: build_uac_req(): build_uac_req(): Error while assembling Via Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: tm [uac.c:338]: t_uac_prepare(): t_uac: Error while building message Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [notify.c:1591]: send_notify_request(): in function tmb.t_request_within Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [notify.c:1678]: notify(): sending Notify not successful Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:678]: update_subscription(): Could not send notify Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:713]: update_subscription(): occured Jul 30 17:03:01 elx3 /usr/sbin/kamailio[6908]: ERROR: presence [subscribe.c:994]: handle_subscribe(): in update_subscription I have found a discussion at http://marc.info/?l=sr-users&m=136624316816288&w=2 which recommends fix_nated_contact(), but I am not really sure about how to add it, or what conditions to check on whether it should be used. I added the call to fix_nated_contact() like this: # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); # # 1 - Contact header field is searched for occurrence of RFC1918 or rfc6598 addresses. # 2 - the "received" test is used: address in Via is compared against source IP address of signaling # 16 - test if the source port is different from the port in Via # 64 - test if the source connection of signaling is a WebSocket # ----- # 83 #!ifdef WITH_WEBSOCKET # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. if (nat_uac_test("83")) { #!else if (nat_uac_test("19")) { #!endif if (is_method("REGISTER")) { fix_nated_register(); #!ifdef WITH_WEBSOCKET if (nat_uac_test("64")) fix_nated_contact(); #!endif } else { if (is_first_hop()) if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } setflag(FLT_NATS); } #!endif return; } After registration, "kamctl ul show" reports this: [root@elx3 js]# kamctl ul show Domain:: location table=512 records=1 max_slot=1 AOR:: admin(a)pbx.villacis.com Contact:: sip:85017jar@192.168.3.2:41777;transport=ws Q= Expires:: 584 Callid:: argl25kg0r4qe8lbtbhn9f Cseq:: 82 User-agent:: SIP.js/0.5.0 Received:: sip:192.168.3.2:41777;transport=WS State:: CS_NEW Flags:: 0 Cflag:: 64 Socket:: tls:192.168.2.18:5061 Methods:: 782 Ruid:: uloc-53d971c7-297f-1 Instance:: <urn:uuid:75464840-96b8-4e22-8eaa-5012ba327086> Reg-Id:: 1 Last-Keepalive:: 1406759390 Last-Modified:: 1406759390 However, it still does not work: Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: WARNING: <core> [msg_translator.c:2506]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [t_msgbuilder.c:1544]: build_uac_req(): build_uac_req(): Error while assembling Via Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: tm [uac.c:338]: t_uac_prepare(): t_uac: Error while building message Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [notify.c:1591]: send_notify_request(): in function tmb.t_request_within Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [notify.c:1678]: notify(): sending Notify not successful Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:678]: update_subscription(): Could not send notify Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:713]: update_subscription(): occured Jul 30 17:30:21 elx3 /usr/sbin/kamailio[10624]: ERROR: presence [subscribe.c:994]: handle_subscribe(): in update_subscription

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Obscuring SIP traffic and using with NoSIP
by Muhammad Shahzad 31 Jul '14

31 Jul '14

Hi, As the mobile voip is getting more and more popular these days, there has been a strong opposition from GSM operators against mobile voip apps. They often use tactics like blocking voip ports, or detect and block voip traffic and in some cases restricting udp traffic altogether to very low upload and download speeds. See below link for some details, http://www.linphone.org/eng/blog/linphone-over-3g.html While not all the problems can be solved right now (especially the limiting udp traffic, since RTP always uses udp transport) I was wondering if we can at least handle the sip related problems. The most important of them is SIP traffic detection. While some forks would suggest using TCP/TLS to encrypt SIP traffic, it has a few problems, e.g. 1. It requires somewhat high resources on mobile devices, so many low-end android phones simply can't use it. 2. There is possibility that encryption signature may identify it as SIP traffic. There exists firewalls (often deployed in middle eastern countries) which have huge database of encryption signatures and patterns which although may not decrypt the sip packet but at least identify it as sip packet and block it. Also with rough agencies of evil empires spying over millions of users worldwide makes the current encryption standards pretty much pointless, at least in terms of user privacy and network security. So there is a strong need to experiment with new ideas and concepts to regain internet freedom. Some of such ideas are, 1. Convert sip traffic which is plain text to binary format just before transmitting it and revert it to plain text upon reception. 2. XOR the sip traffic (pretty much same as binary sip). 3. Use some very lightweight but effective / non-standard encryption algorithm, e.g. https://github.com/mshary/itv All these ideas require that SIP server such as Kamailio is able to adopt to these, preferably with minimal or no change in native code. The NoSIP module seems an interesting module in this regard. It provides all traffic it thinks is not the SIP traffic to configuration script, where we can do our own parsing and do whatever we want with it. I have two questions about this, 1. If parsed message is SIP, we can we send it back to kamailio core to get it processed as if it is a normal SIP message received by kamailio? 2. Can this module or any other module available in kamailio, that can provide us full sip packet that is about to be transmitted over sip socket, so we can "encode" it just before it is sent to next hop? I know this would be like writing a SIP transport in kamailio script which would be very tough if not impossible to implement in native core. But it will really help in winning the modern mobile voip challenges. Thank you.

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[Serusers] how to enable logs in SER
by pawan 31 Jul '14

31 Jul '14

Sir pl help for registration of soft phone to my local sip server my email vijay8538(a)gmail.com --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

1 0

multi-homed rtpengine
by Paul Belanger 30 Jul '14

30 Jul '14

Greetings, We have had good success in the past using rtpproxy with internal and external interfaces however, we are attempting to migrate to rtpengine now but are running into an issue (apparently this feature is not supported). So, my question is as follows, does this mean we need to multiple rtpengine and bind them to each interface? Additionally, in this setup can audio be bridged between the 2 rtpengine processes? If not, how do people over come this difference? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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Re: [SR-Users] LCR
by Keith 30 Jul '14

30 Jul '14

Hi, I want to do LCR based on from uri. I get I need to do regular expression matching but have no idea where to look to start. Any pointers would be good. Thanks Keith

3 2

Warning message on SIP websocket
by Alex Villacís Lasso 29 Jul '14

29 Jul '14

Just a quick question. What is supposed to be wrong with the below message? Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/parse_fline.c:243]: parse_first_line(): parse_first_line: bad message (offset: 22) Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: <core> [parser/msg_parser.c:705]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Via: SIP/2.0/TLS 192.168.3.2:58657#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: Er/CgVyBNzVGREBmt+w3FTff8s4=#015#012Server: kamailio (4.1.4 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012>

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Plz help: MSILO on TLS + cert based client auth enabled server
by Marc M. 29 Jul '14

29 Jul '14

Hi, I have a SIP server with TLS + client certificate based authentication. The server is listening for SIP(UDP) on port 5060, and for SIPS(TLS) on port 5061) The client certificate based authentication is enforced on port 5061. I would like to fire up MSILO and face now a strange problem: At each REGISTER, the stored messages get dumped. Dumped messages get sent to the clients from senders logged in over port 5060(UDP) as SIP:xxxxxxxxx@xx.xx.xx.xx however Dumped messages are NOT sent to clients from senders logged in on port 5061(TLS) as SIPS:xxxxxxxx@xx.xx.xx.xx It looks like the messages get dumped so M-dump() is working, but they will never enter the route. I would assume TLS is stopping the dumped messages to entering the route logic. I would either 1. somehow force m_dump() to pipe the messages over SIP(and not SIPS), or 2. disable client auth for messages recieved from local ip Can you help me how to proceed? What would be the correct approach? Can you help me with either 1 or 2. You help is greatly appreciated!!! Marc

2 1

Only a redirect server
by Jack Smith 29 Jul '14

29 Jul '14

Dear friends, Just new to this mailing list, new to Kamailio, and new to running a sip server. I would like to try only a redirect server. I had a look at http://sourceforge.net/p/openser/code/HEAD/tree/trunk/examples/redirect.cfg which was friendly, but then the default file kamailio.cfg looked frightening ! Could someone help me set up a minimal Kamailio installation only to handle state-less redirect? Best, Jack

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append_branch and t_replicate issue
by Asgaroth 28 Jul '14

28 Jul '14

Hi All, I have an issue that I cant seem to get to the bottom of, I would appreciate if someone could point me in the right direction. I have 3 registrar's and would like to replicate the registration regest from 1 to the other 2, I am trying this with t_replicate, the docs say that to send to multiple destinations an append_branch should be performed prior to the t_replicate, which is what I am trying to do. What appears to happen is that the replication only sends the REGISTER to the server defined in the t_replicate command, it doesnt appear to be sending it to the destination defined in append_branch. Kamailio Version: version: kamailio 4.1.4 (x86_64/linux) 84c1ff flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, 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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 84c1ff compiled on 14:36:04 Jul 20 2014 with gcc 4.4.7 Here is what I have tried so far (on registrar 1): #!define BACKUP_REGISTRAR_1 '1.1.1.1' #!define BACKUP_REGISTRAR_2 '1.1.1.2' #!define BACKUP_REGISTRAR_3 '1.1.1.3' save("location"); if ( src_ip != BACKUP_REGISTRAR_2 && src_ip != BACKUP_REGISTRAR_3 ) { append_branch("sip:" + BACKUP_REGISTRAR_2 + ":5060"); t_replicate("sip:" + BACKUP_REGISTRAR_3 + ":5060"); }; With the above in place, I only see the system attempting to replicate to BACKUP_REGISTRAR_3, I dont see any request to BACKUP_REGISTRAR_2. Am I doing something wrong here? Thanks Bruce

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possible regression on 4.1.3 vs 4.1.4
by Kelvin Chua 26 Jul '14

26 Jul '14

using the default cfg file, whenever a WITHINDLG BYE is received, 4.1.4 needs a retransmission before passing it through, the first BYE would generate this in syslog tm [tm.c:1518]: _w_t_relay_to(): ERROR: w_t_relay_to: unsupported route type: 64 and the 2nd retransmitted BYE would go on fine. i believe route type 64 is branch failure, but using the same cfg and reverting back to 4.1.3 solves it. so whatever changes was done to 4.1.4 broke it. Kelvin Chua

2 4

Rtpengine vs. TURN?
by Peter Villeneuve 25 Jul '14

25 Jul '14

Hi, On my server, I have the option of using either Rtpengine for NAT traversal or pure TURN without rtpengine. Rtpengine has the obvious plus that it only needs 1 public IP, while TURN (with STUN) will need 2 public IPs, although that's not a problem in my case. Having said that, I'd like to take advantage of the huge experience that users of this list have in real world deployments. in your experience, which option is more reliable in a real world deployment? Cheers, Peter

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MSILO not checking Expires header if Allow: is absent and Contact: is present but without expires
by Anthony Messina 25 Jul '14

25 Jul '14

In setting up Kamailio with MSILO, I've noticed that some sip clients (I'm using CSipSimple), upon un-REGISTER-ing do not send an Allow header, but do send an "Expires: 0" header as well as a Contact header. However as the Contact header doesn't contain "expires=0", it appears as though the MSILO skips over checking the Expires header. As such, I've had to add the following workaround to my route[REGISTRAR] block or else these clients trigger the m_dump. # Ensure we aren't unregistering # Right now, CSipSimple doesn't send an Allow header with an un-REGISTER, # so Kamailio skips to checking the Contact header for expires, and appears # to ignore the fact that an Expires: 0 header is present. if($hdr(Expires) != 0) m_dump(); Looking through the msilo check_message_support function, it appears that it never checks for the Expires header if there is no Allow header and the Contact header does not specify an expire setting. I do believe that we're missing the check for the Expires header, regardless of whether or not there is an Allow header, or a Contact header with expires=0 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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407 Proxy Authentication Required with Cisco
by proLogika 25 Jul '14

25 Jul '14

Hello, I am using Kamailio as SIP register with asterisk integration describe from hire: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . asterisk is listen on public ip: 2.3.4.5:5060 kamailio is listen on public ip: 1.2.3.4:5060 everything is working well except some software VoIP clients (like Yate) and CISCO phone like Cisco-CP7940G/8.0 and the new one from Cisco series. I’m testing now with Yate client and Cisco. They are register OK but when a call is made Kamailio is answer back with 407 Proxy Authentication Required. When I register Yate or Cisco to asterisk directly the call is passing normaly. I was trying to manipulate kamailio.cfg and more specifically the part: #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sipusers", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; If i commented out this part the call is passing, but I do not have auth anymore (everyone can register) Here is ngrep: U 2014/07/23 19:17:08.108458 192.168.0.40:5060 -> 1.2.3.4:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 20. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 INVITE. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. Content-Type: application/sdp. Content-Length: 481. . v=0. o=yate 1406132227 1406132227 IN IP4 192.168.0.40. s=SIP Call. c=IN IP4 192.168.0.40. t=0 0. m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. U 2014/07/23 19:17:08.108805 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=6be166fd53062bbc5b6dd79656b620cd.1950. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 INVITE. Proxy-Authenticate: Digest realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I". Server: kamailio (4.0.6 (x86_64/linux)). Content-Length: 0. . U 2014/07/23 19:17:08.128626 192.168.0.40:5060 -> 1.2.3.4:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=6be166fd53062bbc5b6dd79656b620cd.1950. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 ACK. Max-Forwards: 20. Contact: <sip:10891@192.168.0.40:5060>. User-Agent: YATE/5.3.0. Content-Length: 0. . U 2014/07/23 19:17:08.129076 192.168.0.40:5060 -> 1.2.3.4:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 20. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. CSeq: 14 INVITE. Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5. Content-Type: application/sdp. Content-Length: 481. . v=0. o=yate 1406132227 1406132227 IN IP4 192.168.0.40. s=SIP Call. c=IN IP4 192.168.0.40. t=0 0. m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. U 2014/07/23 19:17:08.129622 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: kamailio (4.0.6 (x86_64/linux)). Content-Length: 0. . U 2014/07/23 19:17:08.130107 1.2.3.4:5060 -> 2.3.4.5:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Record-Route: <sip:1.2.3.4;lr=on;ftag=838449717;nat=yes>. Max-Forwards: 16. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. CSeq: 14 INVITE. Content-Type: application/sdp. Content-Length: 499. . v=0. o=yate 1406132227 1406132227 IN IP4 1.2.3.4. s=SIP Call. c=IN IP4 1.2.3.4. t=0 0. m=audio 21888 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. a=nortpproxy:yes. U 2014/07/23 19:17:08.130593 2.3.4.5:5060 -> 1.2.3.4:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0;received=1.2.3.4;rport=5060. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: Asterisk PBX 1.8.29.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c". Content-Length: 0. . U 2014/07/23 19:17:08.130770 1.2.3.4:5060 -> 2.3.4.5:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 16. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 ACK. Content-Length: 0. . U 2014/07/23 19:17:08.131361 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: Asterisk PBX 1.8.29.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c". Content-Length: 0. . U 2014/07/23 19:17:08.149847 192.168.0.40:5060 -> 1.2.3.4:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 ACK. Max-Forwards: 20. Contact: <sip:10891@192.168.0.40:5060>. Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5. User-Agent: YATE/5.3.0. Content-Length: 0. — And kamailio.cfg attached: proLogika Sent with Airmail

2 1

MESSAGE forking bug(?)
by Allen Zhang 24 Jul '14

24 Jul '14

Hi, I believe there is a bug on MESSAGE forking. Test scenario: User A has two records in the location table, with different instance ids: 111 and 222. One of user A's instance is killed, hence 222 is not reachable. User B send a MESSAGE to user A. Kamailio forked the MESSAGE. Kamailio got a '200 Message delivered' response from 111 and forwarded it to user B. The MESSAGE forked to 222 timed out. Kamailio send a '408 Time out' to user B. This violates RFC 3428 section 4. It states that the forking UAS should only send ONE final response to the sender UAC. Forking is done by t_relay(), and the registrar parameter 'append_branches' is set to default value 1. Did I do something wrong, or there is a bug? Regards, Allen

2 5

Late parallel forking
by Guillaume Bour 24 Jul '14

24 Jul '14

Hi, I would like to do some kind of "late parallel forking" (see the example below). Is there a way to do such thing with Kamailio ? Here is an example: . A call B . B device 1 is connected so INVITE is forwarded to it . B device 2 registers succesfully . as device 1 as not yet answered and INVITE transaction is not expired, I want INVITE message to be forwarded to device 2 (so both are ringing) Regards, Guillaume Bour

4 6

Outbound registration?
by James Cloos 23 Jul '14

23 Jul '14

I don't see anything relevant in the docs, nor greping the src, but in case I missed something: Does the core, or do any of the modules support originating registration requests to other proxies a/o endpoints? I'd like to move registration responsibility to kama, and have it add a header on incoming INVITEs and the like from anything with which it has REGISTERed indicating which outbound registration is relevant to said request. If I'm right that there isn't support for that, do any of the app modules expose enough sip capability easily to write such? -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

3 4

Configuration kamailio + siremis
by Tomasz Sidło 23 Jul '14

23 Jul '14

Hello all I'm looking for the person who configures my kamailio + siremis I'll pay for the full configuration. Kamailio is to work together with a asterisk if anyone willing please contact me mail tomasz(a)sidlo.pl

1 0

Kamailio RtpProxy MHomed
by Pascal Fautré 23 Jul '14

23 Jul '14

Hi, I tried to use Kamailio / RTPProxy in mhomed setup without any luck. I had no problem to configure it with only 1 interface, without mhomed, everything worked perfectly. The RTP streams where not established correctly even if I managed to have to proper IP in the SIP INVITE (C & O). Versions: version: kamailio 4.1.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, 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 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 04:23:19 Jun 13 2014 with gcc 4.7.2 RTPProxy -v: Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Here is my RTPProxy config (/etc/default/rtpproxy) : CONTROL_SOCK=udp:127.0.0.1:7722 EXTRA_OPTS=“-l PU.BL.IC.IP/PRI.VA.TE.IP -m 11000 -M 12000 -d DBUG:LOG_LOCAL3 Here are snippets of my kamailio.cfg: port=5060 mhomed=1 # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; xlog("NATMANAGE M=$rm OU=$ou RURI=$ru RD=$rd F=$fu T=$tu NH=$nh(d) IP=$si ID=$ci\n"); if(dst_ip == PUBLIC_IP) { if(is_ipv4($nh(d)) && is_in_subnet($nh(d), PRIVATE_NET)) { xlog("NATMANAGE coei\n"); rtpproxy_manage("coei", PRIVATE_IP); } else { xlog("NATMANAGE coee\n"); rtpproxy_manage("coee", PUBLIC_IP); } } else { if(is_ipv4($nh(d)) && is_in_subnet($nh(d), PRIVATE_NET)) { xlog("NATMANAGE coii\n"); rtpproxy_manage("coii", PRIVATE_IP); } else { xlog("NATMANAGE coie\n"); rtpproxy_manage("coie", PUBLIC_IP); } } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } Calls were correctly going to the desired rtpproxy_manage options. Now I’m not quite sure I’m using the correct ones. I had to specify the PUBLIC_IP or PRIVATE_IP in the rtpproxy_manage calls in order to have the correct IP address in the C and O headers of the SIP INVITE. Without that, the public IP would be sent as C and O params to phones on the private subnet. In fact not a single call direction would give correct RTP streams. Any idea where I missed the turn? Cheers

4 10

core:drop_requests meaning
by Alex Balashov 23 Jul '14

23 Jul '14

Hi, I was wondering what precisely these stats mean: core:drop_replies = 0 core:drop_requests = 3 Assuming one does not explicitly drop() requests anywhere in script, would a continuous monotonic increase in drop_requests be considered a problem? Or are there some normal circumstances and ordinary behaviours (i.e. auth challenges) that would register a request drop? Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 3

server for git repository is down
by Daniel-Constantin Mierla 22 Jul '14

22 Jul '14

Hello, a short note to say that the server for git repository on sip-router.org is down. The hosting provider hasn't reported any maintenance, I am digging further into it to see what happened. Meanwhile, the mirror on github can be used if anyone wants to fetch the source tree: - https://github.com/kamailio/kamailio Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

How install Kamailio with Asterisk
by ANTHONY HESNAUX 22 Jul '14

22 Jul '14

hello all, I would install kamailio with ASTERISK in realtime on CENTOS 6.5 x64. I followed the tutorial aspito: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but its not work. I have a problem with kamailio.cfg Its not possible to start kamailio When I loadmodule "uac.so" I want to make this configuration: SIP phone1 -> Register Kamailio -> ASTERISK Register SIP phone 2 -> Register Kamailio -> ASTERISK Register kamailio: 192.168.0.100 Port: 5060 ASTERISK: 192.168.0.100 Port 5080

3 6

Unknown caller gets online user's identity
by g.aloitus＠gmail.com 21 Jul '14

21 Jul '14

Hello, I'm using Kamailio version 4.1.4+precise (amd64). I have followed "Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database" (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). One main difference in my setup compared to that one is that I continued use of Kamailio's database. The problem is as follows: I decided to put Kamailio and through it Asterisk reachable from internet. I have tried to configure Asterisk so that only calls of registered users would be possible, and they could only call to other registered users or conference rooms and echo test number. Then I took the following steps: I ensured that there was no online users with kamctl online. Then I launched MicroSIP (www.microsip.org), but I did not defined account, I simply set the protocol to tls and media encryption to mandatory, because I'm using these. I called to extension with xxx(a)my.public.ip.address (where xxx is extension) getting "unauthorized". And that was what I wanted. But if there is online users, calls go through, and incoming call is coming from Asterisk (in syslog I can find out that src_user=asterisk). Kamailio and Asterisk are listening the same IP address, but different port. I have refused connections to the Asterisk's port with iptables. I have defined my public IP address as domain in sip.conf. There is also other domain defined which corresponds to users' domain I am using in Kamailio's database. In kamailio.cfg there is if statement which prevents Kamailio not to be open relay: if (from_uri!=myself && uri!=myself) ... If I change this for example: if (from_uri!=myself || uri!=myself) I get what I want this time: no calls from outside, but I somewhat think that this is not a final solution. I have not found from log files such information which would have helped me. I have not yet investigated this problem so much that I could tell the logic behind the selection of online user's identity which is used. However, if I make a call to conference room I notice that Asterisk is thinking that one of online users has joined the conference. If I can recall correctly, I started with Kamailio version 3.2, and integrated it with Asterisk 11 (currently 11.10.2). Is there something which has changed in Kamailio, but what I have not changed in my setup which could explain this. Best, Teijo

6 18

kamcmd htable access problem, probably with int->string conversion
by Moritz Graf 21 Jul '14

21 Jul '14

Hi There, i implemented a mapping via htable. For debugging purpose I use kamcmd to see if the values are correctly filled. Doing so I discovered that there is a problem getting the values of the keys, where the name is a integer value. So it's the following behaviour: ##################################################################### root@kamailio-stage:~# kamcmd htable.dump maptodest { entry: 55 size: 1 slot: { item: { name: 1 value: 0000000000000000001 } } } { entry: 504 size: 1 slot: { item: { name: blub value: 0000000000000000001 } } } { entry: 686 size: 1 slot: { item: { name: 001 value: 0000000000000000001 } } } root@kamailio-stage:~# kamcmd htable.get maptodest blub { item: { name: blub value: 0000000000000000001 } } root@kamailio-stage:~# kamcmd htable.get maptodest 001 error: 500 - Not enough parameters (htable name and key name) ##################################################################### Is there a reasonable explanation why it works with blub, but not with 001? Why is he complaining about the number of parameters? Does anyone know a workarround? Thanks for any advise!! greeetz Moritz -- Moritz Graf, M.Sc. Betrieb NGN-Plattform G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg

2 3

pkg.stats
by Vitaliy Aleksandrov 21 Jul '14

21 Jul '14

Hi list, I have a problem with pkg.stats for tcp.main process on kamailio "4.1.2 73ea61-dirty". kamailio has 64M of private memory for each process (-M 64). pkg.stats for tcp.main process shows that is uses 75M and still has 62M free: { entry: 36 pid: 10032 rank: -4 used: 73109720 free: 65891192 real_used: 79146048 } root@webRTC:~# ps -p 10032 -o pid,tid,state,pcpu,pmem,rss,size,sz,vsz,args PID TID S %CPU RSS SIZE SZ VSZ COMMAND 10032 10032 S 0 26420 67508 76821 307284 /.../kamailio -f /.../kamailio.cfg -P /.../kamailio.pid -m 128 -M 64 -u kamailio -g kamailio This server handles 1000+ tcp/tls connections and works just fine for the last two months. That's why i think this is not a memory leak and just a pkg.stats problem. Does anybody else saw this or maybe somebody is already working to fix it ?

2 3

Msilo not delivering message (status 408)
by Peter Villeneuve 21 Jul '14

21 Jul '14

Hi, This is most likely a stupid mistake on my route config, but msilo correctly stores messages for offline users, but when they come back online, the message never arrives. Looking through the debug logs indeed there is an error, with status 408, which suggests kamaiio can't find a route to the user that just logged. Before I spend hours looking through all the routing blocks, perhaps someone more exprienced in kamailio can see the problem right away and offer a solution? Thanks DEBUG: msilo [msilo.c:1148]: m_dump(): msg [1-12] for: sip:1010@my.domain.com DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8e3040, 9, -1, 0x10) fd_no=2 called DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7f61ced1b1b0, state 1, fd=9, id=1 DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7f61ced1c4f8 DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7f61ced1b1b0, 1 from 0 DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x89ec40, 26, 2, 0x7f61ced1b1b0), fd_no=21 DEBUG: <core> [tcp_main.c:3515]: handle_tcp_child(): handle_tcp_child: CONN_RELEASE 0x7f61ced1b1b0 refcnt= 1 DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f61ced1d868, callback type 1024, id 0 entered DEBUG: msilo [msilo.c:1317]: m_tm_callback(): *completed with status 408* [mid: 140058058335472/13] DEBUG: msilo [msilo.c:1325]: m_tm_callback(): message <13> was not sent successfully DEBUG: msilo [ms_msg_list.c:236]: msg_list_set_flag(): mid:13 fl:8 DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f61ced5f098, callback type 1024, id 0 entered DEBUG: msilo [msilo.c:1317]: m_tm_callback(): completed with status 408 [mid: 140058058329184/12] DEBUG: msilo [msilo.c:1325]: m_tm_callback(): message <12> was not sent successfully DEBUG: msilo [ms_msg_list.c:236]: msg_list_set_flag(): mid:12 fl:8

4 19

please help with Msilo config
by Miguel Rios 19 Jul '14

19 Jul '14

Hi list, I'm a newbie when it comes to kamailio, although I have a fair amount of SIP experience. I'm trying to setup a very basic kamailio install (4.1 on Wheezy) with Msilo support. I just used the default kamailio.cfg file (changing obviously the relevant parameters for my setup) and have not touched the routing blocks. I'm very confused about how routing works, and I don't have a background in C. I've read the wiki up and down and have looked at inumerous tutorials spread out on the internet (most of which just add to my confusion because they seem out of date), but I still don't get it. I managed to setup a working kamailio server where local users can call each other fine. Now when I try to add the msilo module, and specially the routing example from MSILO Module is when things get tricky. I'm sure my error is a fairly basic one and has to do with the routing logic and syntax. Could some kind soul please share a copy of the whole ####### Routing Logic ######## on downwards for a basic no frills kamailio with msilo setup? Thanks, Miguel MSILO Module 3. Parameters 3.1. db_url (string) Database URL. Default value is “mysql://kamailio:kamailiorw@localhost/kamailio”. Example 1.1. Set the “db_url” parameter View on kamailio.org Preview by Yahoo

3 3

Fwd: Msilo not delivering message (status 408)
by Peter Villeneuve 18 Jul '14

18 Jul '14

Thanks Daniel. UDP is the same story as TCP and TLS, unfortunately. Looking through the logs, I see strange errors in parsing the contact header when msilo prepares to send the notification back to 1000. This is for the notification message that should have been sent back to 1000 after he sent a message to 1010 (offline). I also see that kamailio reports that 1000's location was not found, even though he's obviously registered. The more I play around with this, the more confused I become. Do you think this contact parsing error could be the root of the problems? Cheers, Peter Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <MESSAGE> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:1000@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK90dc.68116a97000000000000000000000000.0>; state=16 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg... Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [receive.c:193]: receive_msg(): preparing to run routing scripts... Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=10 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [28]; uri=[sip:1000@my.domain.com] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [< sip:1000@my.domain.com>#015#012] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <10> <MESSAGE> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=25 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-bb2e Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=29 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: siputils [sipops.c:161]: w_is_first_hop(): no 2nd via found - first hop Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: *ERROR: <core> [parser/contact/contact.c:175]: skip_name(): skip_name(): Error in contact, scheme separator not found* Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/contact.c:202]: parse_contacts(): parse_contacts(): Error while skipping name part Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/parse_contact.c:59]: contact_parser(): contact_parser(): Error while parsing contacts Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/parse_contact.c:88]: parse_contact(): parse_contact(): Error while parsing Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: nathelper [nhelpr_funcs.c:300]: get_contact_uri(): failed to parse Contact body Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: siputils [checks.c:103]: has_totag(): no totag Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: NOTICE: <script>: !------------------------------------>WE HAVE A NEW MESSAGE Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: registrar [lookup.c:181]: *lookup(): contact for [1000] found by address* Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==9 && [79.my.server.IP1] == [127.0.0.1] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==14 && [79.my.server.IP1] == [79.my.server.IP1] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: NOTICE: <script>: ------------------MESSAGE PROCESSING: *location not found*, storing offline message ID: (sip:1000@85.xx.xx.xx:59287;transport=TLS) Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:605]: m_store(): ------------ start ------------ Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:648]: m_store(): using user id [sip:1000@85.xx.xx.xx:59287;transport=TLS] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:732]: m_store(): 'From' header: <sip:registrar@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:849]: m_store(): message stored. T:<sip:1000@my.domain.com> F:< sip:registrar@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:858]: m_store(): sending info message. Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:899]: m_store(): contact header found On Thu, Jul 17, 2014 at 11:01 PM, Daniel-Constantin Mierla < miconda(a)gmail.com> wrote: > You should not call m_store() if src_ip==myself -- in this way you avoid > to store notification messages that cannot be delivered. > > Why notifications are not delivered has to be troubleshooted there. Have > you tried also with UDP? > > Cheers, > Daniel > >

1 0

[OT] LVUG, the London VoIP User Group
by Richard Brady 18 Jul '14

18 Jul '14

Hi folks LVUG is a monthly get-together in London to discuss open-source VoIP technology, including Kamailio. If you're based in or near London please join us this coming Tuesday for the third LVUG which will be held in Old Street. The format is a couple short talks followed by beers at the venue or in a nearby pub. Full details here: www.meetup.com/London-VoIP-User-Group-LVUG/ Hope to see you there! Richard -- Richard Brady E: rnbrady(a)gmail.com

1 0

Kamailio presence (NAT, XCAP)
by Radu Vatav 18 Jul '14

18 Jul '14

Hello, I'm just trying to use SIP as a replacement for Skype, nothing more than that. All is working except of presence. Using Jitsi 2.4.4997 (Windows x64) from home behind NAT and kamailio 4.1.4+wheezy on a rented server somewhere. My biggest problem is that I don't know exactly how it is supposed to work. Ok, this is what I understood so far: the clients connect via SIP to the server and can call each other, but for some reason the server can't notify them via SIP of presence changes when they are behind NAT. So the clients also connect to the built in xcap-server, register themselves as presence "watchers" and the xcap-server notifies them via udp/tcp of presence changes of the other users which are watched. All is very nice but it's not working and I don't know where to look for the problem. Which are exactly the steps in communication and what data exactly is there sent which way (when working correctly, which it doesn't) - no clue. I somehow put the configuration together from the 2-3 examples I found on the internet, I don't know if it is ok to attach something to this email, so here it is: https://vatav.cc/kam/kam_cfg.7z Here is the network activity on the server: https://vatav.cc/kam/kam_ngrep.7z So me ("radu") and "alice" are home behind the NAT, one is online and wants to see how the other one goes online and offline. I don't know what exactly to look for, but it doesn't look to me like there is anything sent to one user when the other one goes on-/off-line (there is a lot of "proxy authentication required" which drives me nuts, again I don't know if this is correct or not; Jitsi seems to be connected to the XCAP server, if I type in a wrong address for "XCAP server" Jitsi complains it can't connect). Some other things: - "kamctl online" shows almost real-time who is logged-in, so this works - siremis shows a bunch of users in "watchers list", but none in "active watchers list" - actually when a user goes online it contains the contacts of that user, but only briefly, when I refresh the view, the list is empty again. I don't know if this is how it is supposed to work, anyway, Jitsi doesn't show any change of the status of the user who changed status at all. Any help would be appreciated. It's sad something this simple as I need (basic Skype replacement) is so difficult to get working. Maybe this is why there is relatively low interest in SIP and everybody stays with MS/Google/... because "there isn't something else". Best regards, Radu Vatav

2 1

db_postgres Error
by Rene Montilva 18 Jul '14

18 Jul '14

Hi list i'm config SQLOps Module with postgresql db module or is postgresql driver? this is my kamailio config: #!ifndef DBURL #!define DBURL "postgres://user:ouser@x.x.x.x/kamailio" #!endif sql_query("ca", "select * from xxxxx","ra"); if($dbr(ra=>rows)> 0){ xlog("L_INFO", "saldo:$dbr(ra=>[0,0]) y seg:$dbr(ra=>[0,1]) "); $var(credit) = $dbr(ra=>[0,0]); # 30$ of credit $var(cost_per_sec) = $dbr(ra=>[0,1]); # 1$ per sec $var(i_pulse) = "1"; # 1$ to establish the call $var(f_pulse) = "1"; # 1$ per second sql_result_free ("ra"); if($var(cost_per_sec) > $var(credit)) { sl_send_reply("402","payment required"); exit; } } i got this warning and would like why this happen: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [SSL SYSCALL error: Connection timed out#012] WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [SSL SYSCALL error: EOF detected#012]

2 1

how to m_store to database not just a "body" with MSILO?
by denis kremov 18 Jul '14

18 Jul '14

Hello All, Subj, need you help, How to store this type of message to MSILO database? Content-Type: message/external-body;access-type=URL;URL="https://xxx.com//download/539c28884_3e071c9db68044f687bf.jpg" I modified and recompiled msilo.c to resolve this error: ERROR: msilo [msilo.c:626]: m_store(): body of the message is empty! but after that - m_store just store this information to database: mysql> select * from silo; +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ | id | src_addr | dst_addr | username | domain | inc_time | exp_time | snd_time | ctype | body | extra_hdrs | callid | status | +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ | 30 | sip:777351300@xxx.in | sip:777581288@xxx.in | 777581288 | xxx.in | 1402479947 | 1403479946 | 0 | message/external-body | | | | 0 | +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ 1 row in set (0.00 sec) How can I store to database "access-type=URL" and and URL="xxxx" ? Thanks All! -- denis kremov

3 2

Groups in Kamailio vs Freeswitch
by Peter Villeneuve 18 Jul '14

18 Jul '14

Hi guys, I'm designing a new service for a client and I was wondering what your opinion is of the 2 options I'm considering to separate users on the server. Basically I want to create closed groups where each user can only call and receive calls from members of the same group (single domain install). Initially I was thinking of just forwarding all the invites to FS and using the dialplan to enforce this segregation, but now I realize there is a groups module in kamailio too. Which 2 courses of action do people recommend? I'm more familiar with FS than Kamailio, so it would be easier for me to just forward everything to FS, although that will be more expensive and probably unnecessary if I could do it all in Kamailio with Siremis. Would kamailio's group module do the trick? Are there any tutorials out there I can follow? Cheers, Peter

2 1

trying to conifgure telepresence client software
by Fred Hunter 18 Jul '14

18 Jul '14

Hi, I recently registered with IP tel for the purposes of carrying out SiP based teleconferencing. I am attempting to register my software client, Polycom m100 desktop but am having some difficulty. Can you tell me what I should be entering into the 'registrar server' field, 'domain' and 'registrar server'? [cid:image001.png@01CFA03E.F01888F0] ---------------------------------------------------------------------------- www.puntersouthallgroup.com Punter Southall Group is a trading name of Punter Southall Group Limited. Registered office: 11 Strand, London, WC2N 5HR. Registered in England and Wales No. 04096788. This message and any attachments are confidential. If you have received this message in error please notify the sender and delete this email immediately. Unauthorised dissemination, disclosure, copying or use is prohibited. This email has been scanned for viruses and other malicious items. We reserve the right to monitor the content of all emails sent and received by our systems. Please consider the environment before printing this email. ----------------------------------------------------------------------------

2 1

Re: [SR-Users] how to remove non-digits using dp_translate
by Daniel-Constantin Mierla 18 Jul '14

18 Jul '14

Hmm, I see there is none inside dialplan for the moment. The option would be to do it in config file with a function from textops. Also, check if regex module is a good alternative for you: - http://kamailio.org/docs/modules/stable/modules/regex.html Cheers, Daniel On 16/07/14 16:50, Henry Fernandes wrote: > Can anyone show me how to use the regexp global flag using dp_translate? > -H > > From: Henry Fernandes <henry(a)usinternet.com <mailto:henry@usinternet.com>> > Reply-To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>> > Date: Friday, 11 July, 2014 4:18 PM > To: <miconda(a)gmail.com <mailto:miconda@gmail.com>>, "Kamailio (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>> > Subject: Re: [SR-Users] how to remove non-digits using dp_translate > > I don't know how to apply the global flag in Kamailio. Can you give > me an example? > -H > > From: Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > Reply-To: <miconda(a)gmail.com <mailto:miconda@gmail.com>>, "Kamailio > (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>> > Date: Thursday, 3 July, 2014 1:47 AM > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>> > Subject: Re: [SR-Users] how to remove non-digits using dp_translate > > Hello, > > you have to replace [^0-9] with empty, using the global flag. > > Cheers, > Daniel > > On 02/07/14 23:11, Henry Fernandes wrote: >> I'd like to use the dialplan module to remove all non-digits from a >> string. For example, "1-306-555-1234" becomes "13065551234" and >> "(306) 555-1234" becomes "3065551234". >> >> Unfortunately, I can't figure out the regular expressions for >> subst_exp and repl_exp. I can easily replace one instance of a >> non-digit, however I can't figure out how to replace all non-digits. >> >> -H >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -http://www.asipto.comhttp://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda > _______________________________________________ SIP Express Router > (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > _______________________________________________ SIP Express Router > (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Some of the calls drops after 15 minutes + some seconds
by Andras FOGARASI 18 Jul '14

18 Jul '14

Hi, I have a simple kamailio install (2 servers, using location service and a failover node with dispatcher, STUN, clients behind different NATs), without rtpproxy, only peer-to-peer RTP and TURN server if the connection is really messy (it's not relevant here). Signaling is over TLS. Both of the clients are behind NAT. Basically everything works, but some of the calls are dropped after 15 minutes and some seconds, for me it seems the RTP connection is dropped but not sure. I cannot find find out wether it's a kamailio problem or client/NAT problem. Has anyone idea what is going on? Thanks, Andras

3 8

Really need help on kamailio-snmpstats module
by Allen Zhang 17 Jul '14

17 Jul '14

Hi This is a re-post. I'm desperate. What I'm trying to do: Use rpmbuild to build kamailio rpms from source code. We made a lot of changes in the kamailio code. I can't use the rpms from the build service. The plan is to push these changes to kamailio but that's not done, yet. The problem is when I try to install kamailio-snmpstats module from the rpm built, it reports error: Error: Package: kamailio-snmpstats-4.0.0-0.el6.x86_64 (company_name) Requires: net-snmp-agent-libs I've installed net-snmp on the build machine and the machine that needs to install kamailio-snmpstats. I have tried to use the kamailio.spec file from kamailio build server but it didn't solve the problem. Can anyone help, please? Regards, Allen

2 6

Condition Statements - How many levels
by Joel White 16 Jul '14

16 Jul '14

I was wondering how many levels deep in conditional _if_ _else_ statements can go. I need to create a statement that has 4 possible outcomes and need to know if this is possible. I have two variables I want one thing to happen if only the first variable is present Another if only the second variable is present Yet another if they are both present And a fail over if None of the variables are present I do see that is can be done with 3 conditions using _if_ _if else_ _else_ how can I accomplish 4 conditional statements? Thank you in advance Joel

2 2

Re: [SR-Users] how to remove non-digits using dp_translate
by Henry Fernandes 16 Jul '14

16 Jul '14

Can anyone show me how to use the regexp global flag using dp_translate? -H From: Henry Fernandes <henry(a)usinternet.com> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Date: Friday, 11 July, 2014 4:18 PM To: <miconda(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] how to remove non-digits using dp_translate I don't know how to apply the global flag in Kamailio. Can you give me an example? -H From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: <miconda(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Date: Thursday, 3 July, 2014 1:47 AM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] how to remove non-digits using dp_translate Hello, you have to replace [^0-9] with empty, using the global flag. Cheers, Daniel On 02/07/14 23:11, Henry Fernandes wrote: > > I'd like to use the dialplan module to remove all non-digits from a string. > For example, "1-306-555-1234" becomes "13065551234" and "(306) 555-1234" > becomes "3065551234". > > > > > Unfortunately, I can't figure out the regular expressions for subst_exp and > repl_exp. I can easily replace one instance of a non-digit, however I can't > figure out how to replace all non-digits. > > > > > -H > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listi > nfo/sr-users > -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/lis tinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

rtpengine upgrade fails
by Juha Heinanen 16 Jul '14

16 Jul '14

i build debian packages for rtpengine branch 3.3.0.0+0~mr3.4.1.0. after that i did 'apt-get dist-upgrade' and it failed like this: Setting up ngcp-rtpengine-daemon (3.3.0.0+0~mr3.4.1.0) ... Installing new version of config file /etc/init.d/ngcp-rtpengine-daemon ... Restarting RTP/media proxy: ngcp-rtpengine-daemonFATAL: Module xt_MEDIAPROXY not found. iptables: No chain/target/match by that name. ip6tables: No chain/target/match by that name. FAILED TO CREATE KERNEL TABLE 0, KERNEL FORWARDING DISABLED invoke-rc.d: initscript ngcp-rtpengine-daemon, action "restart" failed. dpkg: error processing ngcp-rtpengine-daemon (--configure): subprocess installed post-installation script returned error exit status 255 however, if i now run 'apt-get install ngcp-rtpengine-daemon', setting up ngcp-rtpengine-daemon succeeds and also packages that depended on it get installed fine: Reading package lists... Done Building dependency tree Reading state information... Done ngcp-rtpengine-daemon is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. 2 not fully installed or removed. After this operation, 0 B of additional disk space will be used. Do you want to continue [Y/n]? Setting up ngcp-rtpengine-daemon (3.3.0.0+0~mr3.4.1.0) ... Restarting RTP/media proxy: ngcp-rtpengine-daemon. Starting RTP/media proxy: ngcp-rtpengine-daemon already running. Setting up ngcp-rtpengine (3.3.0.0+0~mr3.4.1.0) ... any ideas why setting up ngcp-rtpengine-daemon fails with dist-upgrade? -- juha

2 1

KAMILIO-4.1.4 REGISTRATION ISSUE
by Cibin Paul 16 Jul '14

16 Jul '14

I have kamailio 4.1.4 installed with DB Engine as mysql. I added users using the command kamctl add and I could see the entry in subscriber table. When I am trying to register, kamailio is accepting any combination of username and password not listed in the table. What could be wrong? I followed the INSTALLATION document provided in the source file. Please help me troubleshoot this. Kamailio is up and running and not throwing any errors. kamctl moni shows 1 user registered. But the registered user doesn’t even exist Server Details OS : Centos 5.5 Kamailio version - 4.1.4 MySQL version - 5.0.95 Cibin

2 4

Rtpproxy redundancy
by Dren Daka 16 Jul '14

16 Jul '14

Hi, I'm using kamailio with two network interfaces also running rtpproxy on the same interfaces so rtpproxy is load balancing, when one of the connections is down rtprpoxy still sends 50% of the calls in that interface. How can i make rtprpoxy to do failover from the non working connection to the working one instead of load balancing. Thanks.

1 0

BUG: ip_addr2sbuf: unknown address family 0
by AliReza Khoshgoftar Monfared 16 Jul '14

16 Jul '14

Hi, I am running a simple command on timer in a kamailio proxy. Here is the related parts of the config scripts: loadmodule "timer" > modparam("timer", "declare_timer", "tmr1=LOG_STATS,1,fast,enable"); > > route{ > ... > } > > route[LOG_STATS] { > exec_avp("netstat -ul | grep ':sip' | awk '{print > $$2}'","$avp(s:test)"); > xlog("L_ALERT","logged RecvQ as $avp(s:test)"); > return; > } > Each time I run the "netstat" using the exec() inside the timer route block, I get the following error from Kamailio in my log: kamailio[28546]: : exec [../../parser/../ip_addr.h:669]: ip_addr2sbuf(): > BUG: ip_addr2sbuf: unknown address family 0 > What is the meaning of this error, and what does kamailio mean by reporting it as a "BUG"? Is there any obvious reason, I am getting this? Is it related to calling netstat to survey the RecvQ? Thanks, Alireza

2 1

have problem with iptel account
by Samhu 16 Jul '14

16 Jul '14

hi, This is Sam from Draytek Crop. We have a IPPBX device with iptel for SIP account as below. User B dial to IPPBX could heart the answer from Auto Attendant of IPPBX. Then user B cannot heart anything (after input the extension of user A) when user A pickup his telephone . At this momemt, user A could hear everything from user B. This Network Architecture worked fine before. And it still works fine now if we select the account of "draytel.org". Please show us how to solve the problem. Best Regards, Sam Hu

2 1

SCA and PRESENCE
by Joel White 15 Jul '14

15 Jul '14

I am having an issue where I enable presence modules and the SCA module stops working. I disable presence and SCA works fine on Polycom phones. Does anyone know what I am missing or doing wrong? Thank you in advance Joel

3 3

function to reset all flags or bflags?
by Juha Heinanen 15 Jul '14

15 Jul '14

there is a function to reset a flag or bflag, but i haven't found one that would reset all flags or bflags. is there a way to do it? if not, is it ok to add such function? -- juha

2 2

Many instances of "new URI shorter than old URI" in log
by Alex Villacís Lasso 15 Jul '14

15 Jul '14

Jun 27 15:28:42 elx /usr/sbin/kamailio[25189]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:44 elx /usr/sbin/kamailio[25187]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:48 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:49 elx /usr/sbin/kamailio[25188]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI I am not really sure of what this message is supposed to indicate. It looks something related to uac_restore_[from,to] functions, but some debugging statements just before the use show that the message is not originating from there. This message is showing up hundreds of times per day.

2 4

set_rtp_proxy_set behaviour
by Igor Potjevlesch 15 Jul '14

15 Jul '14

Hello, I set two entries of rtpproxy like : modparam("rtpproxy", "rtpproxy_sock", "1 == udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_sock", "2 == udp:127.0.0.1:7723") For a particular case, I want to use the second RTPProxy and the first one as the default one. If in a conditional block I set set_rtp_proxy("2"), all other call to rtpproxy_manage will use the first RTPProxy? Or there is a load-balancing introduce by default. Regards, Igor.

2 3

Routing calls to Asterisk using dispatch
by Olli Heiskanen 15 Jul '14

15 Jul '14

Hello, I've started playing with an idea to add multiple asterisk servers and using dispatcher to balance the sip load between them. I added the code according to dispatcher module documentation ( http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but I think there's something off in my setup: kamctl ul output shows 2 AORs for one client: AOR:: 770(a)testers.com Contact:: sip:770@2.2.2.2:64340;rinstance=c634da314e12385f;transport=UDP Q= Expires:: 3221 Callid:: ZTE1MWYwYzM3NGNjNjMxMmEzM2JjYWNmNzQyZTdiNGI. Cseq:: 2 User-agent:: Z 3.2.21357 r21367 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 5087 Ruid:: uloc-53bfe447-35ae-2a2 Reg-Id:: 0 Last-Keepalive:: 1405174150 Last-Modified:: 1405174150 AOR:: 770(a)1.1.1.1 Contact:: sip:770@1.1.1.1:5070 Q= Expires:: 68 Callid:: 327fcf07641f80006e962821112a61b5(a)testers.com Cseq:: 754 User-agent:: Asterisk PBX 11.10.2 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 4294967295 Ruid:: uloc-53bfe447-35af-a82 Reg-Id:: 0 Last-Keepalive:: 1405174477 Last-Modified:: 1405174477 I don't think I should be seeing an AOR for 770 where Contact is the public address of my server (here 1.1.1.1) and User-Agent which is Asterisk. I'm using Asterisk Realtime integration, and by what I can tell the sip messages are going nicely, client authenticates with Kamailio and sends this message to Asterisk (which is on the same machine; Kamailio at 5060 and Asterisk at 5070): 1.1.1.1.sip > 1.1.1.1.vtsas: SIP, length: 374 REGISTER sip:1.1.1.1:5070 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKbc8a.f4473947000000000000000000000000.0 To: <sip:770@1.1.1.1> From: <sip:770@1.1.1.1>;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-d0c1 CSeq: 10 REGISTER Call-ID: 7ffa0191-13742(a)1.1.1.1 Max-Forwards: 70 Content-Length: 0 Contact: <sip:770@1.1.1.1:5060> Expires: 3600 Currently I can make calls from 770 to 123 which is an Asterisk extension that answers, plays hello world and hangs up. However I can't call another sip clients when I route calls through Asterisk, they do work fine if I don't use Asterisk for handling calls, but I'd like Kamailio to be in the role of proxy/loadbalancer and Asterisk to handle calls. My config is the simple default config, added with realtime stuff and then dispatcher according to the documentation. I wonder if there's something wrong in the REGISTER that Kamailio sends to Asterisk, or maybe something else going wrong? Has anyone seen results like this and do you spot something here that needs fixing? Thanks, Olli

1 1

siremis/ser/xrcmds error help
by MARY 15 Jul '14

15 Jul '14

sr-users，hello：　 http://127.0.0.1:8890/siremis/ser/xrcmds error info: [2014-07-14 08:49:30 (GMT)] An exception occurred while executing this script: Error message: #2, require_once(XML/RPC.php): failed to open stream: No such file or directory Script name and line number of error: /xxxx/siremis-4.1.0/siremis/modules/ser/service/asipto/libs/cmds/serxr.php:2 why? can not find the XML/RPC.php file。 2014-07-1417:56:16 MARY

2 1

Re: [SR-Users] Forward UAC Registration with Authentication (like Upper Registration or Passthrough Registration)
by Angelica Juarez 14 Jul '14

14 Jul '14

Anjelicajuarez desde mi iPhone

1 0

To Rewrite SDP when request goes from kamailio to outside network
by Anurag Mishra 14 Jul '14

14 Jul '14

Hi All, I have been struggling with this issue for a long and could not find a way to solve it. I have setup FreeSWITCH and Kamailio on the single server but on a private server with a private IP and fronted by firewall. Firewall has all the ports opened. FreeSWITCH is running with nonat. Kamailio has public IP advertised on port 5060 and private IP on 5070 for phones on LAN. Problem is with outbound calls there is no audio and reason I can see is contact field in SDP has the private IP of the server instead of the public IP. Internal device to device calls is working fine. I have tried to rewrite the SDP and have written below code, but got no success: request_route > { > xlog("L_DBG", "Request Route Enter"); > if (is_method("INVITE")) > { > xlog("L_DBG", "Request Route Enter in Invite"); > if (has_body("application/sdp")) > { > xlog("L_DBG", "Request Route Enter in SDP of > Invite"); > if (search("IN IP4 PRIVATE_IP")) > { > xlog("L_DBG", "Request Route Search IN IP4 > PRIVATE_IP"); > fix_nated_sdp("8", "PUBLIC_IP"); > fix_nated_sdp("2", "PUBLIC_IP"); > } > } > } > } > Can anyone help me to get rid of this. Thanks, Anurag Mishra

2 1

Re: [SR-Users] how to remove non-digits using dp_translate
by Henry Fernandes 11 Jul '14

11 Jul '14

I don't know how to apply the global flag in Kamailio. Can you give me an example? -H From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: <miconda(a)gmail.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Date: Thursday, 3 July, 2014 1:47 AM To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] how to remove non-digits using dp_translate Hello, you have to replace [^0-9] with empty, using the global flag. Cheers, Daniel On 02/07/14 23:11, Henry Fernandes wrote: > > I'd like to use the dialplan module to remove all non-digits from a string. > For example, "1-306-555-1234" becomes "13065551234" and "(306) 555-1234" > becomes "3065551234". > > > > > Unfortunately, I can't figure out the regular expressions for subst_exp and > repl_exp. I can easily replace one instance of a non-digit, however I can't > figure out how to replace all non-digits. > > > > > -H > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listi > nfo/sr-users > -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Comparing strings
by Alex Balashov 11 Jul '14

11 Jul '14

Is there any compelling reason to use textops' cmp_str()/cmp_istr() versus the 'eq' operator (or the implicit string comparison in "a" == "b")? -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 1

Would like to upgrade to 4.1.4+wheezy -- when will this be available?
by Will Ferrer 11 Jul '14

11 Jul '14

Hi We would like to upgrade or debian instances to the apt repo for 4.1.4+wheezy. Is this release soon to be stable and ready for use? We would like to upgrade so we can get around the memory leak that arises when many reloads occur as described here: http://sip-router.org/tracker/index.php?do=details&task_id=436 Thank you very much for any advice any one can offer us. All the best. Will Ferrer

2 3

Update Interval for Rtimer
by AliReza Khoshgoftar Monfared 11 Jul '14

11 Jul '14

Hi, I have a basic yet weird problem with Rtimer module. I cannot make it execute a route block for me at an interval of my choice (which is what it is supposed to do, right?) Here is a simplified example with relevant parts of my config script: Assume I want to print "hello" every 100 ms: I have a script like this (I've put all my global variables, in case they matter and my relevant route blocks...) Is there anything I am doing wrong here in the usage of rtimer? *####### Global Parameters #########* > > *#!ifdef WITH_DEBUG* > *debug=4* > *log_stderror=yes* > *#!else* > *debug=-2* > *log_stderror=no* > *#!endif* > > *memdbg=5* > *memlog=5* > > *log_facility=LOG_LOCAL0* > > *fork=yes* > *children=4* > > */* comment the next line to enable TCP */* > *disable_tcp=yes* > > */* uncomment the next line to disable the auto discovery of local aliases* > * based on revers DNS on IPs (default on) */* > *auto_aliases=no* > > */* add local domain aliases */* > *# alias="mysipserver.com <http://mysipserver.com>"* > *dns=no # (cmd. line: -r)* > *rev_dns=no # (cmd. line: -R)* > > *port=5060* > > */* uncomment and configure the following line if you want Kamailio to * > * bind on a specific interface/port/proto (default bind on all > available) */* > *listen=eth0:5060* > > *sip_warning=no* > > *####### Modules Section ########* > > *#set module path* > > *#mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/"* > *loadpath "/usr/lib/x86_64-linux-gnu/kamailio/modules"* > > > *#loadmodule "db_mysql.so"* > *loadmodule "mi_fifo.so"* > *loadmodule "kex.so"* > *loadmodule "tm.so"* > *loadmodule "tmx.so"* > *loadmodule "sl.so"* > *loadmodule "rr.so"* > *loadmodule "pv.so"* > *loadmodule "maxfwd.so"* > *loadmodule "textops.so"* > *loadmodule "siputils.so"* > *loadmodule "xlog.so"* > *loadmodule "xprint.so"* > *loadmodule "sanity.so"* > *loadmodule "ctl.so"* > *loadmodule "mi_rpc.so"* > *loadmodule "acc.so"* > *loadmodule "dispatcher.so"* > *loadmodule "exec.so"* > *loadmodule "rtimer.so"* > *loadmodule "benchmark.so"* > > > *# ----------------- setting module-specific parameters ---------------* > > > *# ----- mi_fifo params -----* > *modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")* > > > *# ----- rr params -----* > *# add value to ;lr param to cope with most of the UAs* > *modparam("rr", "enable_full_lr", 1)* > *# do not append from tag to the RR (no need for this script)* > *modparam("rr", "append_fromtag", 0)* > > > *# ----- acc params -----* > *modparam("acc", "log_flag", 1)* > *modparam("acc", "failed_transaction_flag", 3)* > *modparam("acc", "log_extra", * > * > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si")* > > *# ----- tm params -----* > *modparam("tm", "fr_timer", 2000)* > *modparam("tm", "fr_inv_timer", 40000)* > > *# ----- dispatcher params -----* > *modparam("dispatcher", "list_file", > "/home/ubuntu/Downloads/dispatcher.list")* > *modparam("dispatcher", "flags", 2)* > *modparam("dispatcher", "dst_avp", "$avp(AVP_DST)")* > *modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)")* > *modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)")* > > *# ----- xlog params -----* > *modparam("xlog", "buf_size", 8192)* > *modparam("xlog", "log_facility", "LOG_LOCAL0")* > > *# ----- rtimer params -----* > *modparam("rtimer", "timer", "name=ta;interval=1;mode=1;")* > *modparam("rtimer", "exec", "timer=ta;route=8")* > > > *####### Routing Logic ########* > > > *# main request routing logic* > *route {* > * ...* > * # calls various routing blocks, but no explicit call to route("8")* *}* *route[8] {* * # to be executed on timer * * xplog("L_ALERT","hello");* *}*

2 2

INVITE to 200, INVITE to 404, INVITE to 487
by Luis F Urrea 11 Jul '14

11 Jul '14

Hi all, I have seen Siremis charts that account stats such as INVITE to 200 and INVITE to 404. I would like to have this sort of accounting but without the use of the Siremis UI, so I am wondering how is this achieved, is this done through the ACC module and correlating responses? TIA

1 1

Path attribute not updated in location table when user switches between two access points behind NAT
by Prasun Bheri 11 Jul '14

11 Jul '14

Hi Group, I stumbled upon a possible bug in Kamailios (4.0.0) Path module. Setup: Proxy server is behind another Kamailio Load Balancer and both (Proxy & LB) are configured to use Path module. When client registers for the first time path is inserted into Location table successfully however when client switches from currently Access Point to another AP (which are both behind a NAT), Path attribute in Location table is not updated with newer <ip>:<port> When client switched from one AP to another, IP remained same however port changed from 48238 to 33038 and subsequent update query issued by Kamailio is as follows: update location set expires='2014-07-11 19:27:32',q=-1.00 ,cseq=1805331004,flags=0,cflags=64,user_agent='xxxxxxxxxx',received='sip:xx5.xx1.xx1.219: *33038*;transport=tcp',path='<sip:xx3.xx1.xx9.53;transport=tcp;lr;received=\'sip:xx5.xx1.xx1.219:33038;transport=tcp\'>',socket='tcp:xx3.xx1.xx9.28:5060',methods=16351,last_modified='2014-07-11 18:27:32',ruid='uloc-53bf7f9c-4254-e',instance='<urn:imei:911304251588897>',reg_id=1 where username='testuser' *AND **contact='sip:testuser@xx5.xx1.xx1.219:33038;transport=tcp' *AND callid='7983ce39-2ffd3a92-8f82f477-e6313128-84b64128' AND domain=' vvoip.xxxxxxxx.com' In the where clause contact should have been old value taken from initial registration, however contact from new registration is used, so where condition always fails and entry is never updated. as a result no further calls or messages could be received by the client. is there any workaround for this issue? PS: Here is the configuration of path module. # set path to lazy mode modparam("registrar", "path_mode", 1) # set to 1 to use path header modparam("registrar", "use_path", 1) # set to 1 to use path received parameter modparam("registrar", "path_use_received", 1) modparam("path", "use_received", 1) Thanks & Regards, Prasun

1 0

Problem with installing rtpengine
by Muhammad Shahzad 11 Jul '14

11 Jul '14

Hi, I am trying to upgrade from mediaproxy-ng to rtpengine trunk version. The compilation steps go well and i have deb packages created. However when i try to install them (on same machine where they compiled), i get this error for every deb package, ngcp-rtpengine-daemon pre-depends on ngcp-system-tools * ngcp-system-tools is not installed.* I can't find this ngcp-system-tools package anywhere, neither on source nor in built deb packages. So, i am stuck. Can anyone help? Thank you.

2 2

sip_trace
by Rafal Skrzeczkowski 10 Jul '14

10 Jul '14

Hi, I'm trying to make configuration that will trace registrations. I can trace REGISTER messages using sip_trace. But how can i trace responses generated by pv_www_authenticate using sip_trace? Rafal Skrzeczkowski

2 2

Listen to to 2 NIC's IP addresses and establish call between
by Raj Roy Ghandhi 10 Jul '14

10 Jul '14

Hi Friends, I have to configure a Kamailio in to meet following requirements 1. PC having 2 NICs (NIC-A and NIC-B) with 2 different IPs (IP-A and IP-B) from 2 different service providers. Totally 2 different IPs. NIC-A attached to provider is a private SIP traffic provider and having IP-A. (no internet access) NIC-B attached to internet service provider and having IP-B. 2. I get SIP traffic to IP-A of NIC-A from SIP traffic provider. 3. I need to forward that SIP traffic to another SIP carrier in remote location using IP-B of NIC-B through internet and have the call establish without any issue. How can I achieve this with single instance of Kamailio running on same machine. Please advice me. Best Regards, Roy.

2 2

crash after 100 trying on NOTIFY
by Kristian F. Høgh 10 Jul '14

10 Jul '14

Hi, We've had the following crash. Kamailio 4.0.4 Client sends SUBSCRIBE, kamailio answers 202. Kamailio sends NOTIFY, client answers 100 trying. fr_timer is 5000. The client doesn't send "200 OK", so kamailio generates 408. Any suggestions? Regards, Kristian Høgh (gdb) bt #0 0x00007f3ab0c91852 in cancel_branch (t=0x7f3a9c779190, branch=0, reason=<value optimized out>, flags=4) at t_cancel.c:284 #1 0x00007f3ab0cd9c9b in reply_received (p_msg=0x7f3ab26ac560) at t_reply.c:2198 #2 0x0000000000458924 in do_forward_reply (msg=0x7f3ab26ac560, mode=<value optimized out>) at forward.c:799 #3 0x00000000004a108e in receive_msg ( buf=0x8dbd80 "SIP/2.0 100 Trying\r\nVia: SIP/2.0/UDP 10.10.10.10;branch=z9hG4bK83f6.32f26ad2.0\r\nTo: sip:554@<domain>;tag=7ce8fb6e\r\nFrom: sip:144@<domain>;tag=93983715339d3cf39b"..., len=<value optimized out>, rcv_info=0x7fff70f07090) at receive.c:272 #4 0x0000000000533ed6 in udp_rcv_loop () at udp_server.c:557 #5 0x000000000046964a in main_loop () at main.c:1638 #6 0x000000000046c4e2 in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2566

1 0

Need help with Routing presence NOTIFY messages
by Veerabhara Gundu 09 Jul '14

09 Jul '14

Hello Experts, I have two Kamailio servers with built in presence servers with load balanced with DNS, both sharing same data base. Registration, authentication and messaging are working without any problem. When it comes to Presence, NOTIFY messages generated by Kamailio is getting rejected by firewall if the client didn't register to that particular Kamailio. In MESSAGES case, I read the socket info from Location table, if the client is not registered to that Kamailio forwarded to the server which serving that user. But in NOTIFY case, these messages are not coming to routing logic at all, Kamailio is directly sending messages by looking at the contact information from active-watchers table. One possible solution I am thinking is, separating presence logic and starting that as separate server. That way I can play with contact header. Any other thoughts, I appreciate any help with this. I have spent enough time on this. Thanks, Veera Thanks, Veera

2 6

Re: [SR-Users] What is q-value ??
by Daniel Grotti 09 Jul '14

09 Jul '14

Hi, q value is used in location table for serial forking. If you have a subscriber with 2 contact registered you can use the q value to load one contact first (the one with the higher q) and afterwards load and call the second one. Highest is the value stored, highest is the priority. Daniel On 9 Jul 2014 06:20, aawaise <owaisulhaq(a)hotmail.com> wrote: > > There is a mention of highest "q value" in documentation of registrar > module's function "lookup(domain [, uri])". > Can some one please explain what this "q value" is ?? > It is used in 4.2 section of following link. > http://kamailio.org/docs/modules/stable/modules/registrar.html#idp1976344 > > Any help will be appreciated. Thanks. > > > > -- > View this message in context: http://sip-router.1086192.n5.nabble.com/What-is-q-value-tp128895.html > Sent from the Users mailing list archive at Nabble.com. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

What is q-value ??
by aawaise 09 Jul '14

09 Jul '14

There is a mention of highest "q value" in documentation of registrar module's function "lookup(domain [, uri])". Can some one please explain what this "q value" is ?? It is used in 4.2 section of following link. http://kamailio.org/docs/modules/stable/modules/registrar.html#idp1976344 Any help will be appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/What-is-q-value-tp128895.html Sent from the Users mailing list archive at Nabble.com.

1 0

Using 'reg_fetch_contacts' in Kamailio-3.1.1
by aawaise 09 Jul '14

09 Jul '14

I want to extract domain of a user from Location List For this I found "reg_fetch_contacts(table, uri, profile)" function to be the most suitable function. I want to know when was it added to registrar module ? Can it be used in kamailio version 3.1.1 ? Or I should update my kamailio version ?? Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Using-reg-fetch-contacts-in-Kamaili… Sent from the Users mailing list archive at Nabble.com.

1 0

Access the attributes of a location record
by aawaise 09 Jul '14

09 Jul '14

I want to extract domain attribute of a user from Location Table. I have gone through the readme of registrar module, that has an example of using reg_fetch_contacts() and how to access the attributes of a location record ? But I get the callee domain = NULL when I log using command xlog("callee=>domain: $(ulc(callee=>domain)))\n". Can anyone please help me with the correct syntax. Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Access-the-attributes-of-a-location… Sent from the Users mailing list archive at Nabble.com.

1 0

Help me to configure kamailio on EC2
by Jayaraman, Kamalakannan 09 Jul '14

09 Jul '14

Hi, I had installed kamailio on my AWS EC2, by following the steps in http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x . Since I installed in EC2, i changed the listen=<my-private-ip> advertise=<my-public-ip> in kamailio.cfg file too. I open the 5060 port in AWS console too. Kamailio starts without any error, but I can't able to connect in Jitsi client. Please help me out to do this as soon as possible.

3 2

No sound between chrome and firefox
by Yuriy Gorlichenko 08 Jul '14

08 Jul '14

Hello. I have Kamailio 4.1.3+rtpengine_rtpproxy-ng as module for rtpengine. Kamailio installed as frontend (registration, auth, proxy ) of asterisksk servers. WEBRTC users registred at kamailio and asterisk works as media server. When I try to call from Jssip from Firefox to chrome to way audio is fine. When I call from chrome - I see rtp packets only from firefox. Not from chrome. At kamailio log when I call from chrome log the same as whe i try to call from firefox (I can not see anithing wrong) So somebody else have same issue and may be somebody know reasons of this. Thanks for advice

2 2

sip_trace and registration responses
by Rafal Skrzeczkowski 08 Jul '14

08 Jul '14

Hi, I'm trying to make configuration that will trace registrations. I can trace REGISTER messages using sip_trace. But how can i trace responses generated by pv_www_authenticate using sip_trace? Rafal Skrzeczkowski

1 0

Kamailio, CPL, SIP CGI and alternatives.
by LAA 08 Jul '14

08 Jul '14

Hi all, I'm looking for a mechanism to let users personalize the behaviour of their profile, uploading configuration changes to Kamailio. For example voice mail redirecion, time-based routing, etc. I'm testing CPL scripts (cpl-c module with Kamailio 4.1.3), but I have problems with the behaviour of some features. "reject" and "redirect" seem to be working fine, but "proxy" is not working. ¿Has anybody experienced the same problems? I could hardly find any post regarding CPL scripting with Kamailio. What are the most commonly used alternatives for this purpose with Kamailio? ¿SIP CGI? ¿any other? Best regards. Luis.

2 2

SDP_NAT on Retry
by Timo Klecker 08 Jul '14

08 Jul '14

Hello everybody, we have got a problem with sdp-nat on retries of a 200 OK packet. Due to slow response of our trunk the 200 ok is resend (up to 3 times) but sdp-nat is only done on the first two packets. We are running Kamailio 4.1.0 with the following setup: Trunk --> NAT --> Kamailio --> Voip Bridge The rtp stream from the trunk (public ip) is supposed to go to one of our media gateways (behind 1:1 nat). We use sdp_mangle_ip to replace the private IPs of our Mediagateway with the public ones. This goes well with the first 200 OK packet but not with a retry. A.A.A.A (public trunk media IP) B.B.B.B (our public media IP - 1:1 natted to I.I.I.I) INVITE Trunk --> Kamailio Connection Information (c): IN IP4 A.A.A.A INVITE Kamailio --> Voip Bridge Connection Information (c): IN IP4 A.A.A.A 200 OK Kamailio <-- Voip Bridge Connection Information (c): IN IP4 I.I.I.I 200 OK Trunk <-- Kamailio Connection Information (c): IN IP4 B.B.B.B 200 OK Kamailio <-- Voip Bridge Connection Information (c): IN IP4 I.I.I.I 200 OK Trunk <-- Kamailio Connection Information (c): IN IP4 I.I.I.I ACK Trunk --> Kamailio --> Voip Bridge Our configuration used is the following: route[RELAY] { xlog("L_INFO"," In route Block RELAY \n"); # enable additional event routes for forwarded requests if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (!t_relay()) { xlog("L_NOTICE", " (ROUTE_RELAY) Relay failed, sending error \n"); sl_reply_error(); } exit; } onreply_route[MANAGE_REPLY] { xlog("L_INFO"," In route Block MANAGE_REPLY \n"); if ( isflagset(20) ) { -- we have to do nat route(SDP_NAT); force_send_socket(KAMAILIO_TO_TRUNK_IP); } else { force_send_socket(KAMAILIO_TO_BRIDGE_IP); } } How can we handle the 200 OK retry? Is there a way to do this? Kind regards Timo Klecker

1 0

SIP domain translation between providers
by Jeremy Visser 08 Jul '14

08 Jul '14

G'day folks Struggling to work out whether Kamailio can help me or not in this situation, and I hope someone can tell me whether I'm barking up the wrong tree. I have hundreds of users which register to an old legacy Asterisk box that needs retiring. The "users" are customers consisting of VoIP ATAs, PBXes, softphones, and other hardware phones. I've been told I need to retire the Asterisk system in the next month and move customers to our new provider, who provide the SBC themselves (it's a Broadsoft-based system). Obviously a customer once ported to the new provider requires new SIP proxy and SIP domain settings configured in their CPE. For obvious reasons, I cannot require hundreds of customers to change their configuration overnight. However, there is not enough time to assist hundreds of customers changing their SIP proxy/registrar/domain details. So in the meantime I'm trying to work out a transition solution. I cannot simply change my SIP registrar DNS to be a CNAME for the new provider's registrar, as the domain is different, and the new provider doesn't allow domains other than their own to register with them (even if the username/password portion is correct). My new provider has previously told me they are unwilling to accept my old domain whilst authenticating (they are much bigger than me, and I have no power to change them). So I was hoping to use Kamailio and possibly the Path or Dispatch modules to proxy SIP registration to the new provider, rewriting the domain part of the SIP register string as I go But I can't for the life of me work out whether that's even possible. I have spent weeks poring over various Kamailio configurations posted to the interwebs, but nobody seems to have done this before. Specifically the "translation proxy" -- which I am hoping to make Kamailio -- will need to be stateless, and perform no local authentication. Individual registrations will need to be passed through to the new provider (including authentication!), while still translating the domain in the Request-URI, To, From, and various other SIP headers. Because the current legacy Asterisk box has nat=yes set for all users, I will need to proxy media, because if I start proxying SIP traffic to the new provider, I will be confusing their own NAT hacks. I don't understand the documentation for either the Path or Dispatch modules, and the information on the wiki is confusing and poorly written. I am very familiar with SIP, and have no trouble configuring Asterisk, but trying to work with Kamailio is leaving me feeling like an idiot. However, Kamailio's config is very flexible, so despite this I suspect what I'm trying to achieve may be possible. Any pointers in the right direction? Cheers, Jeremy.

3 2

Strange avp_check behavior
by Klaus Darilion 08 Jul '14

08 Jul '14

Hi! Kamailio 4.1.4. When using avp_check with fnmatch and AVP the result is always TRUE if it is true for 1 time. It only happens with $avp(), but not with $var(). Is this a bug or some special behavior of AVPs? xlog("L_ERR", "======= with $$avp(...)================"); $avp(rpid) = "+499539110"; $avp(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$avp(rpid)", "fm/$avp(pattern)/g")) { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): YES"); } else { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): NO"); } $avp(pattern) = "+499539*"; # MATCH if (avp_check("$avp(rpid)", "fm/$avp(pattern)/g")) { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): YES"); } else { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): NO"); } $avp(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$avp(rpid)", "fm/$avp(pattern)/g")) { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): YES"); } else { xlog("L_ERR", "$avp(rpid) matches $avp(pattern): NO"); } xlog("L_ERR", "======= with $$var(...)================"); $var(rpid) = "+499539110"; $var(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$var(rpid)", "fm/$var(pattern)/g")) { xlog("L_ERR", "$var(rpid) matches $var(pattern): YES"); } else { xlog("L_ERR", "$var(rpid) matches $var(pattern): NO"); } $var(pattern) = "+499539*"; # MATCH if (avp_check("$var(rpid)", "fm/$var(pattern)/g")) { xlog("L_ERR", "$var(rpid) matches $var(pattern): YES"); } else { xlog("L_ERR", "$var(rpid) matches $var(pattern): NO"); } $var(pattern) = "+499539[2-5]"; # NO MATCH if (avp_check("$var(rpid)", "fm/$var(pattern)/g")) { xlog("L_ERR", "$var(rpid) matches $var(pattern): YES"); } else { xlog("L_ERR", "$var(rpid) matches $var(pattern): NO"); } --> The result is: <script>: ======= with $avp(...)================ <script>: +499539110 matches +499539[2-5]: NO <script>: +499539110 matches +499539*: YES <script>: +499539110 matches +499539[2-5]: YES <script>: ======= with $var(...)================ <script>: +499539110 matches +499539[2-5]: NO <script>: +499539110 matches +499539*: YES <script>: +499539110 matches +499539[2-5]: NO Thanks Klaus

1 1

Where can I get kamctl.oracle
by Konstantin Khatskevich 08 Jul '14

08 Jul '14

Hello! I attempt to dispose the bond ODBC-Oracle, but I cannot find kamctl.oracle (I have any other)& Where can I download this kamctl.oracle? -- Best regards, Konstantin Khatskevich

2 1

Re-Compiling and Re-Installing kamailio server
by aawaise 08 Jul '14

08 Jul '14

I am trying to make changes in code and recompile kamailio server project. Problem is when I try the following command make PREFIX='/usr/local/kamailio-3.1/' FLAVOUR=kamailio.cfg I get the message; 'kamailio' is up to date. I want to know, how to make my machine forget about the older installation, so that new compilation goes good. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Re-Compiling-and-Re-Installing-kama… Sent from the Users mailing list archive at Nabble.com.

2 1

Help me to configure Kamailio on EC2, not able to connect through Jitsi
by Jayaraman, Kamalakannan 08 Jul '14

08 Jul '14

Hi, I had installed kamailio on my AWS EC2, by following the steps in http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x . Since I installed in EC2, i changed the listen=<my-private-ip> advertise=<my-public-ip> in kamailio.cfg file too. I open the 5060 port in AWS console too. Kamailio starts without any error, but I can't able to connect in Jitsi client. Please help me out to do this as soon as possible. -- *Kamalakannan Jayaraman* *Associate Software Engineer* Pearson English kamalkannan.jayaraman(a)pearson.com +91 8754 7754 53 - Mobile kamalakannan.jayaraman.ge - Skype 6th Foor, 10th Block, DLF IT park, Chennai - 600089 India Pearson English is the world's largest provider of English language learning.

1 0

need help on kamailio-snmpstats module
by Allen Zhang 07 Jul '14

07 Jul '14

Hi, I'm having trouble on installing kamailio-snmpstats module. I'm using kamailio 4.0. I want to install kamailio's snmpstats module from RPM. I compiled net-snmp 5.4 from source and built some RPMS from it. I've installed net-snmp 5.4 from RPM on the target machine. And when I run 'yum install kamailio-snmpstats', I get: Error: Package: kamailio-snmpstats-4.0.0-0.el6.x86_64 () Requires: net-snmp-agent-libs Where do I get this net-snmp-agent-libs? Can I buid this from net-snmp source somehow? Regards, Allen

2 2

Measuring Response Time for a SIP Message
by AliReza Khoshgoftar Monfared 07 Jul '14

07 Jul '14

Hi, I am trying to do two types of measurements on a packet (say an INVITE or BYE) that arrives at a kamailio proxy. 1) The time that packet is spending in Kamailio to be processed 2) The total time from when the packet arrives at the machine that runs the proxy till it leaves that machine. I believe kamailio shall be able to help me with (1) and for (2) , I may need to use some linux tools Have anybody tried to measure response time with either definition? and in case that you have, how have you done it? Thanks Alireza

2 1

$avp(received)
by Kelvin Chua 07 Jul '14

07 Jul '14

is there any way of getting the received ip/port/proto from nathelper? modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") set_contact_alias(); xlog("$avp(RECEIVED)"); this gives me <null> Kelvin Chua

2 3

kamailio server over internet
by Gaurav Chaudhary 07 Jul '14

07 Jul '14

sir , i have set-up a kamailio server on a machine , every thing is in default configuration. but the issue is only local clients are able to register. i have forwarded the port 5060 correctly but the clients from some other network over the internet are not able register every request just got time out at the client side. do i nee to enable some specific setting to make it happen.

2 1

Extract value from a function
by aawaise 07 Jul '14

07 Jul '14

I came across this function of registrar module reg_fetch_contacts (domain, uri, profile) How can I log domain of a user using this function ?? Need help in syntax. Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Extract-value-from-a-function-tp128… Sent from the Users mailing list archive at Nabble.com.

2 1

Re: [SR-Users] UAC remote registration - refreshing database
by Dan Christian Bogos 07 Jul '14

07 Jul '14

Hey Daniel, I totally agree on improving versus adding new module. The only thing which made me going towards porting was not knowing how easy would be enhancing vs porting something which is implemented. Also the fact that Alexa has mentioned it as an often requested feature made me think that it could take a while until someone would be able to do it in the existing module. Anyway, thanks for your input! DanB PS: Ovidiu, I appreciate your feedback as well. Good to know that you are around ;).

2 1

MSILO - dumped messages get duplicated
by Marc M. 06 Jul '14

06 Jul '14

Hi, I have a Kamailio 3.2 setup up and running. Recently I configured msilo for offline message store. Essentialy the setup runs fine. Messages to users without location info getting stored. In case the user has location info, but cannot be reached, I made failover_route, so the messages getting stored in this case too. Messages getting dumped at the next REGISTER. However, I have a group of new mobile clients, with unstable connection. It happens many times that they send a REGISTER, but the connection brakes afterwards. It this case the message is getting dumped, but since it cannot be delivered, it runs on the failover_route which stores the message again. However, the previous message is not removed from the db, so the same message is now twice in the db. Next time the message either gets delivered in double, or in case of a new failure, it gets stored 2x2= four times. I need to avoid this duplicates but still make sure the message get delivered sooner or later. Any help is highly appreciated. Here is my routing table: thanks Marc ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } if (is_method("MESSAGE")) { log(1,"-------------------------------------------------------->WE HAVE A NEW MESSAGE\n"); route(MSILO_MESSAGE); } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } ##storing offline messages route[MSILO_MESSAGE]{ if (lookup("location")){ t_on_failure("FAIL_MESSAGE"); t_relay(); xlog("L_NOTICE","---------------------------------MI: ($mi), MB: ($mb)\n"); }else{ xlog("L_NOTICE","------------------MESSAGE PROCESSING: location not found, storing offline message ID: ($ru)\n"); m_store("$ru"); sl_send_reply("202", "ACCEPTED"); xlog("L_NOTICE","------------------MESSAGE PROCESSING: location not found, stored offline message ID: ($ru)\n"); }; exit; }; failure_route[FAIL_MESSAGE]{ xlog("L_NOTICE","------------------MESSAGE PROCESSING: user cannot be reached, storing offline message ID: ($ou)\n"); m_store("$ou"); t_reply("202", "ACCEPTED"); xlog("L_NOTICE","------------------MESSAGE PROCESSING: user cannot be reached, stored offline message ID: ($ou)\n"); exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); # MSILO - dumping user's offline messages m_dump(); log(1,"-------------------------MESSAGE PROCESSING: dumping offline messages\n"); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rtpproxy_offer(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { rtpproxy_answer(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

3 4

RTP to SRTP encryption with Kamailio.
by Dani Kamailio 04 Jul '14

04 Jul '14

Hello, I need to build a VoIP system who receives SIP and RTP traffic in a public IP and encrypt both of them with TLS and SRTP respectively. The main point is to have security inside of the local network (I know it may sound unuseful). So, I was trying to build the whole system in Kamailio but I got stucked with the RTP to SRTP bridge and I do not really know how to do it. I know that there are some modules like rtpproxy-ng and rtpengine as media relay that can handle SRTP... any ideas? I just want to make sure that I am in the right way. If it could be done in Kamailio, have I to write the code to encrypt RTP, like the algorithm or something? Would it be convenient to send the RTP packets to Asterisk? (and Asterisk would somehow encrypt them?) I am pretty new with all of this, any help will be apreciate, thank you.

2 2

no rtpengine.so module
by Yuriy Gorlichenko 04 Jul '14

04 Jul '14

Hello. I install kamailio 4.1.3 and it works fine. But I neen proxy DTLS SRTP to backend media servers through kamailio. rtpproxy no not work with dtls srtp? so I installed rtpengine but can not find rtpengine.so file for copiyng to kamailio modules directory. Does anyone can install rtpengine as kamailio module?

3 4

FW: Module TOPOH: Does not unmask CALL-ID on request BYE in one direction
by Samuel F. 04 Jul '14

04 Jul '14

Hi, We are setting up a Kamailio Proxy with the Topoh module in place masking everything including CALL-ID. It seems to be masking everything just fine except for CALL-ID in BYE requests (verified with PCAPs) when received from downstream. Upstream works fine. Looking at the source code for the module 4.1.2 we see the unmasking being done like this: if(dialog) { /* dialog request */ th_unmask_ruri(&msg); th_unmask_route(&msg); th_unmask_refer_to(&msg); if(direction==1) { th_unmask_callid(&msg); } } It only seems to be unmasking for one direction. From my understanding BYE can be initated from both directions? Might be something I don’t understand here how the source code works but I wanted to ask for any pointers of help here.. been troubleshooting this for a good while now. Kind regards, Samuel

3 2

Re: [SR-Users] UAC remote registration - refreshing database
by Dan Christian Bogos 04 Jul '14

04 Jul '14

Hey Alex, Many thanks for so fast answer. Could be I have missed the past interest (hence re-posting). Anyway, I wonder what would be the chances that we get Ovidiu's interest so he can port registrant module to kamailio maybe? Unfortunately without reloads it is hard to push remote registration in enterprise solutions. Thanks again, DanB On 03.07.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > On 07/02/2014 06:22 AM, Dan Christian Bogos wrote: >> >Anybody aware if it is possible to refresh the list of remote >> >registrations from the database without restarting the whole server? > This gets asked a lot, and the answer is no. But it is probably a widely > desired feature set by now. > > -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please > be kind to the English language: > http://www.entrepreneur.com/article/232906

3 2

Calls per second
by Olle E. Johansson 03 Jul '14

03 Jul '14

Hi! Have you implemented a per-customer rate limit in Calls per second? If so - how? I've played with ratelimit/pipelimit and it seems like I can define a database with one pipe per customer - but have to restart Kamailio to add customers. There are warnings for low timer settings, like 1 second, but I don't know how up-to-date those warnings are. I guess I could play with hash tables and implement something pike-like there, but it seems like a workaround for something pretty common. So the question remains - how are you limiting on a cps per customer? /O

6 15

sipcapture / raw_interface
by András Stefler 03 Jul '14

03 Jul '14

Hello all, I am new in using Kamailio. I have installed the version 4.0.1 and using it as SIP capture node. My server has lots of physical interfaces where I want to enforce capturing on one of the only. This is called 'csm1'. This is the corresponding part from kamailio.cfg modparam("sipcapture", "capture_node", "sipserv") modparam("sipcapture", "db_url", "mysql://****) modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 0) #modparam("sipcapture", "raw_socket_listen", "192.168.0.1:5060") modparam("sipcapture", "raw_interface", "csm1") modparam("sipcapture", "raw_ipip_capture_on", 0) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "promiscious_on", 1) modparam("sipcapture", "raw_moni_bpf_on", 1) ******************************** stefler@sipserv:~$ ifconfig csm1 csm1 Link encap:Ethernet HWaddr ac:16:2d:96:3f:e0 inet addr:1.0.8.1 Bcast:1.0.8.3 Mask:255.255.255.252 inet6 addr: fe80::ae16:2dff:fe96:3fe0/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 My problem is that capturing is activated on all of the interfaces and only on the selected one. Have you seen this problem before? Regards Andras Stefler

1 0

How to check message queue (buffer) in Kamailio
by AliReza Khoshgoftar Monfared 03 Jul '14

03 Jul '14

Hi, I had another simple question: In a kamailio server (proxy), how do I check the number of messages currently waiting for processing? Is there a variable that I can monitor, say, if I want to make a routing decision in my config based on the number of messages in the queue? Also, is it possible to get a head count by method? or is it only possible after fully parsing the message? I see that "ratelimit" module uses similar information, but I am not sure how to get the status of these queues that the module uses. Thanks

3 5

Server generated 408 time out goes to onreply route?
by Allen Zhang 03 Jul '14

03 Jul '14

Hi, Does local server generated 408 time out reply go to onreply_route? If so, can I change the status code to a 200 class code by doing: $rs = "202"? If I can change it, does it still go to failure_route? (It's not a 400 class response anymore.) The scenario I want to test it's easy to produce. Otherwise I would just try it...... Regards, Allen

2 1

Replys though onsend_route?
by Jöran Vinzens 03 Jul '14

03 Jul '14

Hi all, is there a possibility to pass reply and failure though onsend_route as well? We try to capture via siptrace the incoming and outgoing packet. So far this is only possible with Requests. I've found this: http://lists.sip-router.org/pipermail/sr-users/2012-May/073224.html Can anybody confirm this is still the behavior? Does anybody have a solution passing reply though onsend_route or any other way to capture the outgoing reply via siptrace? Thanks a lot for your help. BR Jöran Vinzens -- Jöran Vinzens - vinzens[at]sipgate.de <vinzens(a)sipgate.de> sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

2 3

how to remove non-digits using dp_translate
by Henry Fernandes 03 Jul '14

03 Jul '14

I'd like to use the dialplan module to remove all non-digits from a string. For example, "1-306-555-1234" becomes "13065551234" and "(306) 555-1234" becomes "3065551234". Unfortunately, I can't figure out the regular expressions for subst_exp and repl_exp. I can easily replace one instance of a non-digit, however I can't figure out how to replace all non-digits. -H

2 1

Making RLS presence work with Blink and Kamailio 4.1.4
by Alex Villacís Lasso 02 Jul '14

02 Jul '14

I am having trouble making all of the supposed features of Blink work with Kamailio 4.1.4. My kamailio.cfg file is attached. Specifically, what I am having trouble is with presence (the way Blink wants to implement it), and MSRP. Ordinary voice calls work correctly. With presence, I have managed to make XCAP storing and retrieval work, in the sense that a document that is being stored with PUT can be later retrieved with GET on the same URL. However, I cannot make Blink use this support in order to actually monitor presence. As far as I understand, when user A subscribes to the presence state of user B, and user B is registered, user B is supposed to receive a notification that user A requests authorization for presence subscription. However, this is not working so far when both A and B use Blink. From what I can see through a packet capture, neither adding a new contact on the Blink window or having existing contacts result in any SUBSCRIBE packets for the contact users being sent to the Kamailio instance. If the subscription is meant to be performed some other way, I am not seeing it being done either. Of course, this results in presence changes not being exchanged between contacts. The only SUBSCRIBE packet I see from Blink is the one for the SIP account used for Blink. With MSRP, I have copied the example from the msrp.so documentation, but I have replaced the authentication shown with a static password, with an actual query using auth_db.so, the very same query used to authenticate other SIP packets. Through the packet capture, when user A tries starting a MSRP chat with user B, an INVITE is sent from user A to Kamailio, which has a special check to NOT send this invite to Asterisk, but rather route it itself, and it gets "correctly" routed to user B. Then user B attempts to start a MSRP session with Kamailio (not user A) without authentication credentials, Kamailio challenges user B for the credentials, and... that's it. User B makes no attempt to repeat the MSRP session with the required credentials, and the communication eventually times out. The presence failure might have something to do with the following exchange I see in the packet capture: GET /xcap-root/pidf-manipulation/users/sip:avillacisIM@pbx.elastix.com/index HTTP/1.1 Accept-Encoding: identity Authorization: Digest username="avillacisIM", realm="pbx.elastix.com", nonce="U6xFz1OsRKNuU11dgljLCgGppv5lK6Eu", uri="/xcap-root/pidf-manipulation/users/sip:avillacisIM@pbx.elastix.com/index", response="3dfdf556c9812f4bebb8b0cc82578ef9", algorithm="MD5" Host: pbx.elastix.com:5060 Connection: close Accept: application/pidf+xml User-Agent: Blink 0.8.2 (Linux) HTTP/1.1 404 Not found Via: SIP/2.0/TCP 192.168.3.2:55567 Server: kamailio (4.1.4 (x86_64/linux)) Content-Length: 0 That is, the Blink client tries to fetch the pidf-manipulation document, and Kamailio answers with a 404. In other cases Blink has responded with a PUT of a minimal document at the required URI (which gets stored to the database), but in this case (pidf-manipulation) this is not done. I am not sure of what is supposed to happen here. This 404 is the same regardless of whether there are other (Blink or otherwise) SIP clients connected. I think this has been asked before, but is there a working implementation of MSRP+presence for Kamailio that works with Blink? Or failing that, a complete kamailio.cfg that has been shown to work with at least one MSRP implementation?

2 5

UAC remote registration - refreshing database
by Dan Christian Bogos 02 Jul '14

02 Jul '14

Hey Guys, Anybody aware if it is possible to refresh the list of remote registrations from the database without restarting the whole server? Ta, DanB

2 1

TCP rest by peer
by Rob Moore 02 Jul '14

02 Jul '14

Hi All, Since we've been using sipPhones as a proxy for registrations and calls to phones over TCP. I've been seeing a couple of these crop up in the logs. /usr/sbin/kamailio[4255]: ERROR: <core> [tcp_read.c:1131]: ERROR: tcp_read_req: error reading /usr/sbin/kamailio[4255]: ERROR: <core> [tcp_read.c:293]: error reading: Connection reset by peer (104) We haven't had any complaints of problems with the phones so if these are not a problem I'd like to know how to clean them up. Any suggestions on cause or solution would be fantastic. Thanks in advance. Rob Kind regards, Robert Moore Telephony System Architect [Description: Aeriandi] m. +44 (0)7766 838 040 t. +44 (0)845 108 0308 [cid:image002.png@01CF8C82.823C49E0]<http://www.retailweektechnologyawards.com/winners-2014> Aeriandi Ltd / Prama House, 267 Banbury Road, Oxford OX2 7HT / +44 (0)845 108 0308 / info(a)aeriandi.com<mailto:info@aeriandi.com> / aeriandi.com<http://www.aeriandi.com/> This transmission is intended solely for the addressee and may be confidential. If you are not the named addressee, or if the message has been addressed to you in error, you must not read, disclose, reproduce, distribute or use this transmission. Please delete the message or contact the sender. Delivery of this message to any person other than the named addressee is not intended in any way to waive confidentiality. Please note that neither Aeriandi Ltd nor the sender accepts any responsibility for any viruses, which may be transmitted, and it is your responsibility to scan attachments (if any).

2 3

msilo possible bug
by Kelvin Chua 02 Jul '14

02 Jul '14

Hi guys, i think i have stumbled on an msilo bug. m_dump() is not working. kamctl fifo get_statistics all | grep mess msilo:dumped_messages = 0 msilo:failed_messages = 0 msilo:stored_messages = 1 so here we can see there is a message to be delivered. i turned on debug mode, then the intended destination of message logs in. then this is what i see on logs DEBUG: msilo [msilo.c:1082]: m_dump(): dumping [1] messages for <sip:XX85X80X74@1X8.1X9.1X7.1X3>!!! DEBUG: msilo [ms_msg_list.c:169]: msg_list_check_msg(): checking msgid=405 DEBUG: msilo [ms_msg_list.c:207]: msg_list_check_msg(): msg already in sent list. DEBUG: msilo [msilo.c:1089]: m_dump(): message[0] mid=405 already sent. using ngrep, no message was ever seen on the wire. then, kamctl fifo get_statistics all | grep mess msilo:dumped_messages = 0 msilo:failed_messages = 0 msilo:stored_messages = 1 still the same. anybody experienced this? Kelvin Chua

2 2

Fedora 18-20 GeoIP missing?
by Roberto Fichera 02 Jul '14

02 Jul '14

Hi All, is there any reason why the GeoIP module is missing in the Fedora repo? Thanks in advance. Roberto Fichera.

2 2

Possible GPL violation by third party (4PSA)
by Matthew Costa 02 Jul '14

02 Jul '14

4PSA develop and sell a software suite called VoipNow Professional (version 2.5) which, amongst other things, includes it's own packaged copy of Kamailio. This version is derived from Kamailio version 1.5.4. The software installs a binary release from a 4PSA repository. The RPM information contains the following: # rpm -qi voipnow-kamailio-1.5.4-130312.41.rhel5 Name : voipnow-kamailio Relocations: (not relocatable) Version : 1.5.4 Vendor: Rack-Soft, Inc <devel(a)4psa.com> Release : 130312.41.rhel5 Build Date: Tue 12 Mar 2013 18:31:30 EST Install Date: Sun 02 Feb 2014 00:15:14 EST Build Host: rhel5-1.64b.build.4psa.ro Group : System Environment/Daemons Source RPM: voipnow-kamailio-1.5.4-130312.41.rhel5.src.rpm Size : 6599403 License: GPL Signature : RSA/SHA1, Tue 12 Mar 2013 18:35:28 EST, Key ID 42ba8c472f75de11 Packager : Rack-Soft, Inc <devel(a)4psa.com> URL : http://kamailio.org/ Summary : Kamailio, a very fast and flexible SIP Proxy I've made a direct request to 4PSA for the source code for this package, only to be told they don't release their sources. The request, and subsequent response, can be found at http://my.4psa.com/4psa/topics/where_can_i_find_the_source_code_for_the_voi pnow_2_5_x_modified_version_of_kamailio. For those interested, the following packages are included with VoipNow which are derived from GPL software. There is no source available for any of these from the vendor. # rpm -qa |grep voipnow voipnow-spandsp-0.0.6-100708.10.rhel5 voipnow-asterisk-1.6.1.20-130312.07.rhel5 voipnow-asterisk-addons-1.6.1-130131.30.rhel5 voipnow-php-2.5.5-130312.11.rhel5 voipnow-sox-14.3.1-100723.30.rhel5 voipnow-asterisk-sounds-1.4.22-100723.25.rhel5 voipnow-kamailio-1.5.4-130312.41.rhel5 voipnow-ejabberd-2.1.4-100723.29.rhel5 voipnow-asterisk-debuginfo-1.6.1.20-120417.09.rhel5 voipnow-asterisk-extra-2.5.5-130312.11.rhel5 My requests are falling on deaf ears. Does anybody know how to make them take this matter seriously? Cheers, Matthew Costa Infrastructure Manager <http://www.greenlight-itc.com/> Suite 703, Level 7 815 Pacific Highway Chatswood 2067 tel: 02 8412 0000 fax: 02 8412 0001

4 4

Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls
by Carlos Rangel 01 Jul '14

01 Jul '14

Hello List Hopefully someone can help. This is the problem when the call is hug up 20-30 seconds after it initiates. The call is only hung on when the remote extension initiates the call. If the remote extension receives the call there is no problem the call is not hung on. I changed the remote cisco phone for a yealink and it is the same behavior. It thought it was the phone. This is what I am using in kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ASTERISK #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD Remote User Internet Internal network Yealink IP TG28P ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN Thanks Carlos Rangel De: Carlos Rangel [mailto:crangel@globaltelesourcing.com] Enviado el: jueves, 26 de junio de 2014 01:27 p.m. Para: miconda(a)gmail.com; 'Kamailio (SER) - Users Mailing List' Asunto: RE: [SR-Users] Kamailio Freepbx Integration Dropping Calls Hi Daniel Thank you so much for your response. Here is the SIP trace of one of the calls, I am not sure where the call initiates but you can see at the end of the file in bold X-Asterisk-HangupCause: No user responding. I am not sure why is it sending this message though. The variables are Extension/Username=XXXXX Ext_IP= Public IP Internal_IP= Asterisk/Kamailio internal IP Sorry for the long file but again I am not sure where the call initiates This is the part where that call is hung on. U 2014/06/26 13:36:11.831965 Kamailio_IP:5080 -> Kamailio_IP:5060 BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0. Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47. Route: <sip:Kamailio_IP;lr=on;ftag=000653dc394000970f227678-1fafb4e2>. Max-Forwards: 70. From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4. To: "User" <sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2. Call-ID: 000653dc-3940000b-33caf1b2-20ccd185(a)192.168.0.22. CSeq: 102 BYE. User-Agent: FPBX-2.11.0(11.10.2). X-Asterisk-HangupCause: No user responding. X-Asterisk-HangupCauseCode: 18. Content-Length: 0. . U 2014/06/26 13:36:11.832260 Kamailio_IP:5060 -> 65.190.71.203:5060 BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0. Via: SIP/2.0/UDP Kamailio_IP;branch=z9hG4bKcf68.d6ef5aa9cc5bd0fb0ab13a563b7cf284.0. Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47. Max-Forwards: 69. From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4. To: "User" <sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2. Call-ID: 000653dc-3940000b-33caf1b2-20ccd185(a)192.168.0.22. CSeq: 102 BYE. User-Agent: FPBX-2.11.0(11.10.2). X-Asterisk-HangupCause: No user responding. X-Asterisk-HangupCauseCode: 18. Content-Length: 0. Descripción: Description: Description: DLR-Logo-No-TextCARLOS RANGEL | INFORMATION TECHNOLOGY DIRECTOR Global Telesourcing México, S. de R.L. de C.V. | Aarón Sáenz #1891-1 | Monterrey, N.L., México Direct 703 894 1667 | Mobile US 703 894 1667 | Mobile MX +52 1 812 000 7362 | crangel(a)globaltelesourcing.com ________________________________________________________________________________________________________________ The information contained in this e-mail and any attached documents may contain information that is confidential or otherwise protected from disclosure. If you are not the intended recipient of this message, or if this message has been sent to you in error, please immediately alert the sender by reply e-mail and then delete this message, including any attachments. Any dissemination, distribution or other use of the contents of this message by anyone other than the intended recipient is strictly prohibited. De: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] En nombre de Daniel-Constantin Mierla Enviado el: jueves, 26 de junio de 2014 03:12 a.m. Para: Kamailio (SER) - Users Mailing List Asunto: Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls Hello, can you gran the SIP trace on kamailio server for such case? You can use ngrep, like: ngrep -d any -qt -W byline port 5060 and send the output to the mailing list. You can replace any sensitive information (e.g., ip address) before sending to mailing list. The typical call drop after 30-40 secs is when ACK is not routed properly, but we have to see that in the sip trace. Cheers, Daniel On 25/06/14 18:50, Carlos Rangel wrote: Hello I have successfully (I believe) implemented Kamailio 4.1.4 integration with Freepbx 5.2.11 taking as a guide Daniel’s tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. I just did not create the voicemail tables because voice mail is handled by Freepbx. I installed the system in a separate box for testing and connected to the Freepbx Production server via IAX trunk. The system is behind a Cisco Firewall and looks like this Remote User Internet Internal network Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN I have configured the FW to allow UDP and TCP traffic from the corresponding IP as well as tfpt that is needed for the Ciscos to pick up the configuration from the server. I have a few remotes Cisco 7960 phones that can register remotely in Kamailio as long as the user is added with kamctl add user password and as long as the extension is created in Freepbx. The problem that I have is when try to make a call from the remote Ciscos the call is dropped after 30 or 40 seconds. I can see from the logs that the problem appears to be that the server is not receiving responses from the phone 06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on transmission 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response [2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Is this something that we can adjust in kamailio or could it be related to the FW configuration?? Sorry but I am very new to kamailio and sip. Thanks Carlos _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

installation siremis error
by Hossein Aghaie 01 Jul '14

01 Jul '14

hi after install kamailio 4.1.0 on ubuntu server,I try to install siremis-4.1.0 of this Doc : http://kb.asipto.com/siremis:install41x:main but to end setup,show this error : http://paste.fedoraproject.org/114029/ how to solved this problem? tnx

3 5

Kamailio Development Workshop, July 9-10, 2014, Paris, France
by Daniel-Constantin Mierla 01 Jul '14

01 Jul '14

Hello, I want to announce the next Kamailio Development Workshop, to take place in Paris, France, during July 9-10, 2014. This event is the next in its series targeting to show the internals of Kamailio and enable more people to become developers as well as let users of Kamailio to get more knowledge about the design of the application which can have relevant impact in operating deployment. Be aware that it is not a workshop about Kamailio installation and administration. Its content is about writing C code to extend Kamailio. More details can be found at: - http://www.kamailio.org/w/2014/06/kamailio-development-workshop-july-9-10-2… Similar to the past events, we are considering to have a social networking event in the evening of July 9 (most probably a dinner meetup at a nice place in Paris), which is open for everyone, each participant taking care of own expenses. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 2

how to compile module db_oracle on Centos 6.5
by Konstantin Khatskevich 01 Jul '14

01 Jul '14

Hello! 5. Installation Because it dependes on an external library, the oracle module is not compiled and installed by default. You can use one of the next options. * - edit the "Makefile" and remove "db_oracle" from "excluded_modules" list. Then follow the standard procedure to install Kamailio: "make all; make install". * - from command line use: 'make all include_modules="db_oracle"; make install include_modules="db_oracle"'. But "db_oracle: requires non-free Oracle client SDK" (http://lists.sip-router.org/pipermail/sr-dev/2013-October/021803.html) What can I do? -- Best regards, Konstantin Khatskevich

2 2

Changes in registrar module source code
by aawaise 01 Jul '14

01 Jul '14

Hello, I am a user of kamailio server. I am in need to pick domain of particular online users from location list. I found out that usrloc module is used for Location Table items. And we cant directly export parameters from usrloc module. I decided to use registrar module to approach domain parameter of usrloc module. I want to understand complete flow control in registrar module source code, so that I can add my commands to pick domain parameter from usrloc module. How can it be done ? Secondly, if I make a change in registrar module, to see its reaction, I will have to recompile kamailio server or is there any shortcut to immediately see effect of changes made in registrar module ?? Any help will be appreciated. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Changes-in-registrar-module-source-… Sent from the Users mailing list archive at Nabble.com.

2 1

Siremis 4.1.0 menu adjustment for different users
by Fred 01 Jul '14

01 Jul '14

Hi, I'm setting up a Registrar server using Kamailio 4.1.3 with Siremis 4.1.0 as WebGUI. On Siremis I struggle to set up different users which only see limited menu items. What I want to do is e.g. an user called "CustomerService" which only sees the "Subscriber_List" menu item and a second user called "SystemAdmin" which only sees the "Dispatcher_List" menu item. I tried different things with new roles, groups and adjustments in the menu administration but didn't manage to get the desired result. Does anyone have a hint for me if that is possible and how? Thanks Fred

1 0

Child Processes Stopped kamailio 4.0.6
by Jonathan Hunter 30 Jun '14

30 Jun '14

Hi Guys, Sorry thought I better raise this in new stream to avoid confusion. We have another crash today, not sure if the below is of much use without debug? Thanks Jon Jun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:788]: handle_sigs(): child process 18532 exited by a signal 11Jun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:09 pbx /usr/sbin/kamailio[18531]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:788]: handle_sigs(): child process 18532 exited by a signal 11Jun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18531]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18533]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18537]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18534]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18535]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18538]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18530]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18539]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18536]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 kamailio[2076]: ERROR: <script>: Failed Reg Block Traffic from 194.72.34.18Jun 26 09:38:09 pbx kamailio[2075]: ERROR: <script>: Failed Reg Block Traffic from 194.72.34.18Jun 26 09:38:09 pbx kamailio[2074]: ERROR: <script>: Routing Call To Asterisk serverJun 26 09:38:09 pbx kamailio[2074]: INFO: <script>: Routing Call to Dispatcher group 1 -Asterisk Media ServersJun 26 09:38:10 pbx kamailio[2072]: ALERT: <core> [main.c:788]: handle_sigs(): child process 2075 exited by a signal 11Jun 26 09:38:10 pbx kamailio[2072]: ALERT: <core> [main.c:791]: handle_sigs(): core was generatedJun 26 09:38:10 pbx kamailio[2072]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:10 pbx kamailio[2074]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2078]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2076]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2077]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2079]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2073]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2080]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2081]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2082]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2072]: ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: No such file or directory (2)Jun 26 09:43:33 pbx kamailio: INFO: <core> [tcp_main.c:4846]: init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected)

2 2

How to extract information from location table
by aawaise 30 Jun '14

30 Jun '14

I want to know is there any indirect method to extract information from location table that is kept by Usrloc module ?? I want to know about the username and domain name of users from Location Table on arrival of INVITE packet. How can it be done ? -- View this message in context: http://sip-router.1086192.n5.nabble.com/How-to-extract-information-from-loc… Sent from the Users mailing list archive at Nabble.com.

2 1

route [Location] Documentation
by aawaise 30 Jun '14

30 Jun '14

I want to know that how route [Location] works when an INVITE packet approaches kamailio server. Basically route [Location] gathers information whether callee is a valid user or not. Secondly if it is online or not. I want to understand this whole procedure. Any help will be appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/route-Location-Documentation-tp1286… Sent from the Users mailing list archive at Nabble.com.

2 1

Voice quality issues when registering to Kamailio
by Travis Dillon 30 Jun '14

30 Jun '14

Hello all, I have an environment setup as follows: VoIP routers -> Kamailio 4.0 -> Asterisk 11.6 When I register the phones to Asterisk the call quality is excellent. So to be clear, the call comes in the VoIP router, routes through Kamailio to the phone registered to Asterisk. When I register the phone to Kamailio the call quality is significantly reduced. So clarity here, the call comes in the VoIP router, routes through Kamailio to the phone registered to Kamailio. Why would the call quality diminish when registered to Kamailio? Where can I start looking for this issue? Thank you, Travis R. Dillon

2 1

UAC module refresh registration, could not reuse callid/fromtag !!!
by nevious 30 Jun '14

30 Jun '14

Hello all! I make kamailio1 remotely register to kamailio2. Here I set K1 modparam("uac", "reg_timer_interval", 15), and expires in uacreg as 30s. Then K1 should refresh registration after 15s. However the callid and from tag of every registration are different, and the Cseq just 10 or 11,never increased.In RFC3261, when UAC want refresh bindings, A UA SHOULD use the same Call-ID for all registrations during a single boot cycle. Is there any way to reuse the callid and fromtag during refresh registration ? Many thanks! And how to make the Cseq become 12 or more?

4 4

Strange application hang querying DB
by Tim Chubb 27 Jun '14

27 Jun '14

Hi, I'm currently wracking my brains trying to deduce where an application hang can be coming from. I have ported the PostGres schema to be compatible with MS Sql Server and have had no issues getting a working dev environment working. However when I try and use SQL Azure as the backend db as opposed to SQL Server 2012 (running locally on my dev machine) Kamailio will seem to hang after about 10 - 15 mins of flawless operation. I have recreated the SQL Azure schema from the working schema running on my dev machine and copied all data in a 1:1 manner so I can be certain it's not a data quality issue. I am using the same FreeTDS/UnixODBC config that I am using for Asterisk realtime (which doesn't exhibit this behaviour). The really strange thing is that I can see the queries being sent by Kmailio in the freetds trace log, and the response from SQL Azure, however its as if the db response handler is deadlocked/hung. The kamailio process continues to work fine, and as long as its not required to commiunicate with the DB all appears well i.e. no error messages in the log file. This behaviour only happens when im using SQL Azure as opposed to SQL Server, so there must be something causing the issue, that's unique to SQL azure as opposed to SQL Server. As far as I can tell this is not something I can create a backtrace for as no core dumps are produced, as far as I can tell there is no easy way debug what the lock is. If anyone has any suggestions I would be very grateful to hear them. One thing that occurs to me is that the db_unixodbc module doesn't have half as many parameters as its equilent in Asterisk, and that connection pooling may be in use, this could be a bit of a show stopper AFAIK you have to disable connection pooling with SQL server, certainly any of the guides I have come across configuring Asterisk to work with SQL server suggests that. Tim Chubb Developer [VS-logo-email] The Coach House, Heywood House, Park Lane, Heywood, Westbury, BA13 4NA Direct: +44 (0)333 0110164 Email: tim.chubb(a)voicesimplified.com<mailto:tim.chubb@voicesimplified.com> Web: www.VoiceSimplified.com<http://www.voicesimplified.com/> Voice Simplified Ltd (registered in England & Wales: 07171825) registered office: Curzon House, 2nd Floor, 24 High Street, Banstead, Surrey. SM7 2LJ. The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. If you are not the intended recipient, please return the message to the sender and delete it from your records.

2 3

Error when install Kamailio 4.1.4 on Mac OS 10.9.
by Son Truong 27 Jun '14

27 Jun '14

Hi Kamailio, - I meet an error when I am trying install Kamailio 4.1.4 on Mac OS 10.9. The error: touch /usr/local/lib64/kamailio/modules/usrloc.so install -m 755 usrloc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp.so install -m 755 xhttp.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp_rpc.so install -m 755 xhttp_rpc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xlog.so install -m 755 xlog.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xprint.so install -m 755 xprint.so /usr/local/lib64/kamailio/modules /bin/sh: -c: line 1: syntax error: unexpected end of file make: *** [install-cfg] Error 2 How can I fix it? I am new in Kamailio, I want to learn Kamailio. -- Thanks and best regards, Phạm Sơn Trường Android, iOS Team Lead at ME Corp. Email1: truonguit2010(a)gmail.com Email2: truongps(a)mecorp.vn Mobile: 01218816208

1 0

Processing of Invite Packet
by aawaise 27 Jun '14

27 Jun '14

When an invite packet is recieved by kamailio server, what processing is done on it ? Which modules and functions are called in its processing and in which sequence ? Thanking in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Processing-of-Invite-Packet-tp12867… Sent from the Users mailing list archive at Nabble.com.

1 0

Transaction Stateful Message Handling
by AliReza Khoshgoftar Monfared 26 Jun '14

26 Jun '14

Hi, I am trying to configure a SIP proxy architecture that is stateful with regard to transaction but not the dialogues, i.e. If an INVITE takes a specific route, the corresponding 180 Ringing, 200 OK and ACK shall traverse the same set of proxies, but the corresponding BYE can take another route (and obv it's own 200 OK takes the same route again) Which modules can help me with that? In specific which functions / modules are recommended to handle requests and responses? Currently, I use "t_relay()" from TM module to forward requests and "loose_route()" from RR module to handle replies, but it looks like that if a BYE ends in a proxy which its corresponding INVITE has not traversed, it is not forwarded by that proxy. Am I making a mistake in my routing? Isn't there a simple way to do transaction stateful routing in Kamailio? Thanks, Alireza

1 0

How to stop kamailio from send 408?
by Allen Zhang 26 Jun '14

26 Jun '14

Hi, when t_relay() times out, calling sl_reply_error() will send 408 to the client. But if I don't want to send 408, how do I interpret the error code t_relay() set? Regards, Allen

2 2

How to find whether or not certain header exists?
by Allen Zhang 26 Jun '14

26 Jun '14

Hi, If I normally receive a INVITE that has a header 'aaa' with value 'bbb', but it's possible that some INVITEs don't have this header. How do I determine in the config file if the header exists? To check the value of the header I do if($hdr(aaa) == "bbb"). But what if the header doesn't exist? Cheer, Allen

2 2

Kamailio Freepbx Integration Dropping Calls
by Carlos Rangel 26 Jun '14

26 Jun '14

Hello I have successfully (I believe) implemented Kamailio 4.1.4 integration with Freepbx 5.2.11 taking as a guide Daniel's tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. I just did not create the voicemail tables because voice mail is handled by Freepbx. I installed the system in a separate box for testing and connected to the Freepbx Production server via IAX trunk. The system is behind a Cisco Firewall and looks like this Remote User Internet Internal network Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN I have configured the FW to allow UDP and TCP traffic from the corresponding IP as well as tfpt that is needed for the Ciscos to pick up the configuration from the server. I have a few remotes Cisco 7960 phones that can register remotely in Kamailio as long as the user is added with kamctl add user password and as long as the extension is created in Freepbx. The problem that I have is when try to make a call from the remote Ciscos the call is dropped after 30 or 40 seconds. I can see from the logs that the problem appears to be that the server is not receiving responses from the phone 06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on transmission 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response [2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Is this something that we can adjust in kamailio or could it be related to the FW configuration?? Sorry but I am very new to kamailio and sip. Thanks Carlos

3 4

Trouble with rtpproxy
by Igor Potjevlesch 26 Jun '14

26 Jun '14

Hello, For a particular gateway, I force the RTPProxy. This work fine but on the other hand, in my side the RTPPRoxy response to the source IP address of SIP packet instead of the IP address in "Connection information". Is there a way to instruct the use of the IP address announce in the SDP instead of the last IP source of the last SIP packet? Regards, Igor.

2 3

segfault at 18 error 4 in auth_db.so on Version 4.0.6-1 Release
by Jonathan Hunter 26 Jun '14

26 Jun '14

Hi Guys, We had our kamailio instance crash today, it appeared to be after multiple registration failures, the resulting logs showing; Jun 25 15:47:55 kernel: : kamailio[14993]: segfault at 18 ip 00007f03102f9609 sp 00007fff1b346500 error 4 in auth_db.so[7f03102f1000+b000]Jun 25 15:47:55 /usr/sbin/kamailio[14989]: ALERT: <core> [main.c:788]: handle_sigs(): child process 14993 exited by a signal 11Jun 25 15:47:55 /usr/sbin/kamailio[14989]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 25 15:47:55 /usr/sbin/kamailio[14989]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLD Any comments welcome. thanks Jon

2 2

Syntax help in comparison of values of variables
by aawaise 26 Jun '14

26 Jun '14

I need some help regarding syntax. Can we put some comparisons on the values of these variables like if ("$ru"=="$fu") {xlog("both are same")}; If this is not the way then what is the way to do so ?? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Syntax-help-in-comparison-of-values… Sent from the Users mailing list archive at Nabble.com.

2 1

Extract information from invite packet
by aawaise 26 Jun '14

26 Jun '14

I want to log some information of callee from incoming invite packet. how can it be done. I want to get it logged in /var/log/messages. Any help will be appreciated. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Extract-information-from-invite-pac… Sent from the Users mailing list archive at Nabble.com.

2 2

Passing Command-Line Arguments to Kamailio Config Script
by AliReza Khoshgoftar Monfared 25 Jun '14

25 Jun '14

Hi, I had a simple question to which I could not find a clear answer in the documentation. How can I pass a command line argument to Kamailio config script? For example, assume that I run my kamailio daemon using script myconf.cfg: >>kamailio -f myconf.cfg Now I want to pass a value, say an IP address and use it inside the script. Sth like >>kamailio -f myconf.cfg --ip __MY_IP__ and then have an assignment of "$var=__MY_IP__" inside myconf.cfg Is there an easy way to do this?

2 2

Re: [SR-Users] IMS and SIP over WebSocket 500/Error
by DURECU, OLIVIER (OLIVIER) 25 Jun '14

25 Jun '14

Hi Carsten, I had a look Kamailio repo and did not find the configuration you proposed to post. Did I look at the bad place or didn't you find the time to do it ? Thanks, Olivier Hi Olivier, i will update the configuration files for WebSockets at the P-CSCF after KamailioWorld. The week after KamailioWorld, we are at another congress, but probably around easter, you should find a proper configuration for WebSocket on the P-CSCF (including media). Kind regards, Carsten 2014-04-01 12:02 GMT+02:00 DURECU, OLIVIER (OLIVIER) <olivier.durecu at alcatel-lucent.com>: > Hello all, > > I have installed Kamailio 4.1.2 in IMS configuration (3 Kamailio instances > to run in P/I/S-CSCF modes) > > Everything works fine with Native IMS clients (Boghe) > > I want now to introduce Web clients using WebSockets. I have added to the > P-CSCF Kamailio.cfg the websockets parts > Registration is Ok but I can't manage placing a call. > > With SIPoWS, the contact header is sip:olivier at 132.invalid. So when the > P-CSCF receives a terminating INVITE, its destination URI is > sip:olivier at 132.invalid , it tries to make a DNS resolution of 312.invalid > and it doesn't forward to the client saying the domain is unknown and can't > be routed > > I guess that something is missing in my configuration around > aliases/nathelper/outbound but didn't find what... > > Thanks for any help > > > Olivier DURECU > Bell Labs / IP Platform Research > Villarceaux Center > Tel: +33(0) 130772759 > GSM: +33(0) 683776482 > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users at lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

2 1

Setting priority (q) for kamctl ul add
by David Wilson 25 Jun '14

25 Jun '14

Thanks Daniel, Rather than changing the kamctl script, I can just run a command like: kamctl mi ul_add location <user> <uri> <expires> <q> 0 0 0 0 When I run this and set expires to 0 (as per the comment in kamctl, "expires 0 means persistent contact"), kamctl ul show shows expires=-1403662437 which siremis shows as 1970-01-01 08:00:00. Within 20 minutes, the entry is gone and I'm not sure why. Is the 8:00 (my timezone is UTC+8) causing a check for zero to fail or is there something else I'm missing? For reference, I'm using db_mode=2 for usrloc. Regards, Dave. On 20 Jun 2014, at 6:00 pm, sr-users-request(a)lists.sip-router.org wrote: > Date: Fri, 20 Jun 2014 07:38:50 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] Setting priority (q) for kamctl ul add > > Hello, > > On 20/06/14 06:16, David Wilson wrote: >> Hi All, >> >> I'm trying to add a permanent usrloc entry via kamctl ul add. >> >> This works, but the created entry has a q value of 1.0 which is higher than I need. >> >> Is there a way to either: >> >> 1. Specify a q value when using kamctl ul add, or > apparently the Q is hardcoded, a way to fix it is to edit kamctl (which > is a shell script), search for 'ul_add' and update the 1.0 value in that > command. > >> 2. Edit the q value of an existing record by using a kamctl command. >> > Maybe removing and add it it with same other attributes... > > Cheers, > Daniel

2 2

Adding/manipulating headers in CANCEL
by Örn Arnarson 24 Jun '14

24 Jun '14

Hello, I'm running Kamailio 4.0.4, and I'm running into problems when trying to insert or update headers when the method is CANCEL. It just seems that nothing at all happens to the message. Is this a bug or a design choice? Regards, Örn

2 4

How to route on whether INVITE is for MSRP
by Alex Villacís Lasso 24 Jun '14

24 Jun '14

So far, I have a kamailio.cfg that routes INVITEs between Asterisk and the external networks. Now I want to start adding MSRP support. However, with my current configuration, the test client (Blink) sends an INVITE with a SDP payload that specifies msrps media. This gets routed to Asterisk, and it gets promptly rejected with "Not Acceptable Here". I need to make Kamailio route the request by itself if the INVITE is for MSRP, and delegate to Asterisk for other INVITEs. What is the most elegant way to test for the INVITE packet to see if it is MSRP? This testing should be at least mentioned in the documentation for the msrp.so module.

2 1

Setting up two Kamailio Proxies in Tandem
by AliReza Khoshgoftar Monfared 23 Jun '14

23 Jun '14

Hi, I am trying to build a simple architecture like this: UAC <--> Proxy1 <--> Proxy2 <--> UAS (each on a different VM) Is it possible to have this architecture built without any "registrar" module? I have simulated my UAC and UAS using SIPp and the routing in Proxy1 and Proxy2 is to forward to the next hop. for example in Proxy1, I have: *route{ record_route(); if (loose_route()) { route(FORWARD); break; } if (uri == myself){ setdsturi("sip:__ADRESS_OF_PROXY2__"); route(FORWARD);* * break;* * } route(FORWARD); }route[FORWARD]{ if (!t_relay()) { sl_reply_error(); }}* I simulate packets in UAC in the following manner: *sipp -sn uac -rsa __ADDRESS_OF_PROXY1__ __ADDRESS_OF_PROXY1__* The problem is that my first proxy (Proxy1) is always returning "404 Not Found" on any INVITE and never forwards to Proxy2, even though I have asked it to do so in "if(uri==myself)". Am I missing something in connecting the proxies in such a simple architecture?

1 0

kamailio not sending 202 after 408 is sent
by Allen Zhang 23 Jun '14

23 Jun '14

Hi all, I have an edge proxy (E) and a registrar (R) behind it. In R, if a MESSAGE failed to deliver for any reason, R stores the MESSAGE in msilo. If the MESSAGE timed out, R sends a 408 time out first and then send a 202 Accepted after the MESSAGE is stored. The problem is, E happily forwards the 408 to the sender of the MESSAGE, but decides not to forward the 202. The log: DEBUG: t_reply_matching: hash 16662 label 0 branch 0 DEBUG: tm [t_lookup.c:1032]: DEBUG: t_reply_matching: reply matched (T=0x7f99eebb2ae0)! DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=3 global id=3 T end=0x7f99eebb2ae0 DEBUG: tm [t_reply.c:2085]: DEBUG: reply_received: org. status uas=408, uac[0]=408 local=0 is_invite=0) DEBUG: tm [t_reply.c:1207]: ->>>>>>>>> T_code=408, new_code=202 DEBUG: tm [t_reply.c:1706]: DEBUG: relay_reply: branch=0, save=0, relay=-1 Can I change this behaviour to also forward the 202 withouting hacking in the source code? Regards, Allen

2 4

Specifying Upstream and Downstream Servers for a simple SIP proxy
by AliReza Khoshgoftar Monfared 23 Jun '14

23 Jun '14

Dear Kamailio users, I am trying to set up a simple scenario as follows: UAC --> Proxy_1 --> Proxy_2 --> Proxy_3 --> UAC I am new to Kamailio and had the following basic questions that came to my mind after reading the documentation an the default kamilio.cfg config script: 1) Is there any good example of the above scenario in which the UAC and UAS (caller and callee) are modeled using SIPp and proxies are simple kamailio instances run on different machines that simply forward the SIP packets? 2) In specific, I want the proxies to simply forward the messages to their downstream (I do not care about registration or other operations and look for a minimally working simple SIP scenario). Is there any example of such configuration? 3) Let's imagine Proxy_2 in the above example, in the default config, it looks like that with modifying line 164, "listen=udp:10.0.0.10:5060" I can specify the address of its upstream, Proxy_1, but how and where shall I set the specifications of the downstream, Proxy_3? I suspect it is somewhere in the Routing Logic block (line 449) but not sure how it is exactly done. I see a "route(SIPOUT)" call for example but I am not sure how and where the value of SIPOUT is modified. Is it a representation of the downstream servers in the config file? Thanks, Alireza

2 4

Is Kamailio PacketCable 2.0 compatible
by Horo, Priyam 23 Jun '14

23 Jun '14

I was looking at installing Kamailio and was wondering it is Packetcable 2.0 compatible? Thanks!

2 1

Disabling rtpengine abrt handler
by Alex Balashov 22 Jun '14

22 Jun '14

Hello, Despite this fix https://github.com/sipwise/rtpengine/commit/cbe1f805363b3d6a117e9e5425d7994… I am continuing to experience periodic crash problems under high loads with rtpengine. This is after upgrading mediaproxy-ng -> rtpengine. Unfortunately, after this upgrade, it is impossible to tell the cause of the crash because the [seemingly] newer rtpengine SIGABRT and/or crash handler is not running properly and is getting in my way: Jun 20 15:47:55 rtphost abrtd: Directory 'ccpp-2014-06-20-15:47:54-1123' creation detected Jun 20 15:47:55 rtphost abrt[2159]: Saved core dump of pid 1123 (/usr/sbin/rtpengine) to /var/spool/abrt/ccpp-2014-06-20-15:47:54-1123 (97013760 bytes) Jun 20 15:47:55 rtphost abrtd: Executable '/usr/sbin/rtpengine' doesn't belong to any package and ProcessUnpackaged is set to 'no' Jun 20 15:47:55 rtphost abrtd: 'post-create' on '/var/spool/abrt/ccpp-2014-06-20-15:47:54-1123' exited with 1 Jun 20 15:47:55 rtphost abrtd: Deleting problem directory '/var/spool/abrt/ccpp-2014-06-20-15:47:54-1123' I am not sure why this script is exiting, but the result is that I have no /var/spool/abrt/ccpp-2014-06-20-15:47:54-1123 and no core dump to work with. This is on a standard minimal CentOS 6.5.x install. Is there any way to either disable this signal handler and just let it dump core normally, or, otherwise, to definitively fix the crash handler? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

3 7

MSILO storing wrong content-type.
by Prasun Bheri 22 Jun '14

22 Jun '14

Hi All, I have Kamailio 4.0.0 configured with MSILO in centos 6. When a Message destined to offline recipient is received by kamailio, MSILO stores the message with wrong content type. In my scenario Incoming message has content type "message/cpim" where as in database its is stored as "text/plain". and obviously when recipient registers back in the stored message is sent with content type as "text/plain". I have attached a wireshark capture where in you can see packet #23 (recived Message) and #88 (dispatched Message). Could you guys let me know how to fix this. Thanks & Regards, Prasun

1 0

UAC module refresh registration, could not reuse callid/fromtag !!!
by nevious 21 Jun '14

21 Jun '14

Hello all! I make kamailio1 remotely register to kamailio2. Here I set K1 modparam("uac", "reg_timer_interval", 15), and expires in uacreg as 30s. Then K1 should refresh registration after 15s. However the callid and from tag of every registration are different, and the Cseq just 10 or 11,never increased.In RFC3261, when UAC want refresh bindings, A UA SHOULD use the same Call-ID for all registrations during a single boot cycle. Is there any way to reuse the callid and fromtag during refresh registration ? Many thanks! And how to make the Cseq become 12 or more?

1 0

Kamailio clean unconfirmed dialog
by Arun Kumar 21 Jun '14

21 Jun '14

Hi Server : kamailio 4.0 Configuration :Outbound with engage_media_proxy (See the attachment ) Shared Mem : 1024 MB Dialog TimeOut : 600 (Seconds) Problem : Dialog's are not cleared sometimes , So after sometime server is not responding (out of memory error), After investigating long time , the problem is non confirmed dialog's are not cleared even dialog timeout Thanks Arunkumar

2 4

Re: [SR-Users] SIP Trunk
by Keith 20 Jun '14

20 Jun '14

Hi, So when I use the uac_reg_request_to, for some reason it sends an invite and not a register?? This means I'm not getting a 407 back to the use sac_auth in the failure route. Any ideas? Thanks Keith

1 0

mediaproxy-ng query
by Alex Balashov 20 Jun '14

20 Jun '14

Hello, I'm invoking mediaproxy-ng with rtpproxy_offer/answer("ow") and am getting a scenario where: INVITE is forwarded to signalling gateway xxx.xxx.xxx.xxx, which returns an SDP answer of yyy.yyy.yyy.yyy:51964. rtpproxy is invoked in both directions. However, media forwarded from media endpoint A is sent to xxx.xxx.xxx.xxx:51964 (the correct port, but to the signalling gateway), rather than yyy.yyy.yyy.yyy:51964, which is what's specified in the SDP answer in the 200 OK from xxx.xxx.xxx.xxx. No media is coming back to mediaproxy-ng from yyy.yyy.yyy.yyy:51964. The cause of this is a separate and logically unrelated issue. Nevertheless, I am curious as to why mediaproxy-ng doesn't send media from the near end to yyy.yyy.yyy.yyy:51964--in other words, to the specified SDP endpoint on the far end, rather than to the signalling gateway. I initially imagined that this might be due to the "w" option to rtpproxy_offer(), which the documentation says forces symmetric RTP. However, this behaviour is not elaborated upon or defined. I tried removing the 'w' option but seem to be getting the same result. Is this a default behaviour of COMEDIA-style source port latching? In other words, does mediaproxy-ng need to latch onto incoming RTP from yyy.yyy.yyy.yyy:51964 before it will send media there, as opposed to the signalling gateway? If so, is there any way this behaviour can be disabled, so that mediaproxy-ng simply follows SDP in all cases? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 4

Kamailio b2bua
by Grant Bagdasarian 20 Jun '14

20 Jun '14

Hello, OpenSIPS has a module for B2BUA. Does Kamailio have a similar one or one that comes close to it? Perhaps by combining modules to make Kamailio act as a B2BUA?

3 4

Initial Install
by Phil Daws 20 Jun '14

20 Jun '14

Hello and good evening all! Am trying to build a Kamailio installation in my lab and have hit the first hurdle; database build. I have a three node MariaDB cluster in place and attempting to get kamdbctl create to set up the tables. What seems apparent is that command requires the database to not already exist. Where can I find the necessary schema to load into MariaDB myself please ? Thank you. Phil

2 2

Setting priority (q) for kamctl ul add
by David Wilson 20 Jun '14

20 Jun '14

Hi All, I'm trying to add a permanent usrloc entry via kamctl ul add. This works, but the created entry has a q value of 1.0 which is higher than I need. Is there a way to either: 1. Specify a q value when using kamctl ul add, or 2. Edit the q value of an existing record by using a kamctl command. Cheers, Dave.

3 2

Starting Kamailio SIP server
by Horo, Priyam 19 Jun '14

19 Jun '14

Hi, I am following the quick-start installation guide at http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x and I am stuck when I start the kamailio, using /etc/init.d/kamailio start it gives me an error of "invalid configuration file". I am using the /usr/local/etc/kamailio/kamailio.cfg file which came along with tar.gz file. I do not want sql and no authentication to begin with. What else needs to be modified or is required in the config file. Thanks!

1 0

Error codes with management commands
by PIERRE Laurent 19 Jun '14

19 Jun '14

Hello, It's unfortunate that error codes are not included with management commands. For example, if you tape " kamctl diaplan rm rule 2 999 " instead of " kamctl diaplan rmrule 2 999 " then all the entries in dialplan table will be deleted. In fat the system perform "kamctl diaplan rm" It's very annoying . -- Laurent PIERRE http://www.linkedin.com/in/lpierre

1 0

Unable To Start MSRP Session
by Megha Patil 19 Jun '14

19 Jun '14

Hi, I am trying to establish a chat session(MSRP) with two sipclients endpoints by registering to kamailio server(4.0.0), but I am getting 500 Internal server error(Reason: SIP ;text="media stream failed to start" ;cause=500). What might be the problem?? Does it mean that kamailio can not handle msrp session? If it can handle, how to enable it? I have attached the captures with this mail. Please help me in this regard.

1 0

Re: [SR-Users] SIP Trunk
by Keith 19 Jun '14

19 Jun '14

Hi, Thanks Daniel, however don't I need to send a register to get the 401 back? If so how do I do this bit? Thanks Keith

2 1

Possible Oversight, but CDRs are not being written
by Joel White 19 Jun '14

19 Jun '14

I made a significant change to the way the system I am building is designed. I removed the Database to a redundant cluster and not on the same server as the kamailio install. I changed all references to the DBURL to reflect the IP of the new cluster. I also changed the SQLOPS statement for CDRs to reflect the new IP modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@10.11.12.5/kamailio") All ACC transactions are being recorded to the new DB and the configuration is running well with the exception of CDRs. What am I missing that would prevent writing of CDRs to the DB? I did check connectivity from the Kamailio server to the DB server as well as access to the cdrs table

2 3

Cisco 79XX Series Phones and TLS integration with kamailio
by Jonathan Hunter 19 Jun '14

19 Jun '14

Hi All, As you guys might remember I have been doing alot of work with these legacy handsets recently, in particular the Cisco IP phones 7945G and 7965G. They work well now with kamailio, using UDP or TCP as the transport protocol. I am now looking to implement SIP over TLS with them, and wondered if anyone had completed the same, as it appears they were designed very much to support SIP over TLS and SRTP but with Cisco call manager and not other SIP devices. I am trying to understand if its possible to integrate with kamailio, as from Cisco documentation it appears a CTL file (client trust list) is required which seems to be generated by cisco software, and the handset needs it before it will try to initiate a SIP connection over TLS. I have implemented TLS /SRTP with Cisco SPA's, Bria etc on kamailio fine, so its more a question around the handset and if anyone has achieved this. Thanks Jon

3 8

Internal server error
by Deepak Bhagwat 19 Jun '14

19 Jun '14

Hi all, I am trying to establish a chat session(MSRP) with two sipclients endpoints by registering to kamailio server(4.0.0), but I am getting 500 Internal server error. What might be the problem?? Does it mean that kamailio can not handle msrp? If yes how to enable it in the server? Packets captured are attached. Please help me in this regard. -- Thanks Deepak Bhagwat

1 0

dialog module -- timeout parameter
by Måns Nilsson 19 Jun '14

19 Jun '14

Hi, I need to allow arbitrarily long calls while still using the dialog module. The module documentation does not mention a maxiumum value (2^32 sec?) nor a setting to disable the timeout functionality. I'd guess that setting it to 0 is a common convention in such cases, but I'd rather get it verified. Anything above 10 years is probably a safe bet, so a 32-bit unsigned int is quite fine. :-) Is there a max value? Is there a way to disable the check? -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 I think I'll KILL myself by leaping out of this 14th STORY WINDOW while reading ERICA JONG'S poetry!!

2 1

One way Audio with kamailio -Rtpproxy
by master voip 18 Jun '14

18 Jun '14

Hi can some one help me . I am using kamailio 4.1.3 and having one way audio issue. Scenario is Pjsip 2.0 based UA (private IP)------->Router(publicIP)--------> Kamailio (with RTP proxy)-----------> Third Party_Sip Server ------>PSTN UA is registering fine with sip server but when i make a call they can hear but i can not hear the other party . What i found is in SDP first c=IN is private ip of UA and second c=IN is public ip of UA i dont know if this is the cause When I trace the call on sip server , sip server is sending RTP to UA private ip so it is not reaching kamailio back to forward to UA . I am attaching the kamailio configuration file as well as SIP/SDP trace user agent private ip : 192.168.1.4 User agent public ip : 61.61.61.61 Kamailio IP : 81.81.81.81 Sip Server IP: 71.71.71.71 (* ip addresses are not actual ips) Please reply with detailed instructions to fix this isue config file: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # #!define WITH_NAT # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ' '; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ' '; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=3 log_stderror=no #!endif memdbg=1 memlog=1 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=7878 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" loadmodule "nat_traversal.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 7) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication # Below line commented by test-MATHUR as PC-2-PC calls are not getting through #route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { sl_send_reply("100", "Trying"); setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { xlog("testM : Unable to relay !!!! :{ \n"); sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp )\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$s i:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { #xlog("testM : Got BYE from some PSTN/UAC.\n"); setflag(FLT_ACC); # do accounting ... #xlog("testM : Did accounting.\n"); setflag(FLT_ACCFAILED); # ... even if the transa ction fails #xlog("testM : Calling route{NATMANAGE} from WI THINDLG.\n"); #t_newtran(); #t_reply("200", "OK"); #xlog("testM : Called route{NATMANAGE} for BYE froom WITHINDLG.\n"); #exit; } 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 R FC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... i gnore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } #if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) #{ # rtpproxy_manage(); # return; #} rtpproxy_manage("z90"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { add_rr_param(";nat=yes"); } } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n" ); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bu g in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_i p) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } ------------------------------------------------------------------------------------------------------------------ SIP TRACE REGISTER sip:71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW To: "57778" <sip:test@71.71.71.71> Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 CSeq: 62147 REGISTER User-Agent: pj_arubaslim-16/r2 Contact: "57778" <sip:test@192.168.1.4:58931;ob> Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 P-hint: outbound SIP/2.0 407 Proxy Authentication Required CSeq: 62147 REGISTER Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 To: "57778" <sip:test@71.71.71.71>;tag=170604141058 Proxy-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301389817101016806045824224" Content-Length: 0 REGISTER sip:71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW To: "57778" <sip:test@71.71.71.71> Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 CSeq: 62148 REGISTER User-Agent: pj_arubaslim-16/r2 Contact: "57778" <sip:test@192.168.1.4:58931;ob> Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Proxy-Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301389817101016806045824224", uri="sip:71.71.71.71", response="0ca765f266b6bfc8ea0ef00227272393" Content-Length: 0 P-hint: outbound SIP/2.0 200 OK CSeq: 62148 REGISTER Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 To: "57778" <sip:test@71.71.71.71>;tag=170604141059 Contact: "57778" <sip:test@192.168.1.4:58931;ob>;expires=600 Expires: 600 Content-Length: 0 SUBSCRIBE sip:test@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC To: "57778" <sip:test@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB CSeq: 10987 SUBSCRIBE Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: pj_arubaslim-16/r2 Content-Length: 0 P-hint: outbound SIP/2.0 401 Unauthorised CSeq: 10987 SUBSCRIBE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=170605141000 Content-Length: 0 WWW-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301390017101016806050024224" Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> SUBSCRIBE sip:test@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC To: "57778" <sip:test@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB CSeq: 10988 SUBSCRIBE Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: pj_arubaslim-16/r2 Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301390017101016806050024224", uri="sip:test@71.71.71.71", response="15ee3de757c18e084fca607438017093" Content-Length: 0 P-hint: outbound SIP/2.0 200 OK CSeq: 10988 SUBSCRIBE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=170605141001 Expires: 3600 Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> NOTIFY sip:test@71.71.71.71 SIP/2.0 CSeq: 100 NOTIFY Via: SIP/2.0/UDP 71.71.71.71:5060 From: "57778" <sip:test@71.71.71.71>;tag=170605141001 Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Contact: <sip:71.71.71.71:5060;transport=udp> Subscription-State: active Event: message-summary Content-Type: application/simple-message-summary Content-Length: 65 Messages-Waiting: no Message-Account: sip:test@71.71.71.71 SIP/2.0 404 Not here CSeq: 100 NOTIFY Via: SIP/2.0/UDP 71.71.71.71:5060;rport=5060 From: "57778" <sip:test@71.71.71.71>;tag=170605141001 Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Server: kamailio (4.1.3 (i386/linux)) Content-Length: 0 INVITE sip:919000000002@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23382 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: pj_arubaslim-16/r2 Content-Type: application/sdp Content-Length: 356 P-hint: outbound v=0 o=- 3612003234 3612003234 IN IP4 81.81.81.81 s=pjmedia c=IN IP4 192.168.1.4 t=0 0 m=audio 43314 RTP/AVP 3 18 0 8 101 c=IN IP4 81.81.81.81 a=rtcp:43315 a=sendrecv a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes SIP/2.0 407 Proxy Authentication Required CSeq: 23382 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Proxy-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301391017101016806051024224" Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> ACK sip:919000000002@71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23382 ACK Content-Length: 0 INVITE sip:919000000002@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23383 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: pj_arubaslim-16/r2 Proxy-Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301391017101016806051024224", uri="sip:919000000002@71.71.71.71", response="4edf5e2efd1e8e03bdbe4a693fdfee4c" Content-Type: application/sdp Content-Length: 356 P-hint: outbound v=0 o=- 3612003234 3612003234 IN IP4 81.81.81.81 s=pjmedia c=IN IP4 192.168.1.4 t=0 0 m=audio 43314 RTP/AVP 3 18 0 8 101 c=IN IP4 81.81.81.81 a=rtcp:43315 a=sendrecv a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes SIP/2.0 183 Session Progress CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 180 Ringing CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 200 OK CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ACK sip:71.71.71.71:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.e624c28002d41f48ae76813b26e516f3.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJ8XsuToNj.92GHJLiuK-JZ5T5PpfTbhr Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23383 ACK Content-Length: 0 BYE sip:71.71.71.71:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23384 BYE User-Agent: pj_arubaslim-16/r2 Content-Length: 0 SIP/2.0 200 OK CSeq: 23384 BYE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

1 0

Trigger m_dump() with REGISTER
by Marc M. 18 Jun '14

18 Jun '14

Hi, I have a basic MSILO setup up and running. For the moment every time the client sends a REGISTER, m_dump() is called. I would like to achieve the following: The clients are registering with either 200 or 300 sec expiration time. I would like to dump the messages only when the client is registering for 300 secs. Only a REGISTER with an expiration time of 300 should trigger m_dump(). Can you help me with this? Any help or idea is appreciated. thanks Marc

1 0

Re: [SR-Users] SIP Trunk
by Keith 18 Jun '14

18 Jun '14

Hi, I have a carrier who requires SIP authentication on calls, can anyone point me in the right direction? Normally in Asterisk or Freeswitch you can authenticate a peer, but can't see how to do this in Kamailio. Thanks Keith

2 1

Use of asterisk * in Kamailio routing
by Joel White 18 Jun '14

18 Jun '14

I am having an issue using an asterisk * in the kamailio routing. How would I implement this? Thank you in advance

3 6

JSON RPC Usage in Kamailio
by Jaytirth S Khairatkar 18 Jun '14

18 Jun '14

Hello, Can anyone give me a link or document which explains how to use the JSON and JSON-RPC module in Kamailio SER? I'll greatly appreciate it. Thanks and regards Jaytirth *Disclaimer* ------------------------------------------------------------ Notice: This e-mail and / or attachments (communication) may contain information that is confidential and proprietary to The Tata Power Company Limited, Strategic Engineering Division (Tata Power SED). If you are not the intended recipient of this communication, please do not use or disseminate the information contained therein, notify the sender by e-mail or telephone and delete the communication (permanently) from your system. Any information in this communication that does not relate to the official business of Tata Power SED shall be understood as neither given nor endorsed by it. Please note that e-mails are susceptible to change and Tata Power SED shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt. This communication is not guaranteed to be free from computer viruses and it is recommended that you check for all viruses before download. Print this communication only if absolutely essential. Thank you for your co-operation. ------------------------------------------------------------

1 0

s-cscf - limit of entries for 1 contact
by Ciprus, Daniel 17 Jun '14

17 Jun '14

All, is any of you aware about limitation how many reg entries can be stored in s-cscf for 1 contact ? From what I'm observing here : s-cscf can have up to 12 reg entries for 1 contact and then other entries are silently discarded. Use case: - crash of app happens 12 times - app retries to register - s-cscf sends back 200 OK with 13 contacts - another app registers and tries to send SIP MESSAGE to 1st app Result: message never gets delivered and tcpdump shows 12 messages sent out to p-cscf but not 13th one which is actually live session between client and proxy. thanks Dan ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 2

Install SEMS
by Agiftel 17 Jun '14

17 Jun '14

Hi All, i am trying to figure out how to get SEMS and install it. If i go to iptel website, documentations and last versions give always back NOT FOUND. Is this project still maintaned?/avaible? Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Install-SEMS-tp128500.html Sent from the Users mailing list archive at Nabble.com.

2 1

Re: [SR-Users] LCR from URI (CLI)
by Keith 17 Jun '14

17 Jun '14

Hi all, I have an issue where I have two carriers, one only accepts +44 then the CLI and the other only accepts the number with the leading zero dropped. I am using LCR to route to carriers, however I can only switch carriers by manually altering the CLI. Is there a way I can manipulate the CLI based on the destination the call is being sent to? Thanks Keith

2 1

Problems to start kamailio
by Carlos Rangel 17 Jun '14

17 Jun '14

Hello I am having problems to start kamailio 4.1.3 on a fresh centos 6.5 install with Asterisk 11.9 following the tutorial. Everything has gone well except that I cannot start kamailio. Basically I have two problems. The message that I got from the console when I try to start is [root@gt-ev-i08 kamailio-4.1.3]# kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed An when I look at the error messages in var/log/messages I see the following errors. Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: syntax error Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: bad command Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: syntax error Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: missing '(' or ')' ? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 16: '('')' expected (function call) Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 16: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 18: bad command I did not touch that section of the kamailio.cfg file. I am really lost now. Hopefully someone can help. Thanks Carlos

2 2

Unable to make calls within the extensions
by Chandramouli P 17 Jun '14

17 Jun '14

Hi, I am new to Kamailio. I started RTPProxy using "rtpproxy -A 54.85.12.15 -F -l 10.0.0.122 -s udp:localhost:7722" command and see that my sip phones are registered with Kamailio. I am able to see using "kamctl ul show" command. But, I am unable to establish call between my registered extension through Asterisk using Kamailio. I see that calls are hitting my Asterisk server when I call from extension. I am not getting any audio and the other extension is also not at all ringing. I could not able to figure out where I am doing the mistake? I am not sure whether the mistake is in Kamailio or Asterisk. In my Asterisk server, I am seeing the correct configuration using "odbcinst -q -d" and "isql -v MySQL-asterisk chandra test123456" and "odbc show (At CLI>)". Please find the below environment: Operating System: Ubuntu 14.04 Server (64-Bit) Kamailio: 4.0 Asterisk: 11.10 Database: MySQL (UNIX ODBC) Environment: Amazon EC2 Follwed Links: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb http://pastebin.com/VPpfErYn Daniel's Patched RTPProxy Installation (Please note that My Asterisk and Kamailio servers are behind NAT in Amazon EC2) Below is my configuration: *http://pastebin.com/fr8m9gr7 <http://pastebin.com/fr8m9gr7>* *Note:* I inserted the records in to the respective tables with "cmp" as context. When I call from 100 to 500, I am not getting any sound and another extension is not ringing and getting the below messages at Asterisk CLI: *http://pastebin.com/y4tXXrnF <http://pastebin.com/y4tXXrnF>* Any update would be appreciated. Thanks in advance. Regards, Chandra.

3 5

Problem with CANCEL handling
by Mordechay Kaganer 17 Jun '14

17 Jun '14

B.H. Hello, all! I'm trying to set up kamailio as a simple load balancer proxy between PSTN gateway and a cluster of asterisk servers - rather common use case IMHO. I'm completely new to kamailio. I have started with the default kamailio.cfg that comes with Ubuntu package and added the dispatcher module as described in the docs ( http://www.kamailio.org/docs/modules/4.0.x/modules/dispatcher.html). At the end, i got this up and running, but with one exception: if incoming call is not answered by the asterisk end (i.e. it is in ringing state), and then the gateway sends CANCEL request, kamailio is unable to recognize the transaction and is not forwarding the request to asterisk. The result is: the call is left "dangling" in the PBX and is not properly hang up. When the gateway sends CANCEL to kamailio, i get the following log messages: 4(25442) DEBUG: tm [t_lookup.c:769]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: searching on hash entry 31287 4(25442) DEBUG: tm [t_lookup.c:492]: matching_3261(): DEBUG: RFC3261 transaction matching failed 4(25442) DEBUG: tm [t_lookup.c:859]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: no CANCEL matching found! 4(25442) DEBUG: tm [t_lookup.c:861]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT completed 4(25442) DEBUG: tm [tm.c:1039]: w_t_lookup_cancel(): lookup_original: t_lookupOriginalT returned: (nil) 4(25442) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=485 a=25 n=xdbg 4(25442) DEBUG: <script>: is_method(CANCEL): t_check_trans()==false 4(25442) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=487 a=2 n=exit The relevant script snippet is standard from the distro package: # CANCEL processing if (is_method("CANCEL")) { xdbg("is_method(CANCEL): entry"); if (t_check_trans()) { xdbg("is_method(CANCEL): t_check_trans()==true"); route(RELAY); } else { xdbg("is_method(CANCEL): t_check_trans()==false"); } exit; } How can i fix this? Thanks! -- משיח NOW! Moshiach is coming very soon, prepare yourself! יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!

1 2

Re: [SR-Users] sip aliases
by Anna Schuster 17 Jun '14

17 Jun '14

Hi Quick question: If I create different sip aliases under my account, is any way I can see under Missed Called section on serweb.iptel.org or any elsewhere on which alias the call came on? Thank you, Anna On Sat, Jun 7, 2014 at 3:11 PM, <registrar(a)iptel.org> wrote: > > Thank you for registering with iptel.org. > > We are reserving the following SIP address for you: sip:mielul@iptel.org > > To finalize your registration please check the following URL within 24 > hours: > http://serweb.iptel.org/user/reg/confirmation.php?nr=0eb4729fad0f99434d205f… > (If you confirm later you will have to re-register.) > > Here are the details of your account: > --------------------------------------------------------------------- > Username: mielul(a)iptel.org > Password: q1q1q1q1 > Email: annasc74(a)gmail.com > Full Name: Anna M > Sip address: sip:mielul@iptel.org > (You can always retrieve these via the "Forgot Password" link on the > signup page) > > > > -- This message is automatically generated by SerWeb. >

2 1

Re: [SR-Users] exec_avp problem
by Ansuman Manna - Asergis Telecom 17 Jun '14

17 Jun '14

Hi, I am using the exec_avp command as followed: exec_avp("/usr/bin/mwi/test.sh $fU"); $fU taking command line arguments. Here is the shell script content: #!/bin/bash -e echo $1 logger done! $1 logger done mkdir /usr/bin/mwi/$1 || { logger 'touch failed' $? $errorMessage ; exit 1; } logger is working fine but.mkdir and other system commands are not executing. I cant figureout my misses Regards, Ansuman Manna Software Developer Asergis Telecom Services Private Limited Office Tel: +91 11 6606 8400 Office Fax: +91 112554 5156 http://www.asergis.in <http://www.asergis.in>

2 1

How to use unixodbc/freetds for kamailio backend
by Tim Chubb 17 Jun '14

17 Jun '14

Hi Im just getting started with kamailio with a view of evaluating it for production use. Is it possible to do the following: 1) I would like to use ODBC/FreeTDS to connect to an existing ms sql server cluster and use it for all DB related activity relating to kamailio, is this possible as we would be running kamailio in a HA configuration in production and I really don't want to have to support another DB cluster? 2) Looking at the kamdbctl scripts it seems that currently every popular db but sql server is supported which is a bit of a shame, however am I right in saying that if I was to port the sql files to SQL server, and implemented a script similar to kamdbctl.pgsql to wrap around the freetds TSQL executable that would be all that I would need to do add support for SQL server? 3) Is there any particular reason why there isn't an ODBC option for the kamdbctl script? Cheers.

2 1

Kamailio and Freepbx Integration
by Carlos Rangel 16 Jun '14

16 Jun '14

Hello List I am trying to intgrate Kamailio latest release 4.1.4 and Freepbx 5.2.11 Asterisk 11.0.9 in the same box on an internal network following Daniel´s tutorial (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ) . I am very new with Kamailio and I tried like 15 times, seriously. The difference is that I am not using the sipusers table nor sipregs (I may later use sipregs) nor the voicemail tables (Freepbx can manage voicemail). I believe I was able to modify the kamailio.cfg file (DBURL and DBASTURL) in order to fetch the user information from the Freepbx tables. There is a table called devices that contents all the information needed. I also managed to not fill the secret parameter (password) on the extension configuration so Asterisk does not ask for authentication twice as recommended in the tutorial. I put the password in other table and then tell Kamailio to fetch it from the corresponding table. I can control if the users are registered in the system as long as the user is created in Freepbx with the right password, again password is not on the secret column. The weird thing is that my users can register in Asterisk even if they are not created in Kamailio (kamctl add). I was expecting them not to register if they are not created on the database. I have configured Freepbx to bind to a specific internal IP and port 5080 and configures kamailio as in the post #!ifdef WITH_ASTERISK asterisk.bindip = "xxx.xxx.xxx.xxx" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "xxx.xxx.xxx.xxx" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif Also I have added to kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_ANTIFLOOD I dont think this is the normal behavior, does anybody have an insight of what could be happening?? MY goal is to add the #!define WITH_NAT functionality (I havent done it yet)and then have remote trusted users connected to the PBX. Of course the concerns is to keep hackers away from my system. Thanks in advance for your help and for this great piece of software!!!! Regards and greetings from Mexico Carlos

1 0

stampeeding server.
by Måns Nilsson 16 Jun '14

16 Jun '14

Hi, I've got a 3.3.2 Kamailio that from time to time gets very upset and starts rushing these log entries to syslog: Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore Jun 16 10:12:18 landsort /usr/local/kamailio/sbin/kamailio[37542]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x80a4fb370/0x80a4fb380) - ignore The rate is about 10000 lines per second. This fills the log partition, which locks other things up and the users are unhappy. The server recovers nicely if restarted. I have scarce logs because I have to remove logs to get service back. Any suggestions? More data on request. -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 HELLO, everybody, I'm a HUMAN!!

2 3

Kamailio help
by Nouri El Sebai (Student) 16 Jun '14

16 Jun '14

Dear Sir/Madam, I am trying to connect (in Lebanon) as a subscribed user on Kamailio that has been installed on a remote server in the US. I am being faced with the error: Forbidden (403) Reason: Not relaying Kindly advise what I should do. Best regards, Nouri El Sebai

2 1

rtpproxy_manage and ip addresses
by tonchek＠mac.com 13 Jun '14

13 Jun '14

HI to ALL! As I understand the logic of rpproxy_manage work, it uses packet source IP and port number from m= field of SDP as socket for rtpproxy process. So, we can see it in SIP and rtpproxy debug messages: IP 77.94.133.231.5060 > 77.94.133.232.5060: UDP, length 1170 [...headers stripped...] v=0 o=root 1523540081 1523540081 IN IP4 188.40.66.75 s=Asterisk PBX 1.6.1.20 c=IN IP4 188.40.66.75 t=0 0 m=audio 15260 RTP/AVP 8 0 101 [...sdp stripped...] Jun 13 21:38:09 puma rtpproxy[4728]: INFO:handle_command: new session on a port 57680 created, tag FppNrF9K7S8gF;1 Jun 13 21:38:09 puma rtpproxy[4728]: INFO:handle_command: pre-filling caller's address with 77.94.133.231:15260 where 172.22.11.16 is SIP packet source IP address and 17646 is the port number from SDP. Is it possible to change this behavior and use IP address from incoming SDP as well as port number? Thank you Anton

1 1

inbound kamailio dispatcher faulty routing of ack after 200 OK
by Gijs Kwakkel 13 Jun '14

13 Jun '14

client1 <> provider <> kamailio <> asterisk <> client2 when client1 calls client2, everything seems to work. sounds works. after a few seconds the call gets terminated. after tcpdump syslog and config checks i figured out that asterisk sends 200 OK through kamailio to the provider, after this the provider sends a ACK to kamailio, kamailio however sends this ACK to itself and not to asterisk. I added the config, syslog and wireshark output (converted)

2 6

Re: [SR-Users] Is t_on_failure with uac_req_send possible?
by Ian Lin 13 Jun '14

13 Jun '14

Hi Klaus, I've got the same problem on kamailio-4.1.0 and I couldn't see the sip messages reach to onreply route while using uac_req_send. Could you please give me a hint about this? Thanks! Cheers, Ian -- ------------------------ Ian Lin 林于巽 ------------------------

2 2

MSILO add on deliver event route still no joy
by Roberto Fichera 13 Jun '14

13 Jun '14

Hi All, I'm trying to add an event route to be execute for each successfully delivered message from msilo module, but I'm still not getting running the event route I have defined for it in the script. If someone can have a look at the patch and give me some hint would be really great. Thanks in advance. Roberto Fichera.

2 15

Transport protocol related question - how to bind a dialog to a negotiated transport protocol
by Klaus Feichtinger 12 Jun '14

12 Jun '14

Hello, I wonder if it is allowed using transport protocol UDP for SIP NOTIFY requests (which are generated by Kamailio/presence module), when the SUBSCRIBE dialog was established using TCP as transport protocol. In other words: this is a principal question if it is allowed changing the transport protocol for in-dialog transactions e.g. from TCP to UDP. Initially I rather thought that in-dialog transactions shall use the same transport protocol as the dialog itself (e.g. SIP INFO requests within a standard media session), except the message would be too big for UDP, where a change to TCP is recommended. Can anybody give me a hint, if the current behaviour of Kamailio is correct or not? Or - how can I bind Kamailio to a specific transport protocol (for messages that are created by modules themselves)? Kamailio is sending NOTIFY requests with UDP, even when the subscription was done with TCP (see excerpt below). 09:58:10.360749 IP (tos 0x0, ttl 64, id 35302, offset 0, flags [DF], proto TCP (6), length 444) 10.1.1.14.37883 > 10.1.1.44.5060: P, cksum 0x1cb3 (correct), 1:393(392) ack 1 win 457 <nop,nop,timestamp 624699305 795715664> SUBSCRIBE sip:116006@10.1.1.44;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 10.1.1.14:5070;rport;branch=z9hG4bK1540213071 From: <sip:1@10.1.1.14:5070>;tag=620071678 To: <sip:116006@10.1.1.44;transport=TCP> Call-ID: 449986375 CSeq: 20 SUBSCRIBE Contact: <sip:1@10.1.1.14:37883> Max-Forwards: 70 Expires: 1200 Event: presence Content-Length: 0 09:58:10.361324 IP (tos 0x10, ttl 64, id 65438, offset 0, flags [DF], proto TCP (6), length 431) 10.1.1.44.5060 > 10.1.1.14.37883: P, cksum 0x29fb (correct), 1:380(379) ack 393 win 215 <nop,nop,timestamp 795715792 624699305> SIP/2.0 202 OK Via: SIP/2.0/TCP 10.1.1.14:5070;rport=37883;branch=z9hG4bK1540213071 From: <sip:1@10.1.1.14:5070>;tag=620071678 To: <sip:116006@10.1.1.44;transport=TCP>;tag=4f7a7e54f75c89f5b968c90011d693b5-9eed Call-ID: 449986375 CSeq: 20 SUBSCRIBE Expires: 1200 Contact: <sip:10.1.1.44:5060;transport=tcp> Content-Length: 0 09:58:10.361507 IP (tos 0x10, ttl 64, id 32295, offset 0, flags [none], proto UDP (17), length 484) 10.1.1.44.5060 > 10.1.1.14.37883: SIP, length: 456 NOTIFY sip:1@10.1.1.14:37883 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.44;branch=z9hG4bKc408.509e6347000000000000000000000000.0 To: sip:1@10.1.1.14;tag=620071678 From: sip:116006@10.1.1.44;tag=4f7a7e54f75c89f5b968c90011d693b5-9eed CSeq: 2 NOTIFY Call-ID: 449986375 Content-Length: 0 User-Agent: kamailio (4.0.4 (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:10.1.1.44:5060;transport=tcp> Subscription-State: active;expires=1200 Thx Klaus

2 2

Re: [SR-Users] Crash Kamailio 4.1.3
by Igor Potjevlesch 12 Jun '14

12 Jun '14

Hello, Do you any ideas about the reason of this issue? Is it caused by the last call? Do you have enough information with the PAI? Regards, Igor. 2014-05-28 13:47 GMT+02:00 Igor Potjevlesch <igor.potjevlesch(a)gmail.com>: > Hello, > > p *msg->pai : > > $4 = {type = HDR_PAI_T, name = { > s = 0x7fad2cf8af74 "P-Asserted-Identity: > \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: > application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 > 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 > a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 > PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 > telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", > len = 19}, body = { > s = 0x7fad2cf8af89 "\"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: > application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 > 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 > a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 > PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 > telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", > len = 43}, len = 66, > parsed = 0x7fad389a7220, next = 0x7fad2cf8baf0} > > > p *((p_id_body_t*)msg->pai->parsed) : > > $5 = {id = 0x0, num_ids = 0, next = 0x0} > > Regards, > > > > Igor. > > > 2014-05-28 13:29 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > > Hello, >> >> I need now: >> >> p *msg->pai >> p *((p_id_body_t*)msg->pai->parsed) >> >> Cheers, >> Daniel >> >> >> On 28/05/14 12:46, Igor Potjevlesch wrote: >> >> Hello Daniel, >> >> Here is the result of p *msg : >> >> $3 = {id = 128120, pid = 2414, tval = {tv_sec = 1401265832, tv_usec = >> 5951}, fwd_send_flags = {f = 0 '\000', blst_imask = 0 '\000'}, >> rpl_send_flags = {f = 0 '\000', >> blst_imask = 0 '\000'}, first_line = {type = 1, len = 48, u = >> {request = {method = { >> s = 0x7fad2cf8ac30 "INVITE sip:0123456789@domainSIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 6}, uri = { >> s = 0x7fad2cf8ac37 "sip:0123456789@domain SIP/2.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 31}, version = { >> s = 0x7fad2cf8ac57 "SIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 7}, >> method_value = 1}, reply = {version = { >> s = 0x7fad2cf8ac30 "INVITE sip:0123456789@domainSIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 6}, status = { >> s = 0x7fad2cf8ac37 "sip:0123456789@domain SIP/2.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 31}, reason = { >> s = 0x7fad2cf8ac57 "SIP/2.0\r\nVia: SIP/2.0/UDP >> a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: 69\r\nFrom: >> \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 7}, >> statuscode = 1}}}, via1 = 0x7fad2cf8b158, via2 = 0x0, headers = >> 0x7fad2cf8b118, last_header = 0x7fad2cf8bb30, parsed_flag = >> 18446744073709551615, >> h_via1 = 0x7fad2cf8b118, h_via2 = 0x0, callid = 0x7fad2cf8b7a0, to = >> 0x7fad2cf8b548, cseq = 0x7fad2cf8b7e0, from = 0x7fad2cf8b300, contact = >> 0x7fad2cf8b760, >> maxforwards = 0x7fad2cf8b2c0, route = 0x0, record_route = 0x0, >> content_type = 0x7fad2cf8baf0, content_length = 0x7fad2cf8bb30, >> authorization = 0x0, expires = 0x0, >> proxy_auth = 0x7fad2cf8b8d0, supported = 0x7fad2cf8ba70, require = 0x0, >> proxy_require = 0x0, unsupported = 0x0, allow = 0x7fad2cf8ba30, event = >> 0x0, accept = 0x0, >> accept_language = 0x0, organization = 0x0, priority = 0x0, subject = >> 0x0, user_agent = 0x7fad2cf8b850, server = 0x0, content_disposition = 0x0, >> diversion = 0x0, >> rpid = 0x7fad2cf8b890, refer_to = 0x0, session_expires = 0x0, min_se = >> 0x0, sipifmatch = 0x0, subscription_state = 0x0, date = 0x7fad384baa90, >> identity = 0x0, >> identity_info = 0x0, pai = 0x7fad2cf8bab0, ppi = 0x0, path = 0x0, >> privacy = 0x0, body = 0x0, >> eoh = 0x7fad2cf8afea "\r\nv=0\r\no=root 1751857229 1751857230 IN IP4 >> a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 a.b.c.d\r\nt=0 >> 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:3 >> GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> unparsed = 0x7fad2cf8afea "\r\nv=0\r\no=root 1751857229 1751857230 IN >> IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 a.b.c.d\r\nt=0 >> 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:3 >> GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> rcv = {src_ip = {af = 2, len = 4, u = {addrl = {1950469485, >> 140381949486832}, addr32 = {1950469485, 0, 943417072, 32685}, addr16 = >> {52589, >> 29761, 0, 0, 26352, 14395, 32685, 0}, addr = >> "m\315At\000\000\000\000\360f;8\255\177\000"}}, dst_ip = {af = 2, len = 4, >> u = {addrl = {67638457, 0}, >> addr32 = {67638457, 0, 0, 0}, addr16 = {5305, 1032, 0, 0, 0, 0, >> 0, 0}, addr = "\271\024\b\004", '\000' <repeats 11 times>}}, src_port = >> 5060, dst_port = 5060, >> proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = >> 2, sa_data = "\023\304m\315At\000\000\000\000\000\000\000"}, sin = >> {sin_family = 2, >> sin_port = 50195, sin_addr = {s_addr = 1950469485}, sin_zero = >> "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, >> sin6_flowinfo = 1950469485, sin6_addr = {__in6_u = {__u6_addr8 = >> '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, >> __u6_addr32 = {0, 0, 0, >> 0}}}, sin6_scope_id = 0}}, bind_address = 0x7fad383b6588, >> proto = 1 '\001'}, >> buf = 0x7fad2cf8ac30 "INVITE sip:0123456789@domain SIP/2.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nProxy-Authorization: >> Digest username=\"0123456789\", realm=\"domain\", algorithm=MD5, uri=\ >> "sip:0123456789@domain\", nonce=\"U4Wf01OFnqcywgUETmE15dMbFu/Y1hpt\", >> response=\"d055e0f3ba4646773d6ab6c5a05feec7\"\r\nDate: Wed, 28 May 2014 >> 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, >> NOTIFY, INFO\r\nSupported: replaces, timer\r\nP-Asserted-Identity: >> \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 1254, new_uri = { >> s = 0x7fad2ceabdef "sip:0123456789@gw SIP/2.0\r\nRecord-Route: >> <sip:a.b.c.d;lr=on>\r\nVia: SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 INVITE\ >> sendrecv\r\n", len = 1254, new_uri = { >> s = 0x7fad2ceabdef "sip:0123456789@gw SIP/2.0\r\nRecord-Route: >> <sip:a.b.c.d;lr=on>\r\nVia: SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 INVITE\---Type >> <return> to continue, or q <return> to quit--- >> r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nDate: Wed, 28 >> May 2014 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO\r\nSupported: replaces, >> timer\r\nP-Asserted-Identity: \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 19}, dst_uri = {s = 0x0, len = 0}, parsed_uri_ok = 0, parsed_uri = >> {user = { >> s = 0x7fad2ceabdf3 "0123456789@gw SIP/2.0\r\nRecord-Route: >> <sip:a.b.c.d;lr=on>\r\nVia: SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nDate: Wed, 28 >> May 2014 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO\r\nSupported: replaces, >> timer\r\nP-Asserted-Identity: \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 10}, passwd = {s = 0x0, len = 0}, host = { >> s = 0x7fad2ceabdfe "gw SIP/2.0\r\nRecord-Route: <sip:a.b.c.d;lr=on>\r\nVia: >> SIP/2.0/UDP >> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0\r\nVia: >> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5060\r\nMax-Forwards: >> 69\r\nFrom: \"FAX\" <sip:0123456789@domain>;tag=as729b3498\r\nTo: >> <sip:0123456789@domain>\r\nContact: <sip:0123456789@a.b.c.d>\r\nCall-ID: >> 419b254b1c0531af3bf256b14f1881e2@domain\r\nCSeq: 103 >> INVITE\r\nUser-Agent: Asterisk PBX 1.6.2.9-2+squeeze3\r\nRemote-Party-ID: >> \"FAX\" <sip:0123456789@domain>;privacy=off;screen=no\r\nDate: Wed, 28 >> May 2014 08:30:32 GMT\r\nAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, >> SUBSCRIBE, NOTIFY, INFO\r\nSupported: replaces, >> timer\r\nP-Asserted-Identity: \"0123456789\"<sip:0123456789@a.b.c.d>\r\nContent-Type: >> application/sdp\r\nContent-Length: 298\r\n\r\nv=0\r\no=root 1751857229 >> 1751857230 IN IP4 a.b.c.d\r\ns=Asterisk PBX 1.6.2.9-2+squeeze3\r\nc=IN IP4 >> a.b.c.d\r\nt=0 0\r\nm=audio 11886 RTP/AVP 8 3 0 101\r\na=rtpmap:8 >> PCMA/8000\r\na=rtpmap:3 GSM/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >> telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n", >> len = 4}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, >> sip_params = {s = 0x0, len = 0}, headers = { >> s = 0x0, len = 0}, port_no = 0, proto = 0, type = SIP_URI_T, flags >> = 0, transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = >> {s = 0x0, >> len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, >> lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, >> transport_val = { >> s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = >> {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, >> len = 0}, lr_val = { >> s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, >> len = 0}}, parsed_orig_ruri_ok = 0, parsed_orig_ruri = {user = {s = 0x0, >> len = 0}, passwd = { >> s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len >> = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers >> = {s = 0x0, >> len = 0}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 0, >> transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = {s = >> 0x0, len = 0}, >> maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = >> 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, >> transport_val = {s = 0x0, >> len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, >> len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, >> lr_val = {s = 0x0, >> len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = >> 0}}, add_rm = 0x7fad2ce523c8, body_lumps = 0x0, reply_lump = 0x0, >> add_to_branch_s = '\000' <repeats 57 times>, add_to_branch_len = 0, >> hash_index = 26854, msg_flags = 145, flags = 2, set_global_address = {s = >> 0x0, len = 0}, >> set_global_port = {s = 0x0, len = 0}, force_send_socket = 0x0, path_vec >> = {s = 0x0, len = 0}, instance = {s = 0x0, len = 0}, reg_id = 0, ruid = {s >> = 0x0, len = 0}, >> location_ua = {s = 0x0, len = 0}, ldv = {flow = {decoded = 0, rcv = >> {src_ip = {af = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, >> addr16 = {0, 0, 0, 0, 0, >> 0, 0, 0}, addr = '\000' <repeats 15 times>}}, dst_ip = {af >> = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, addr16 = {0, 0, >> 0, 0, 0, 0, 0, 0}, >> addr = '\000' <repeats 15 times>}}, src_port = 0, dst_port = >> 0, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 0, >> sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, >> sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = >> "\000\000\000\000\000\000\000"}, sin6 = { >> sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr >> = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, >> 0, 0, 0, 0, 0, 0}, >> __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, >> bind_address = 0x0, proto = 0 '\000'}}}} >> >> Regards, >> >> >> >> Igor. >> >> >> >> 2014-05-28 12:01 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> >>> Hello, >>> >>> can you give the output in gdb for: >>> >>> set print elements 2000 >>> p *msg >>> >>> Cheers, >>> Daniel >>> >>> >>> On 28/05/14 11:48, Igor Potjevlesch wrote: >>> >>> hello, >>> >>> Kamailio 4.1.3 crashed today, with an segmentation fault error : >>> kernel: kamailio[2406]: segfault at 18 ip 0000000000495792 sp >>> 00007fff3d9d2d80 error 4 in kamailio[400000+275000] >>> >>> Here is the result of BT : >>> #0 0x0000000000495792 in pv_get_strval (msg=0x7fad2cf8a530, >>> param=0x7fad3858c288, res=0x7fff3d9d2eb0, sval=0x18) at pvapi.c:521 >>> #1 0x00007fad365b0da8 in pv_get_pai (msg=0x7fad2cf8a530, >>> param=0x7fad3858c288, res=0x7fff3d9d2eb0) at pv_core.c:1021 >>> #2 0x0000000000499518 in pv_get_spec_value (msg=0x7fad2cf8a530, >>> sp=0x7fad3858c270, value=0x7fff3d9d2eb0) at pvapi.c:1266 >>> #3 0x00007fad34c6d7d9 in extra2strar (extra=0x7fad3858c260, >>> rq=0x7fad2cf8a530, val_arr=0x7fad34e7ae30, >>> int_arr=0x7fad34e7b37c,type_arr=0x7fad34e7b4e7 "\002\002\002\002") at >>> acc_extra.c:261 >>> #4 0x00007fad34c65393 in acc_db_request (rq=0x7fad2cf8a530) at acc.c:474 >>> #5 0x00007fad34c6fb43 in acc_onreply (t=0x7fad2ce0b138, >>> req=0x7fad2cf8a530, reply=0x7fad3859faa8, code=200) at acc_logic.c:471 >>> #6 0x00007fad34c7008d in tmcb_func (t=0x7fad2ce0b138, type=512, >>> ps=0x7fff3d9d3120) at acc_logic.c:559 >>> #7 0x00007fad36e6a478 in run_trans_callbacks_internal >>> (cb_lst=0x7fad2ce0b1a8, type=512, trans=0x7fad2ce0b138, >>> params=0x7fff3d9d3120) at t_hooks.c:290 >>> #8 0x00007fad36e6a68a in run_trans_callbacks_with_buf (type=512, >>> rbuf=0x7fad2ce0b1f8, req=0x7fad2cf8a530, repl=0x7fad3859faa8, flags=183) at >>> t_hooks.c:336 >>> #9 0x00007fad36e9cc06 in relay_reply (t=0x7fad2ce0b138, >>> p_msg=0x7fad3859faa8, branch=0, msg_status=183, cancel_data=0x7fff3d9d3480, >>> do_put_on_wait=1) at t_reply.c:2001 >>> #10 0x00007fad36e9f0b7 in reply_received (p_msg=0x7fad3859faa8) at >>> t_reply.c:2499 >>> #11 0x000000000045d837 in do_forward_reply (msg=0x7fad3859faa8, mode=0) >>> at forward.c:777 >>> #12 0x000000000045e0f8 in forward_reply (msg=0x7fad3859faa8) at >>> forward.c:860 >>> #13 0x00000000004a5887 in receive_msg ( >>> buf=0x924600 "SIP/2.0 183 Session Progress\r\nVia: SIP/2.0/UDP >>> a.b.c.d;branch=z9hG4bK6e86.62fc1f802e2e40a8a17cb1556d5ccfa8.0;received=a.b.c.d\r\nVia: >>> SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK6b726b55;rport=5"..., len=872, >>> rcv_info=0x7fff3d9d3800) at receive.c:273 >>> #14 0x000000000053c838 in udp_rcv_loop () at udp_server.c:536 >>> #14 0x000000000053c838 in udp_rcv_loop () at udp_server.c:536 >>> #15 0x000000000046d42b in main_loop () at main.c:1617 >>> #16 0x00000000004704d3 in main (argc=7, argv=0x7fff3d9d3b38) at >>> main.c:2533 >>> >>> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> >> >

2 19

Kamailio v4.1.4 Released
by Daniel-Constantin Mierla 12 Jun '14

12 Jun '14

Hello, Kamailio SIP Server v4.1.4 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.3. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.4. For more details about version 4.1.4 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/06/kamailio-v4-1-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Problem with MSRP AUTH
by Muhammad Shahzad 12 Jun '14

12 Jun '14

Hi, I am trying to authentication MSRP connection using the example code of msrp event route in module documentation here, http://kamailio.org/docs/modules/4.1.x/modules/msrp.html#idp119248 -- ... } else if ($msrp(method)=="AUTH") { ... if (!pv_www_authenticate("WEBRTC_SIP_REALM", "$var(passwd)", "0", "$msrp(method)")) { if (auth_get_www_authenticate("WEBRTC_SIP_REALM", "0", "$var(wauth)")) { xlog("L_INFO", "Generated www authenticate header for MSRP is [$var(wauth)] \n"); msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Internal Server Error"); }; exit; }; ... -- However i see in logs following error message and authentication fails, -- ERROR: auth [auth_mod.c:690]: pv_www_authenticate2(): failed to get method value -- Which is indicates that value of $msrp(method) is null. However, as you can see in example code in URL provided above we have an IF condition which explicitly checks $msrp(method) == "AUTH". For the sake of testing i even replaced the variable with actual string value, and still get the same error. So i am guessing it is a bug. What do you guys suggest? Kamailio: v4.1.3 (i386/linux) 236326 MSRP Lib: Crocodile MSRP - v1.0.0 Thank you.

2 17

Download SER SIP server
by Horo, Priyam 12 Jun '14

12 Jun '14

Hi, I cannot find the link to download the SER SIP server. I went to http://www.iptel.org/download, but do not see how to download 0.9.6 version. Could you please point me to the link. Thanks, Priyam.

3 2

check for the number of open branches in Reply
by Sebastian Damm 12 Jun '14

12 Jun '14

Hi, I have a scenario where I want to send a custom SIP package whenever a call is successfully established or definitely missed. This works pretty well, except when a user has multiple devices online. I need to send exactly one packet. When initiating the call I set a flag, in the reply route I check for that flag, and on the first 4xx or 2xx response the packet gets sent out. After that I reset the flag, so no other packet is generated for another response coming in. Now when a user has two devices and rejects the call on the first one but accepts it on the second, I see this call as missed, not answered. So I need to use the first 2xx response or the last 4xx response. For that I need to find out how many branches are open at the moment. But I'm missing something. I registered two devices on the same AOR. I tried accessing $branch(count) in the onreply route, but it is always 0. $br and $bR are <null>. Does anyone have an idea, how to get that information in the reply_route? Best Regards, Sebastian

3 5

planning the release of v4.1.4
by Daniel-Constantin Mierla 12 Jun '14

12 Jun '14

Hello, I am considering to release v4.1.4 next week, in one of the days within the interval June 10-12, depending on work that has to be done. If anyone is aware of issue not reported on tracker or patches that have not been backported, add to the tracker or write a message to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 4

Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval
by Alex Villacís Lasso 11 Jun '14

11 Jun '14

I am trying to set up XCAP support for SIP SIMPLE, and using jitsi as the sample client. The kamailio database was previously configured using ODBC and kamailio-4.1.3 and was apparently working correctly. I was following the guide at http://nil.uniza.sk/instant-messaging/simple/configuring-xcap-support-im-an… as well as the standard documentation at http://kamailio.org/docs/modules/4.1.x/modules/xcap_server.html . However, the rls support is currently disabled. When I use jitsi to connect, and add one contact to the empty contact list, I see the database being populated (mysql backend): mysql> select * from xcap; +----+-------------+-----------------+--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+-----------------------+--------+-----------------------------------------------------------------------+------+ | id | username | domain | doc | doc_type | etag | source | doc_uri | port | +----+-------------+-----------------+--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+-----------------------+--------+-----------------------------------------------------------------------+------+ | 3 | avillacisIM | pbx.elastix.com | <?xml version="1.0" encoding="UTF-8" standalone="no"?><resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="RootGroup"><entry uri="sip:bmacias@pbx.elastix.com"><display-name>bmacias</display-name></entry></list></resource-lists> | 4 | sr-1401316328-31413-1 | 0 | /xcap-root/resource-lists/users/sip:avillacisIM@pbx.elastix.com/index | 0 | | 4 | avillacisIM | pbx.elastix.com | <?xml version="1.0" encoding="UTF-8" standalone="no"?><ruleset xmlns="urn:ietf:params:xml:ns:common-policy"><rule id="presence_allow"><conditions><identity><one id="sip:bmacias@pbx.elastix.com"/></identity></conditions><actions><sub-handling xmlns="urn:ietf:params:xml:ns:pres-rules">allow</sub-handling></actions><transformations><provide-devices xmlns="urn:ietf:params:xml:ns:pres-rules"><all-devices/></provide-devices><provide-persons xmlns="urn:ietf:params:xml:ns:pres-rules"><all-persons/></provide-persons><provide-services xmlns="urn:ietf:params:xml:ns:pres-rules"><all-services/></provide-services></transformations></rule></ruleset> | 2 | sr-1401316328-31416-1 | 0 | /xcap-root/pres-rules/users/sip:avillacisIM@pbx.elastix.com/presrules | 0 | +----+-------------+-----------------+--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+-----------------------+--------+-----------------------------------------------------------------------+------+ 2 rows in set (0.00 sec) As far as I can see, both XML documents are well formed. However, if I shutdown and then restart jitsi, I see (with wireshark) that the server returned a 500 Server Error when trying to fetch resource-lists . At the same time, the following appears on the log: May 28 17:39:12 elx /usr/sbin/kamailio[31421]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/xcap-caps/global/index May 28 17:39:12 elx /usr/sbin/kamailio[31421]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/xcap-caps/global/index May 28 17:39:12 elx /usr/sbin/kamailio[31421]: ALERT: <script>: ===== xhttp: xcap-caps : <null> May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/resource-lists/users/sip:avillacisIM@pbx.elastix.com/index May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ALERT: <script>: ===== xhttp: request [HTTP/1.1] GET => /xcap-root/resource-lists/users/sip:avillacisIM@pbx.elastix.com/index May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ALERT: <script>: ===== xhttp: resource-lists : sip:avillacisIM@pbx.elastix.com May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ERROR: xcap_server [xcap_server.c:758]: xcaps_get_db_doc(): invalid xml doc retrieved from database May 28 17:39:12 elx /usr/sbin/kamailio[31413]: ERROR: xcap_server [xcap_server.c:1171]: w_xcaps_get(): could not fetch xcap document What can I do to solve this issue. This is preventing contacts from being retrieved or added once the first contact is stored. My event-route looks like this (WITH_XCAPSRV is defined elsewhere in the configuration): #!ifdef WITH_XHTTP event_route[xhttp:request] { #!ifdef WITH_XHTTP_RPC $var(xhttp_rpc_root) = $(hu{s.substr,0,9}); if ($var(xhttp_rpc_root) == "/http_rpc") { dispatch_xhttp_rpc(); } #!endif #!ifdef WITH_XHTTP_PI $var(xhttp_rpc_root) = $(hu{s.substr,0,8}); if ($var(xhttp_rpc_root) == "/http_pi") { dispatch_xhttp_pi(); } #!endif #!ifdef WITH_XCAPSRV if ($hu =~ "^/xcap-root/") { route(XCAPSRV); } #!endif set_reply_close(); set_reply_no_connect(); # if ($Rp != 80 ##!ifdef WITH_TLS # && $Rp != 443 ##!endif # ) { # # xlog("L_WARN", "HTTP request received on $Rp\n"); # xhttp_reply("403", "Forbidden", "text/html", "Forbidden"); # exit; # } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } #!endif #!ifdef WITH_XCAPSRV route[XCAPSRV] { xlog("L_ALERT", "===== xhttp: request [$rv] $rm => $hu\n"); # Remove port specification from Host header and use as authentication domain $var(http_domain) = $(hdr(Host){s.select,0,:}); if (!www_authorize("$var(http_domain)", "subscriber")) { www_challenge("$var(http_domain)", "0"); exit; } set_reply_close(); set_reply_no_connect(); if ($hu=~"^/xcap-root/oma_status-icon/users/.*/.*$") $var(doc_uri) = $(hu{re.subst,/(^\/xcap-root\/)oma_status-icon(\/users\/.*\/).*$/\1org.openmobilealliance.pres-content\2oma_status-icon\/preseImage/}); else $var(doc_uri) = $hu; # xcap ops $xcapuri(u=>data) = $var(doc_uri); if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:" + $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; xlog("L_ALERT", "===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); if ($xcapuri(u=>auid) == "xcap-caps") { if ($rm == "GET") { $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.pres-content</auid> <auid>org.openmobilealliance.search</auid> <auid>org.openmobilealliance.xcap-directory</auid> </auids> <extensions></extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:prs:pres-content</namespace> <namespace>urn:oma:xml:xdm:search</namespace> <namespace>urn:oma:xml:xdm:xcap-directory</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "OK", "application/xcap-caps+xml", "$var(xbody)"); } else { append_to_reply("Allow: GET\r\n"); xhttp_reply("405", "Method Not Allowed", "", ""); } exit; } # be sure auth user access only its documents if ($au != $(var(uri){uri.user})) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>$si:$sp</body></html>"); exit; } switch($rm) { case "PUT": xcaps_put("$var(uri)", "$var(doc_uri)", "$rb"); if ($xcapuri(u=>auid) =~ "pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); # } else if ($xcapuri(u=>auid) =~ "rls-services" || $xcapuri(u=>auid) =~ "resource-lists") { # rls_update_subs("$var(uri)", "presence"); } else if ($xcapuri(u=>auid) =~ "pidf-manipulation") { pres_refresh_watchers("$var(uri)", "presence", 2, "$xcapuri(u=>uri_adoc)", "$xcapuri(u=>file)"); } exit; break; case "GET": xcaps_get("$var(uri)", "$var(doc_uri)"); exit; break; case "DELETE": xcaps_del("$var(uri)", "$var(doc_uri)"); if ($xcapuri(u=>auid) =~ "pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); # } else if ($xcapuri(u=>auid) =~ "rls-services" || $xcapuri(u=>auid) =~ "resource-lists") { # rls_update_subs("$var(uri)", "presence"); } else if ($xcapuri(u=>auid) =~ "pidf-manipulation") { pres_refresh_watchers("$var(uri)", "presence", 2, "$xcapuri(u=>uri_adoc)", "$xcapuri(u=>file)"); } exit; break; case "POST": if ($xcapuri(u=>auid) =~ "search") { xhttp_reply("501", "Not Implemented", "", ""); } else { if ($xcapuri(u=>auid) =~ "xcap-directory") { append_to_reply("Allow: GET\r\n"); } else { append_to_reply("Allow: DELETE, GET, PUT\r\n"); } xhttp_reply("405", "Method Not Allowed", "", ""); } exit; break; } # other http requests xhttp_reply("404", "Not Found", "", ""); exit; } #!endif

2 11

Re: [SR-Users] Kamailio and Asterisk
by Carlos Rangel 11 Jun '14

11 Jun '14

Found the problem per a previous post. I had uncommented in kamctlrc by mistake. #STANDARD_MODULES="standard acc lcr domain group permissions registrar usrloc msilo # alias_db uri_db speeddial avpops auth_db pdt dialog dispatcher # dialplan" Thanks CR De: Carlos Rangel [mailto:crangel@globaltelesourcing.com] Enviado el: miércoles, 11 de junio de 2014 02:19 p.m. Para: sr-users(a)lists.sip-router.org Asunto: Kamailio and Asterisk Hello Trying to do my first Kamailio and Asterisk integration following the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I am using Centos though. Everything goes well except when trying to create the kamailio DB. The core tables cannot be created. Any ideas how to solve this problem??. Thanks CR [root@gt-ev-i08 kamailio-4.1.3]# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... INFO: creating standard tables into kamailio ... /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql: line 264: /usr/local/share/kamailio//mysql/#-create.sql: No such file or directory ERROR: Creating core tables failed at #!

1 0

Kamailio and Asterisk
by Carlos Rangel 11 Jun '14

11 Jun '14

Hello Trying to do my first Kamailio and Asterisk integration following the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I am using Centos though. Everything goes well except when trying to create the kamailio DB. The core tables cannot be created. Any ideas how to solve this problem??. Thanks CR [root@gt-ev-i08 kamailio-4.1.3]# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... INFO: creating standard tables into kamailio ... /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql: line 264: /usr/local/share/kamailio//mysql/#-create.sql: No such file or directory ERROR: Creating core tables failed at #!

1 0

kamailio and app_python
by Victor Seva 11 Jun '14

11 Jun '14

Hi, is there any way to return values from a call to python_exec function orther than the default returned integer? Following code generates this error: Oct 1 09:08:23 zskio /usr/sbin/kamailio[21789]: ERROR: app_python [python_support.c:38]: python_exec2: Unhandled exception in the Python code: Oct 1 09:08:23 zskio /usr/sbin/kamailio[21789]: ERROR: app_python [python_support.c:78]: #011RuntimeError: no such function if (!python_exec("get_provider", $(var(clean_uri){s.prefixes}))) { xlog("L_ERR","Can not choose a provider for $rU\n"); sl_send_reply("503","Can not choose a provider for $rU"); exit; } def get_provider(self, msg, param): # snipped code... result = self.util.get_provider(prefixes, position, mode) if result is not None: LM_ERR("mode: %s position: %d result: %s" % (mode, position, result)) # this raises error msg.call_function('rewritehost', result['provider']['ip']) return 1 return -1 I tried to use headers as transport but this simple code does not work: Oct 1 09:45:31 zskio /usr/sbin/kamailio[22303]: ERROR: <script>: No header if (!is_present_hf("X-Test-header")) { xlog("L_ERROR", "No header\n"); } else { xlog("L_INFO", "Header added\n"); } def test_set_header(self, msg): msg.call_function('append_hf', 'X-Test-header: test\r\n') return 1 Thanks in advance.

4 6

many-to-one network topology hide with TOPOH
by Maciej Marczyński 11 Jun '14

11 Jun '14

Hi Everyone, recently I'm playing around with the topoh module but I couldn't find the functionality I'm looking for. I'd like to hide my network topology before the end-user but not only the IPs of all VIA and RR but also the amount of hops - so to map 4 VIAs to 1 different VIA etc. The idea is to reduce the size of packets as some of my subscribers suffers from packets fragmentation. Do you think it's possible now or maybe easy to implement? Best Regards, Maciej

2 1

Routing logic and record-routing
by DURECU, OLIVIER (OLIVIER) 11 Jun '14

11 Jun '14

Hello all, There is something I can't understand using Kamailio as proxy to route SUBSCRIBE/NOTIFY dialogs between a client and a server I do record_route() on the first SUBSCRIBE received by Kamailio The next SUBSCRIBE is well handled by Kamailio in the [WITHIN_DIALOG] bloc but the 200/OK no more contains record-route header So the 3rd SUBSCRIBE no more sets a relevant route and Kamailio rejets it with a 404 ? How to proceed so that 200/Ok within dialog still contains Record-route header ? Here are the piece of routing logic involved if (is_method("PUBLISH|SUBSCRIBE")) { record_route(); $ru = "sip:192.168.0.250:6070"; route(RELAY); exit; } And the as-is route[WITHINDLG] function route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(RELAY); } Thanks for your help Olivier

2 1

kamailio + rtpengine: damaged media (Red Hat Linux)
by Alexey Rybalko 11 Jun '14

11 Jun '14

Hello to all! I encountered strange issue with rtpengine: voice during a call is heard like random binary data. (Video freezes during a video call). *It goes fine during 2-3 seconds* before the media flow becomes glitched. It's related to SRTP calls only (RTP-SRTP, SRTP-RTP, SRTP-SRTP). RTP calls through mediaproxy are ok. Used platform: Red Hat Linux 6.4, rtpengine (3.3.0 ?, latest form GitHub), kamailio (4.2.0-dev6) + websocket module. I have successfully installed (manually) mediaproxy daemon, Kamailio module, iptables library and kernel module. Kernel module was loaded (lsmod lists it, there are file in /proc/mediaproxy/0) and iptables were configured as expected. Mediaproxy log looks fine except of a few suspicious messages "No support for kernel packet forwarding available". I'm not sure they are related to the mentioned problem but I would like to have working kernel support. Attached example is for SIP-to-websocket call. rtpengine confguration is: *KERNEL=yes TABLE=0 FALLBACK=no RTP_IP=10.xx.xx.xxLISTEN_NG=127.0.0.1:2223 <http://127.0.0.1:2223> * That would be great if you make any suggestions/comments on this issue. Need to mention that I haven't saw such problems on Debian Linux. Couple of other questions: - What does message "Call-ID to delete not found" means? - "x_tables" is mentioned as required in the documentation for rtpengine kernel module (xt_MEDIAPROXY). Seems to be for Debian only. Is it required for other distros? with best regards, Alexey

2 4

Location issue
by Igor Potjevlesch 11 Jun '14

11 Jun '14

Hello, I'm running Kamailio 4.1.3 and having a strange issue. After a restart of Kamailio, some username are still in memory and updated accordingly with usual re-REGISTER flow. But the entries in location db is deleted. Kamailio tries to do a SQL UPDATE which fails because of that. So, for each username which are not in location DB, I must do a "kamctl ul rm" and wait for the next REGISTER. The new REGISTER will challenge the UAC for a new Authorization which should result in an INSERT INTO instead of an UPDATE. Can someone help me to solve this? Regards, Igor.

2 6

Kamailio and Freepbx Integration
by Carlos Rangel 10 Jun '14

10 Jun '14

Hello I am working on a project which looks for integration between Kamailio with Freepbx. We want to have remote sip users connected to the Freepbx server which is behind a NAT Firewall. The remote users are dynamic so IP addresses are always changing so we want Kamailio to authenticate this users and then fREEPBX to handle the call flow. I just started and have done a few trials without success based on the Kamailio-Asterisk integration tutorials on the documentation page. From other post I have read I know the following: - Kamailio and Asterisk can be on the same or different servers. I would look for a integrating with different servers. - Freepbx does not use realtime, instead uses the .conf files however the user information is contained on the database tables, I just need to select the correct table and configure Kamailio to get the user information from there. - Since the system will work behind a Nat firewall I assume that need to include the -- #!define WITH_NAT--- functionality on the kamailio.cfg. - Freepbx or Asterisk should not ask for authentication from Kamailio--- although I am not sure how exactly to accomplish this. I am writing this post just to see if anyone has had a previous experience to share or someone can give me an insight about the tricks on this. I will be working on this on the next days and share my experiences. Thanks a lot for your help. Carlos Rangel

1 0

Crash Kamailio 4.1 (qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten)
by Andrés Souto 10 Jun '14

10 Jun '14

Hello, Kamailio crashes with this message: May 29 12:03:45 kamfree-manager kamailio[8705]: : <core> [mem/q_malloc.c:159]: qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten(c0c0c000, abcdefed)[0x7ff5a5031a98:0x7ff5a5031ac8]! May 29 12:03:45 kamfree-manager kamailio[8697]: ALERT: <core> [main.c:775]: handle_sigs(): child process 8705 exited by a signal 6 May 29 12:03:45 kamfree-manager kamailio[8697]: ALERT: <core> [main.c:778]: handle_sigs(): core was generated The problem is produced in this line of the script: $rU = $dbr(ra=>[0,0]); I'm using last commit in 4.1 branch (ad5235229f59ba5d1f4216e144291e110fee0211) on Debian wheezy. The backtrace: #0 0x00007ff5a538b475 in raise () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007ff5a538e6f0 in abort () from /lib/x86_64-linux-gnu/libc.so.6 #2 0x0000000000546a00 in qm_debug_frag (qm=0x7ff5a4f58010, f=0x7ff5a5031a98) at mem/q_malloc.c:161 #3 0x000000000054796d in qm_malloc (qm=0x7ff5a4f58010, size=1024, file=0x5c9ec1 "<core>: action.c", func=0x5cb1d8 "do_action", line=832) at mem/q_malloc.c:386 #4 0x000000000041affa in do_action (h=0x7fff810cd2b0, a=0x7fff810cd390, msg=0x7ff5a502ed28) at action.c:832 #5 0x00007ff5a2295d88 in pv_set_ruri_user (msg=0x7ff5a502ed28, param=0x7ff5a4fb3750, op=254, val=0x7fff810cd4e0) at pv_core.c:2073 #6 0x0000000000467bbf in lval_pvar_assign (h=0x7fff810d0200, msg=0x7ff5a502ed28, lv=0x7ff5a501ea00, rv=0x7ff5a501f110) at lvalue.c:362 #7 0x00000000004680a5 in lval_assign (h=0x7fff810d0200, msg=0x7ff5a502ed28, lv=0x7ff5a501ea00, rve=0x7ff5a501f108) at lvalue.c:410 #8 0x0000000000423fec in do_action (h=0x7fff810d0200, a=0x7ff5a501eb68, msg=0x7ff5a502ed28) at action.c:1478 #9 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a5013ee0, msg=0x7ff5a502ed28) at action.c:1599 #10 0x000000000041c18d in do_action (h=0x7fff810d0200, a=0x7ff5a5020a10, msg=0x7ff5a502ed28) at action.c:1090 #11 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a5020a10, msg=0x7ff5a502ed28) at action.c:1599 #12 0x000000000041c18d in do_action (h=0x7fff810d0200, a=0x7ff5a5020e50, msg=0x7ff5a502ed28) at action.c:1090 #13 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a50119b0, msg=0x7ff5a502ed28) at action.c:1599 #14 0x000000000041a2b7 in do_action (h=0x7fff810d0200, a=0x7ff5a4fd3fc0, msg=0x7ff5a502ed28) at action.c:715 #15 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a4fd2800, msg=0x7ff5a502ed28) at action.c:1599 #16 0x0000000000422c70 in do_action (h=0x7fff810d0200, a=0x7ff5a4fd7af0, msg=0x7ff5a502ed28) at action.c:1235 #17 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a4fd1ed8, msg=0x7ff5a502ed28) at action.c:1599 #18 0x000000000041c18d in do_action (h=0x7fff810d0200, a=0x7ff5a4fd7d10, msg=0x7ff5a502ed28) at action.c:1090 #19 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a4fcf1d8, msg=0x7ff5a502ed28) at action.c:1599 #20 0x000000000041a2b7 in do_action (h=0x7fff810d0200, a=0x7ff5a4fb7d88, msg=0x7ff5a502ed28) at action.c:715 #21 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a4fb6b68, msg=0x7ff5a502ed28) at action.c:1599 #22 0x000000000041c18d in do_action (h=0x7fff810d0200, a=0x7ff5a4fb9838, msg=0x7ff5a502ed28) at action.c:1090 #23 0x0000000000424d70 in run_actions (h=0x7fff810d0200, a=0x7ff5a4faf5a0, msg=0x7ff5a502ed28) at action.c:1599 #24 0x0000000000425524 in run_top_route (a=0x7ff5a4faf5a0, msg=0x7ff5a502ed28, c=0x0) at action.c:1685 #25 0x00000000004a6fdb in receive_msg ( buf=0x921920 "INVITE sip:2000@vpbx2.ur SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.101;rport;branch=z9hG4bK14cFrXZ0N53jS\r\nMax-Forwards: 67\r\nFrom: \"1000\" <sip:1000@vpbx1.ur>;tag=FjmFN79ZrXH0e\r\nTo: <sip:2000@vpbx2.ur>\r\nCall-"..., len=1061, rcv_info=0x7fff810d04b0) at receive.c:212 #26 0x000000000053c544 in udp_rcv_loop () at udp_server.c:536 ---Type <return> to continue, or q <return> to quit--- #27 0x000000000046ee54 in main_loop () at main.c:1617 #28 0x0000000000471df4 in main (argc=3, argv=0x7fff810d07e8) at main.c:2533 Complete log: http://pastebin.com/YZirUkfe Thanks. Regards, -- *Andrés Souto* VoIP Engineer @ Quobis <http://www.quobis.com/> | e: andres.souto(a)quobis.com | t: +34902999465

2 8

re-use Expires from 200 OK
by Daniel Ciprus 10 Jun '14

10 Jun '14

Is there any way to extract Expires from 200 OK (response to REGISTER) ? I will need to re-use this value in 3rd party registration being sent to application server. Thanks for any hints .. -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 1

Iptel.org cannot make call
by Tommy Yan 09 Jun '14

09 Jun '14

Dear Sir/Madam, We setup the Linksys SPA3000 to make a call via IPTEL.ORG in between my family members around the world. But recently, we found out that we cannot dial out. So, would you please help to check this ? Regards, Tommy.

2 1

kamailio dispatcher failover setup
by bajeesh＠itplus.in 09 Jun '14

09 Jun '14

Hi, I am first in kamailio,I have installed the kamailio from source code and added a user using kamctl,Now I can connect with that user from soft phone. My task is making a Load balancer and fail over for asterisk servers, I have some other asterisk servers, I need make the kamailio as a Load balancer, How can I do that? 1. I need to forward all calls to asterisks servers. 2. I have created a Dispatcher list in db, how can I configure that? Please help me on this. Hopefully, Bajeesh T.B

2 1

Kamailio on PPC
by ashwin 09 Jun '14

09 Jun '14

Hi, I tried to compile Kamailio on Freescale P2020RDB-PCA.And I'm getting the following error bison -d -b cfg cfg.y make: *** [cfg.tab.h] Broken Pipe When i run that command without running 'make'. Just 'bison -d -b cfg cfg.y'.It was working.So found from Internet that it could be the problem with memory when the bison process is running.So I allocated a swap file for the build process. dd if=/dev/zero of=swapfile.1 bs=1048576 count=100 ; mkswap ./swapfile.1 ; swapon ./swapfile.1 But after that attempt also I'm still getting the make: *** [cfg.tab.h] Broken Pipe when Im doing the compilation. Yours recommendations will be very helpful. Thanks Ashwin -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-on-PPC-tp128097.html Sent from the Users mailing list archive at Nabble.com.

2 1

How connect kamailio to remote redis server ?
by yann christophe 09 Jun '14

09 Jun '14

Hello, I have two machines , kamailio 1.1.1.1 and redis 1.1.1.2 this is my configuration in the kamailio.cfg for redis: modparam("ndb_redis", "server", "name=local;addr=1.1.1.2;port= 6379") but it's correct ? because i don't see my kamailio in my client list redis thanks for your help Yann

2 1

how to call the user of other sip proxy server
by 启程 09 Jun '14

09 Jun '14

Hi,I installed kamailio . The user of other sip proxy server can call me ,but I can not call them.I　want to call them ,how to config it ?

2 1

Branch feedback behaviour
by Alex Balashov 08 Jun '14

08 Jun '14

Hello, I've got a situation like this: UAC ---> (Kamailio) ----------> 1. redirect server on branch 1 <---------- 2. 302 redirect ----------> 3. ACK ----------------------------> UAC B Problem is, if CANCEL comes in at the exact moment that the INVITE to the redirect server (#1) is "in flight", i.e. has not returned a 302 yet, then the transaction gets cancelled and 302 is fed back to the calling UAC. Having the 302 fed back to the calling UAC in any circumstances is quite undesirable. I just need to drop it. Problem is, in all other situations, I need to use it--I consume it in a failure_route and create new branches based on it. What's the best way to accomplish this? I'm concerned that if I use a global onreply_route to filter all 302s going back to callers, that will prevent it from percolating down to the failure_route. Suggestions appreciated, and thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 3

SDP and NAT
by Marc Soda 06 Jun '14

06 Jun '14

Hey all, I have a Kamailio box functioning as a proxy. When some UAs send an INVITE the SDP has their private IP (c=IN IP4 10.x.x.x). Kamailio is passing this on without changing it to the proper public IP. What's the best way to rewrite that? I'm using a config based on the default one.

2 1

Webrtc: Don't catch 488 between JSSIP and SIP UA's
by LAA 06 Jun '14

06 Jun '14

Apologize. Previous message was too long. L. El 02/06/2014 20:25, "LAA" <ornitorrinco7424(a)gmail.com> escribió: > Hi all, > > Another guy strugling his mind trying to get a configuration to enable > calls between WebRTC UA (JSSIP) to standard SIP UA (Twinkle or SjPhone) > I've been working with the examples that were shared by Carlos Ruiz Diaz > and Peter Dunkley (thanks to both). > > http://www.slideshare.net/crocodilertc/webrtc-websockets > http://caruizdiaz.com/2014/02/26/webrtc-kamailio/ > > Kamailio is not running behind NAT. I'm using rtpproxy-ng module with > Kamailio 4.1.3, and Rtpengine. > > I share a link with my current configuration, wich is based in Peters > example, with websocket support from websocket.cfg example. > > - Calls between SIP standard UA's are working OK. I have some endpoint > behind nat. > - Calls between JSIP UA's are working OK. So, websocket support is > running. > - Calls from JSIP and Twinkle are NOT WORKING OK. sip UA send's back a > 488 response, and Kamailio send it back to JSSIP (Incompatible SDP). > - Calls from Twinkle to Jsip are NOT WORKING OK: Kamailio sends an INVITE > to JSIP, and it returns an error. And Kamailio sends 488 to Twinkle. > > > It seems as if Kamailio is not catching 488. I share a snippet of my > config, and links to tcpdump captures: > > https://www.dropbox.com/s/i7c9ty57oauujc4/fromws0.pcap > https://www.dropbox.com/s/q3q30pgzvdoswts/kamailio.cfg > https://www.dropbox.com/s/rqtjwcbgg1foaoq/tows0.pcap > > What am I missing? > > > Best regards. > > Luis. > > > > > > > >

1 1

HOW CONNECT KAMAILIO TO REMOTE REDIS SERVER
by yann christophe 06 Jun '14

06 Jun '14

Hello, I have two machines , kamailio 1.1.1.1 and redis 1.1.1.2 this is my configuration in the kamailio.cfg for redis: modparam("ndb_redis", "server", "name=local;addr=1.1.1.2;port=6379") but it's correct ? because i don't see my kamailio in my client list redis thanks for your help Yann

1 0

CANCEL requests
by Andrea Meroni 06 Jun '14

06 Jun '14

Hi all, just a simple question: which is the timer that triggers if no reply is received for the current transaction and makes Kamailio send a CANCEL request? Andrea

2 2

many-to-one network topology hide with TOPOH
by Maciej Marczyński 06 Jun '14

06 Jun '14

Hi Everyone, recently I'm playing around with the topoh module but I couldn't find the functionality I'm looking for. I'd like to hide my network topology before the end-user but not only the IPs of all VIA and RR but also the amount of hops - so to map 4 VIAs to 1 different VIA etc. The idea is to reduce the size of packets as some of my subscribers suffers from packets fragmentation. Do you think it's possible now or maybe easy to implement? Best Regards, Maciej

2 1

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