- sr-users - kamailio.org

Kamailio consumes memory
by Martin Tichatschek 15 Mar '12

15 Mar '12

Hi everyone, I have a problem with Kamailio consuming memory. I started testing with Kamailio 3.2.1 upgraded to 3.2.2 and also tested 3.1.5, everywhere the same problem. Whereas Kamailio 1.5.0 is working fine. Kamailio -V shows the following: kamailio@test:~$ kamailio -V version: kamailio 3.2.2 (i386/linux) 98ba92 flags: STATS: Off, USE_IPV6, 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: 98ba92 compiled on 17:51:30 Feb 29 2012 with gcc 4.2.4 kamailio@test:~$ My test scenario consists of two SIPp servers (a UAC and a UAS) and the Kamailio in the middle proxying the SIP traffic. The UAC sends 300 calls per second (INVITE, 100, 180, 200, ACK) and releases the calls (BYE, 200) after 20 sec call duration. UAS is simply sending back the right responds. Protocol is UDP. After some minutes the Kamailio floods the syslog file with error messages indicating that he is running out of memory. The SIPp shows retransmissions and also timeouts. When I stop the test and do a single call I get some lines of error messages. I found three different sets of error messages for a single call so far: Mar 2 09:35:44 test P-CSCFtest[21251]: ERROR:<core> [parser/msg_parser.c:172]: get_hdr_field: out of memory Mar 2 09:35:44 test P-CSCFtest[21251]: INFO:<core> [parser/msg_parser.c:353]: ERROR: bad header field [To:<sip:00498111111] Mar 2 09:35:44 test P-CSCFtest[21251]: ERROR: tm [t_lookup.c:1118]: ERROR: reply cannot be parsed Mar 2 09:35:44 test P-CSCFtest[21251]: ERROR:<core> [msg_translator.c:1884]: ERROR: build_res_buf_from_sip_res: out of mem Mar 2 09:35:44 test P-CSCFtest[21251]: ERROR:<core> [forward.c:794]: ERROR: forward_reply: building failed Mar 2 09:35:44 test P-CSCFtest[21228]: ERROR:<core> [msg_translator.c:1884]: ERROR: build_res_buf_from_sip_res: out of mem Mar 2 09:35:44 test P-CSCFtest[21228]: ERROR: tm [t_reply.c:1786]: ERROR: relay_reply: no mem for outbound reply buffer Mar 2 09:35:44 test P-CSCFtest[21228]: ERROR:<core> [msg_translator.c:2036]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 311 Mar 2 10:19:43 test P-CSCFtest[21346]: ERROR:<core> [msg_translator.c:1884]: ERROR: build_res_buf_from_sip_res: out of mem Mar 2 10:19:43 test P-CSCFtest[21346]: ERROR: tm [t_reply.c:1786]: ERROR: relay_reply: no mem for outbound reply buffer Mar 2 10:19:43 test P-CSCFtest[21346]: ERROR:<core> [msg_translator.c:2036]: ERROR: build_res_buf_from_sip_req: out of memory ; needs 311 Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR:<core> [action.c:852]: ERROR: do_action: memory allocation failure Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR:<core> [action.c:1568]: run action error at: /usr/local/etc/kamailio//kamailio.cfg:28 Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR:<core> [action.c:852]: ERROR: do_action: memory allocation failure Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR:<core> [action.c:1568]: run action error at: /usr/local/etc/kamailio//kamailio.cfg:32 Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR:<core> [parser/parse_from.c:71]: ERROR:parse_from_header: out of pkg_memory Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR: tm [t_lookup.c:1326]: ERROR: new_t: no valid From in INVITE Mar 2 10:20:16 test P-CSCFtest[21328]: ERROR: tm [t_lookup.c:1478]: ERROR: t_newtran: new_t failed Mar 2 10:20:16 test P-CSCFtest[21328]: WARNING:<core> [receive.c:209]: WARNING: receive_msg: error while trying script I repeated the test various times, it always generates a memory problem. The "top" command shows that the free memory is decreasing constantly during the test. Stopping the test and waiting for some time (1 hour) doesn't increase the free memory showed by "top". Only a restart of the Kamailio frees the memory again. I executed the test with 30 calls per second too and got the same behaviour (after some more minutes). I finally simplified the Kamailio config as far as possible and came to this minimum: #!KAMAILIO listen=10.10.10.10:5060 children=30 user="kamailio" group="kamailio" debug=2 disable_core_dump=yes log_facility=LOG_LOCAL0 log_name="P-CSCFtest" server_signature=no sip_warning=0 syn_branch=0 disable_tcp=yes ####### Modules Section ####### mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" loadmodule "tm.so" ####### Main Routing Logic ######## route{ if (src_ip==1.2.3.4) { rewritehost("5.6.7.8"); } else { rewritehost("1.2.3.4"); }; rewriteport("5061"); t_relay(); } This config still generates the memory problem. After replacing the "t_relay()" with the stateless "forward()" the problem is gone. I played around with the number of child processes too, but no difference. Is this a memory leak of the TM module or is there a different explanation? Regards Fred

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Timer
by Ravi DHIRAJLAL 15 Mar '12

15 Mar '12

Hello we installed kamailio succesfully but we have a few problems regarding calls towards special numbers in which an INVITE is not replied with 200 OK but instead with 183 session progress sent at 1 min interval, during which the caller hears a waiting message. unless a human operator answer the call, no 200 OK is replied. if that waiting time is too long, we noticed that the call is hanged up after 3 min, 180s, that is 3 * session progress . 408 timeout is sent to the caller and cancel to the callee. we tried to adjust many timers listed in the tm module, but not seems to work we'd like to see if anybody has a clue about that ? thank you very much

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How to connect two legs of a call
by Chester Wood 15 Mar '12

15 Mar '12

Hello, SIP newbie here. We are a new WISP offering voip to our residential customers. Due to IP address scarcity we have to use large scale NAT in our network. In order for our local customers to call each other and have good call quality we wish to connect their ATA's directly to each other rather than have the RTP proxied 6000 miles across the public internet. I'm wondering about the best way to program an outbound proxy to monitor the calls and bypass the ITSP's B2BUA for local calls. We want to rely on the ITSP's server for authentication (so we would only REGISTER our UA's locally after we get an OK from the remote server. Or perhaps we would not need a local registrar at all.) The remote server is also where the users manage their account. So we only want to attempt to connect the 2 UA's directly after the callee has sent an "OK" response to the INVITE from the ITSP. (The callee may have set "Do not disturb" on his account at the ITSP, and it needs to handle voicemail, forwarding etc.) We have other constraints. The server hardware needs to be very low power to be installed at our solar-powered head end. Thus for potentially several hundred clients it needs to handle only SIP, no media. We could potentially install a media relay server elsewhere, but hopefully we won't need to do any media proxying at all. The NAT assignment is all handled by a Mikrotik router at our head end. I've looked at the milkfish configuration file and it seems like in some ways, ours should be simpler. Milkfish assumes it is running on the gateway, so when it sets up an rtpproxy it doesn't cost anything. For us, it doesn't seem like we should have to do any NAT fixup or any media proxying. If the connection is local, the 2 UA's should connect directly with each other. Otherwise, the connection will be directly between our UA and whatever the ITSP sets up as the other end, and the ITSP already knows how to traverse our NAT very well. One idea I had is that, for local calls, when the initial INVITE is forwarded from the caller to the ITSP, the caller's original SDP payload is stashed away somewhere. Then when the INVITE comes through on the way to the callee, replace the ITSP's SDP info with the stashed original. Then when the callee responds with an "OK", I forward the response directly to the caller instead of the ITSP. There would have to be some fixup of loose ends, also. How to let the ITSP that its proxy services are no longer needed? (It could have decided to forward the call to voicemail in the meantime, etc.) Can you query the SER transaction database by Call-ID to find the other side of the call and extract information from it? Perhaps I'm barking up the wrong tree, but I'm sure there's a correct way to do this. Any help would be appreciated. thanks.

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Re: [SR-Users] Enable session_id in ServerHello for TLS
by Daniel-Constantin Mierla 14 Mar '12

14 Mar '12

Hello, thanks for reporting back it's working -- please keep the mailing list cc-ed, so people looking for same issue will be able to find it when searching the web archive. I am using snom3xx with tls and kamailio 3.x a lot, never had issues, but I have no clue about the softphone.exe Cheers, Daniel On 3/11/12 8:09 PM, Kristijan Vrban wrote: > Hello Daniel, > > many thanks for the fast reply, And yes, the session_cache option > solved my problem. Well... the device i used was the immemorial > snom360 softphone.exe > running with wine :) The softphone i use since years for TLS testing. > > Kristijan > > 2012/3/11 Daniel-Constantin Mierla<miconda(a)gmail.com>: >> Hello, >> >> >> On 3/11/12 1:28 AM, Kristijan Vrban wrote: >>> Hello, how to tell that Kamailio should juse a session_id for tls ? >>> See ssldump output below. I reckon that this is the reason the >>> client i use end with "handshake_failure". Because when is use >>> opensips, there is the session_id, and it's working. >>> >>> Kristijan >>> >>> 2 1 0.0228 (0.0228) C>S Handshake >>> ClientHello >>> Version 3.1 >>> cipher suites >>> TLS_RSA_WITH_RC4_128_MD5 >>> TLS_RSA_WITH_RC4_128_SHA >>> TLS_RSA_WITH_NULL_MD5 >>> TLS_RSA_WITH_NULL_SHA >>> TLS_DH_anon_WITH_3DES_EDE_CBC_SHA >>> TLS_DH_anon_WITH_RC4_128_MD5 >>> TLS_RSA_WITH_DES_CBC_SHA >>> TLS_RSA_EXPORT1024_WITH_RC4_56_SHA >>> TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA >>> TLS_DH_anon_WITH_DES_CBC_SHA >>> compression methods >>> NULL >>> 1 0.0519 (0.0519) C>S TCP FIN >>> 2 2 0.0432 (0.0204) S>C Handshake >>> ServerHello >>> Version 3.1 >>> session_id[0]= >>> >>> cipherSuite TLS_RSA_WITH_RC4_128_MD5 >>> compressionMethod NULL >>> 2 3 0.0432 (0.0000) S>C Handshake >>> Certificate >>> 2 4 0.0432 (0.0000) S>C Handshake >>> ServerHelloDone >>> 2 5 0.0452 (0.0020) C>S Alert >>> level fatal >>> value handshake_failure >>> 1 0.0744 (0.0225) S>C TCP FIN >>> 2 0.0681 (0.0228) S>C TCP FIN >> the tls module has now the option to turn on/off session caching, which was >> on by default in openser 1.x. Now it is off as it does not make much >> benefits with out multi-process architecture. Try to add to your config: >> >> modparam("tls", "session_cache", 1) >> >> Let me know if works -- the module parameter is missing from the readme, >> perhaps the author forgot to add it at the time of development -- I will try >> to sync the sources and the readme for tls module asap. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany >> http://www.asipto.com/index.php/kamailio-advanced-training/ >> -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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OT: is it allowed to send requests without waiting for the response of the previous request?
by Klaus Darilion 14 Mar '12

14 Mar '12

Hi all! I wonder if a presence server may send a NOTIFY if the previous NOTIFY in the dialog did not received an answer yet. I "grep"ed the relevant RFCs but couldn't find a definition. Thus, pointers are appreciated. Thanks Klaus

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kamailio issues??? or ekiga problem??
by Vineet Menon 14 Mar '12

14 Mar '12

Hi, I just installed kamailio for my sip testing. I am using it for simple registrar purposes... The config file is default file with debugging enabled. I registered 2 UAs with kamailio, but am not able to call using aor dialing. I inspected packets with wireshark, which i think is a bit problematic. my server's at 31.21.33.238, while UAs are at 31.21.33.250 and 31.21.33.236. I have attached the wireshark file for packets. see packet number.5 and 7. Why has ekiga put the from field as ip:0.0.0.0.??? Isn't der something wrong here??? Regards, Vineet Menon

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rfc: unique id for profiles
by Daniel-Constantin Mierla 13 Mar '12

13 Mar '12

Hello, from the last IRC devel meetings discussions, one topic was about adding an unique id field per user profiles, with the decision to bring it to mailing lists for broader audience. That will help to reference easier across records. At this moment, the most relationships are via (username) or (username,domain) when use_domain=1, bringing some complexity. The main reasons I came up to this were: - in the perspective of gruu implementation for usrloc, with global addresses, an unique (opaque) id seem to simplify things, no matter SIP address of the user changes or not -- the public gruu id can be different than the internal one, though - it will open a new group of K and S modules for merging, S using unique ids mostly everywhere Moving towards it will be done gradually, keeping backward compatibility for a while. The first steps will be to add new column in few tables (subscriber, location), the username/domain columns will stay there forever in most of the cases, they will become additional info over the time (e.g., primary sip address in subscriber). From the point of view of filling up the column, can be as simple as setting it to username@domain, or generate some unique string with tools out there. Overall, there will be a bit more db space used at the benefit of reduced complexity in the code. Jason Penton mentioned that IMS architecture relies also on unique ids. Looking forward to comments, alternatives, etc ... sr-users community was cc-ed because it is not strictly technical discussion/decision. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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Bad AVP Flags When Using avp_db_load()
by Simpson Chua 13 Mar '12

13 Mar '12

Hi Daniel, Thanks! I changed to use the $avp() syntax and that fixed it. The examples I followed was from an old wiki. Simpson ________________________________ From: "sr-users-request(a)lists.sip-router.org" <sr-users-request(a)lists.sip-router.org> To: sr-users(a)lists.sip-router.org Sent: Tuesday, March 13, 2012 12:45 PM Subject: sr-users Digest, Vol 82, Issue 41 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. log not visible in stderr (Vineet Menon) 2. Re: Compilation errors... (Krishna Kurapati) 3. Bad AVP Flags When Using avp_db_load() (Simpson Chua) 4. Re: log not visible in stderr (Daniel-Constantin Mierla) 5. Re: Bad AVP Flags When Using avp_db_load() (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Tue, 13 Mar 2012 16:39:27 +0530 From: Vineet Menon <mvineetmenon(a)gmail.com> Subject: [SR-Users] log not visible in stderr To: sr-users(a)lists.sip-router.org Message-ID: <CAD26Kg268dDfFK=UE4BSbyvSee33nvcybMD3ESEpNYtNXTnCeA(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, I am new to kamailio and want to install a simple registrar. Now, I have a ubuntu machine in which I have installed kamailio 3.2.x. I have included #!define WITH_DEBUG in configuration file, while the rest of the file is same as default. But I am not able to see any debug message. Any help? Regards, Vineet Menon

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Compilation errors...
by Krishna Kurapati 13 Mar '12

13 Mar '12

Hi, git cloned main branch and tried to compile it on Cent OS 6. The mi_xmlrpc module is failing to compile make Makefile:46: You are using an unsupported libxmlrpc-c3 version (1.16.24), compile at your own risk! Makefile:46: You are using an unsupported libxmlrpc-c3 version (1.16.24), compile at your own risk! CC (gcc) [M mi_xmlrpc.so] abyss_conf.o CC (gcc) [M mi_xmlrpc.so] abyss_conn.o CC (gcc) [M mi_xmlrpc.so] abyss_data.o abyss_data.c:219: error: expected ‘)’ before ‘*’ token abyss_data.c:236: error: expected ‘)’ before ‘*’ token abyss_data.c:249: error: expected ‘)’ before ‘*’ token abyss_data.c:287: error: expected ‘)’ before ‘*’ token abyss_data.c:299: error: expected ‘)’ before ‘*’ token abyss_data.c:312: error: expected ‘)’ before ‘*’ token abyss_data.c:326: error: expected ‘)’ before ‘*’ token abyss_data.c:332: error: expected ‘)’ before ‘*’ token make: *** [abyss_data.o] Error 1 I have installed following rpms: xmlrpc-c-devel-1.16.24-1200.1840.el6_1.4.x86_64 xmlrpc-c-client-1.16.24-1200.1840.el6_1.4.x86_64 xmlrpc-c-client++-1.16.24-1200.1840.el6_1.4.x86_64 xmlrpc-c-c++-1.16.24-1200.1840.el6_1.4.x86_64 xmlrpc-c-1.16.24-1200.1840.el6_1.4.x86_64 Can someone help with this? Thanks Krish

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Bad AVP Flags When Using avp_db_load()
by Simpson Chua 13 Mar '12

13 Mar '12

Hi, I'm trying to use avp_db_load to check for some IPs but I'm getting bad avp flags errors. As far as I understand, the syntax is correct. TableName: labservers uuid username domain attribute value type "" "" "domain1" "lab1" "192.168.55.1" 0 "" "" "domain1" "lab1" "192.168.55.2" 0 "" "" "domain2" "lab2" "192.168.55.3" 0 "" "" "domain2" "lab2" "192.168.55.4" 0 if ((avp_db_load("$ruri/domain","s:lab1/labservers") && avp_check("s:lab1", "eq/$Ri/gi"))) { # Do something } Mar 13 10:55:24 ubuntu /usr/local/sbin/kamailio[12012]: DEBUG: <core> [rvalue.c:2857]: RV fixing type 3 Mar 13 10:55:24 ubuntu /usr/local/sbin/kamailio[12012]: DEBUG: <core> [rvalue.c:2857]: RV fixing type 2 Mar 13 10:55:24 ubuntu /usr/local/sbin/kamailio[12012]: DEBUG: <core> [rvalue.c:2865]: RV is str: "REGISTER" Mar 13 10:55:24 ubuntu /usr/local/sbin/kamailio[12012]: DEBUG: <core> [route.c:506]: right EXP optimized: op10(O51, _ST22_) => op10(O51, _ST1_) Mar 13 10:55:24 ubuntu /usr/local/sbin/kamailio[12012]: DEBUG: <core> [rvalue.c:2857]: RV fixing type 4 Mar 13 10:55:24 ubuntu /usr/local/sbin/kamailio[12012]: DEBUG: <core> [route.c:929]: fixing avp_db_load() Mar 13 10:55:24 ubuntu /usr/local/sbin/kamailio[12012]: ERROR: avpops [avpops_parse.c:129]: error - bad avp flags Mar 13 10:55:24 ubuntu /usr/local/sbin/kamailio[12012]: ERROR: avpops [avpops.c:299]: parse failed Mar 13 10:55:24 ubuntu /usr/local/sbin/kamailio[12012]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:71 Line 71 is the "if" statement above. Could someone see could be the issue? Thanks, Simpson

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log not visible in stderr
by Vineet Menon 13 Mar '12

13 Mar '12

Hi, I am new to kamailio and want to install a simple registrar. Now, I have a ubuntu machine in which I have installed kamailio 3.2.x. I have included #!define WITH_DEBUG in configuration file, while the rest of the file is same as default. But I am not able to see any debug message. Any help? Regards, Vineet Menon

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exec_avp problem
by Uri Shacked 12 Mar '12

12 Mar '12

Hi, i am using the exec_avp command as followed: *exec_avp("[/usr/local/etc/kamailio/maintenance/service/service_pr.sh", "$avp(xxx),$avp(yyy)");* ** here is the shell script content: ** *RETVAL=-1 RES=`kamctl cr reload` RETVAL=$? echo "$RETVAL" echo "$RES"* ** I keep on getting 127 as the retval. any ideas why? I must say that two days ago, the script was OK..... :-( where i run the script manualy i have no problems... retval is 0. here is the log with debug 4: *cfg_set_now(): core.debug has been changed to 4 Mar 12 14:20:43 server-Test /usr/local/sbin/kamailio[16589]: DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: 64d3de5c2447acd6006b8539d1246e10 Mar 12 14:20:43 server-Test /usr/local/sbin/kamailio[16589]: DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_LENGTH=0 Mar 12 14:20:43 server-Test /usr/local/sbin/kamailio[16589]: DEBUG: exec [exec_hf.c:267]: SIP_HF_CSEQ=1 OPTIONS Mar 12 14:20:43 server-Test /usr/local/sbin/kamailio[16589]: DEBUG: exec [exec_hf.c:267]: SIP_HF_CALLID=123 Mar 12 14:20:43 server-Test /usr/local/sbin/kamailio[16589]: DEBUG: exec [exec_hf.c:267]: SIP_HF_TO=<**you(a)kamailio.org* <you(a)kamailio.org>*> Mar 12 14:20:43 server-Test /usr/local/sbin/kamailio[16589]: DEBUG: exec [exec_hf.c:267]: SIP_HF_FROM=<**you(a)kamailio.org>;tag=123*<you(a)kamailio.org%3E;tag=123> *Mar 12 14:20:43 server-Test /usr/local/sbin/kamailio[16589]: DEBUG: exec [exec_hf.c:267]: SIP_HF_VIA=SIP/2.0/UDP 127.0.0.1 Mar 12 14:20:43 server-Test /usr/local/sbin/kamailio[16589]: DEBUG: exec [exec_mod.c:208]: executing [/usr/local/etc/kamailio/maintenance/service/service_pr.sh] Mar 12 14:20:43 server-Test /usr/local/sbin/kamailio[16589]: DEBUG: <core> [main.c:852]: SIGCHLD received: we do not worry about grand-children Mar 12 14:20:43 server-Test /usr/local/sbin/kamailio[16589]: CRITICAL: <script>: 127,0*

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Registration Limits
by Reda Aouad 12 Mar '12

12 Mar '12

Hi, Is there a way to ensure single-registration per user-agent for a user, which overwrites previous registration ? Or is there a way to limit the number of registrations per user, but overwriting the earliest registration for each new one ? Thanks, Reda

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Kamailio + Asterisk, what happens when limit for gateway is reached
by Sasa Vilic 12 Mar '12

12 Mar '12

Hello, I have read here about load balancing between multiple gateways by using carrierroute module, but there is one particular question for which I found no answer. Let's say I have two asterisk servers (X und Y), that are my gateways to PSTN. Let we do it simple. Let's say that each gateway supports only one outgoing call at the time and there is no incoming calls from PSTN. Now: Bob calls one number and Kamailio uses X gateway. Alice calls other number and Kamailio uses Y gateway. Alice terminates the call. Alice place a new call, but Bob's call is still in progress. If we were used round robin algorithm, Kamailio would then use X gateway, but X gateway is currently occupied. What shell happen now? And what if Alice didn't tear down call ordinary? What if it her SIP client simply crashed? (Then kamailio would think that there is now free gateway) So the main question is: can kamailio know/detect, how much calls are going over one particular gateway and what can be done when limit for this gateway is reached? Thank you in advance. Kind Regards, Sasa Vilic

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Plans for next major release v3.3.0
by Daniel-Constantin Mierla 12 Mar '12

12 Mar '12

Hello, during the last IRC devel meeting, we discussed about next major release - v3.3.0, proposing to do it before the summer holidays. That means we are going to have about one more month of development, then go to testing phase and release end of May/beginning of June. If anyone is considering different plans/proposals, let us know. The draft of new features in current devel branch are collected at: * http://www.kamailio.org/wiki/features/new-in-devel Please add any missing items there. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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IRC Devel Meeting Transcript
by Daniel-Constantin Mierla 12 Mar '12

12 Mar '12

Hello, took me some time to upload, however, if anyone is interested to read the discussions done during last IRC devel meeting, here is the link: * http://www.kamailio.org/wiki/devel/irc-meetings/2012a#transcripts_of_discus… I will bring soon to the lists via dedicated messages some of the topics that need further discussions. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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XHTTP or XMLRPC modules
by Uri Shacked 11 Mar '12

11 Mar '12

Hi, it worked. i had some kind of error in my config. a question - can XMLRPC and XHTTP work together on the same config? is there no collision there? BR, Uri >Hello, > >can you get the http request with ngrep and send it here? It will be >helpful to see it all, not only what kamailio could parse. > >Also, if the http request is not sending content length header, set the >global parameter: > >http://www.kamailio.org/wiki/cookbooks/devel/core#tcp_accept_no_cl > >Cheers, >Daniel > >On Tue, Mar 6, 2012 at 9:56 AM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: > >* hi, *>* *>* i need to start some kind of proccess in my kamailio server (manualy, not *>* by timer). *>* the best way i can think of is to send a HTTP request. *>* *>* i can do it with the XMLRPC module, but i get an error because i dont *>* realy send XML. *>* i try the XHTTP module. but the event_rout[xhttp_request] does not run. *>* *>* i get : *>* 19(2796) ERROR: <core> [tcp_read.c:965]: ERROR: tcp_read_req: bad request, *>* state=7, error=4 buf: *>* POST / HTTP/1.0 *>* User-Agent: Fiddler *>* Host: 10.2.0.55:5060 *>* *>* parsed: *>* POST / HTTP/1.0 *>* User-Agent: Fiddler *>* Host: 10.2.0.55:5060 *>* any ideas? *

2 1

Enable session_id in ServerHello for TLS
by Kristijan Vrban 11 Mar '12

11 Mar '12

Hello, how to tell that Kamailio should juse a session_id for tls ? See ssldump output below. I reckon that this is the reason the client i use end with "handshake_failure". Because when is use opensips, there is the session_id, and it's working. Kristijan 2 1 0.0228 (0.0228) C>S Handshake ClientHello Version 3.1 cipher suites TLS_RSA_WITH_RC4_128_MD5 TLS_RSA_WITH_RC4_128_SHA TLS_RSA_WITH_NULL_MD5 TLS_RSA_WITH_NULL_SHA TLS_DH_anon_WITH_3DES_EDE_CBC_SHA TLS_DH_anon_WITH_RC4_128_MD5 TLS_RSA_WITH_DES_CBC_SHA TLS_RSA_EXPORT1024_WITH_RC4_56_SHA TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA TLS_DH_anon_WITH_DES_CBC_SHA compression methods NULL 1 0.0519 (0.0519) C>S TCP FIN 2 2 0.0432 (0.0204) S>C Handshake ServerHello Version 3.1 session_id[0]= cipherSuite TLS_RSA_WITH_RC4_128_MD5 compressionMethod NULL 2 3 0.0432 (0.0000) S>C Handshake Certificate 2 4 0.0432 (0.0000) S>C Handshake ServerHelloDone 2 5 0.0452 (0.0020) C>S Alert level fatal value handshake_failure 1 0.0744 (0.0225) S>C TCP FIN 2 0.0681 (0.0228) S>C TCP FIN

2 1

PUA_XMPP: Notify in a non existing dialog
by Dan-Cristian Bogos 10 Mar '12

10 Mar '12

Guys, After my previous discovered issue (Kamailio not happy to talk with Openfire for subscriptions), trying to find work arounds, thought about using pua_mi to enforce subscriptions coming from XMPP side. The following events happen: * I subscribe over mi interface successfully. * Presence sends NOTIFY out. * PUA_XMPP rejects the NOTIFY saying that it does not match an existing dialog. I was wondering where does PUA_XMPP check the existing dialogs and if there is other way to fix my issues? Bellow you can find commands, ngrep as well as kamailio error to syslog. Ta for any kind of tip, DanB """ ### Console ### kamctl fifo pua_subscribe sip:dan@hiddendom.com sip:user01@hiddendom.com presence 3600 ### Trace ### U 2012/03/02 18:17:07.318753 127.0.0.1:5060 -> 127.0.0.1:5060 SUBSCRIBE sip:dan@hiddendom.com SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK5ddb.d2ee4046.0. To: sip:dan@hiddendom.com. From: sip:user01@hiddendom.com;tag=7bcd2f0283a17a6a3eb34b70a7dbcaec-55da. CSeq: 10 SUBSCRIBE. Call-ID: 77d7b19267fdcae8. Content-Length: 0. User-Agent: kamailio (3.2.0 (x86_64/linux)). Max-Forwards: 70. Event: presence. Contact: <sip:user01@hiddendom.com>. Expires: 3610. . # T 2012/03/02 18:17:07.319501 127.0.0.1:35067 -> 127.0.0.1:3306 [AP] .....select status,reason from watchers where presentity_uri='sip:dan@hiddendom.com' AND watcher_username='user01' AND watcher_domain='hiddendom.com' AND event='presence' # T 2012/03/02 18:17:07.319795 127.0.0.1:3306 -> 127.0.0.1:35067 [AP] .....7....def.prxdb.watchers.watchers.status.status.?...........7....def.prxdb.watchers.watchers.reason.reason...@........................... ## T 2012/03/02 18:17:07.319885 127.0.0.1:35067 -> 127.0.0.1:3306 [AP] .....insert into watchers (presentity_uri,watcher_username,watcher_domain,event,status,inserted_time ) values ('sip:dan@hiddendom.com','user01','hiddendom.com','presence',1,1330712227) # T 2012/03/02 18:17:07.320079 127.0.0.1:3306 -> 127.0.0.1:35067 [AP] ........... # U 2012/03/02 18:17:07.320250 127.0.0.1:5060 -> 127.0.0.1:5060 SIP/2.0 202 OK. Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK5ddb.d2ee4046.0;rport=5060. To: sip:dan@hiddendom.com;tag=c87c23e595340de8a004872f7dfab287-19b1. From: sip:user01@hiddendom.com;tag=7bcd2f0283a17a6a3eb34b70a7dbcaec-55da. CSeq: 10 SUBSCRIBE. Call-ID: 77d7b19267fdcae8. Expires: 3600. Contact: <sip:127.0.0.1>. Server: kamailio (3.2.0 (x86_64/linux)). Content-Length: 0. . # T 2012/03/02 18:17:07.320317 127.0.0.1:35067 -> 127.0.0.1:3306 [AP] .....select body,expires,etag,sender from presentity where domain='hiddendom.com' AND username='dan' AND event='presence' order by received_time # T 2012/03/02 18:17:07.320541 127.0.0.1:3306 -> 127.0.0.1:35067 [AP] .....7....def.prxdb presentity presentity.body.body.?...........=....def.prxdb presentity presentity.expires.expires.?...........7....def.prxdb presentity presentity.etag.etag...@.....P...;....def.prxdb presentity presentity.sender.sender...........................Y.<?xml version="1.0"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" entity="dan(a)hiddendom.com"> <tuple id="0x7f7e550f0d70"> <status> <basic>open</basic> </status> </tuple> </presence> 1330712820.a.1330712207.16317.1.1.......... # U 2012/03/02 18:17:07.320996 127.0.0.1:5060 -> 127.0.0.1:5060 NOTIFY sip:user01@hiddendom.com SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKae68.a038faa7.0. To: sip:user01@hiddendom.com;tag=7bcd2f0283a17a6a3eb34b70a7dbcaec-55da. From: sip:dan@hiddendom.com;tag=c87c23e595340de8a004872f7dfab287-19b1. CSeq: 2 NOTIFY. Call-ID: 77d7b19267fdcae8. Content-Length: 345. User-Agent: kamailio (3.2.0 (x86_64/linux)). Max-Forwards: 70. Event: presence. Contact: <sip:127.0.0.1>. Subscription-State: active;expires=3670. Content-Type: application/pidf+xml. . <?xml version="1.0"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" entity="dan(a)hiddendom.com"> <tuple id="0x7f7e550f0d70"> <status> <basic>open</basic> </status> </tuple> </presence> ############# kamailio.log: ############# Mar 2 18:17:07 prx3 /usr/sbin/kamailio[16319]: INFO: presence [notify.c:1593]: NOTIFY sip:user01@hiddendom.com via sip:user01@hiddendom.com on behalf of sip:dan@hiddendom.com for event presence Mar 2 18:17:07 prx3 /usr/sbin/kamailio[16317]: ERROR: pua_xmpp [simple2xmpp.c:198]: Notify in a non existing dialog """

2 4

Kamailio and MySQL/Galera Replication...
by Krishna Kurapati 10 Mar '12

10 Mar '12

Hi, I am trying to setup Kamailio in an Active-Active configuration behind a TCP load balancer. On the database side, looking at MySQL cluster as independent node set vs. Kamailio/MySQL server with Galera synchronous replication. The first configuration requires 3 or 5 servers. 2 for Kamailio and 3 for MySQL. The second probably just requires 2 servers. I am leaning towards Kamailio/MySQL/Galera. Could you share setup/performance issue... Thanks Krish Kura

1 0

insert_hf in CANCEL message
by Reda Aouad 10 Mar '12

10 Mar '12

Hi, insert_hf doesn't work for CANCEL messages. It doesn't work either on 487 replies in failure_route.. Why? Is it because they are re-generated on every hop? Is there a workaround? Reda

2 1

SIP problems using a comrex Access
by Andy Bosley 10 Mar '12

10 Mar '12

Hi Support, We have a comrex access unit that we have previously successfully used for connections using your SIP service. For some reason now though, we can no longer get the unit to work. We know the username (LCR.ACCESS1) is ok as we can access the account through the website. However, entering the same details in to the comrex fails. The comrex status displays "OFFLINE (Registering)" and does not get any further. Is it possible for you to check that the account is ok and that the unit has successfully been authenticated. regards Andy __________________________ Andy Bosley Supervisor, London Control Room Technology Operations Technology Division, BBC Room 3045, Broadcasting House Tel External: +44 02077 654976 Tel Internal: 54976 http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this.

2 1

XHTTP or XMLRPC modules
by Uri Shacked 10 Mar '12

10 Mar '12

hi, i need to start some kind of proccess in my kamailio server (manualy, not by timer). the best way i can think of is to send a HTTP request. i can do it with the XMLRPC module, but i get an error because i dont realy send XML. i try the XHTTP module. but the event_rout[xhttp_request] does not run. i get : 19(2796) ERROR: <core> [tcp_read.c:965]: ERROR: tcp_read_req: bad request, state=7, error=4 buf: POST / HTTP/1.0 User-Agent: Fiddler Host: 10.2.0.55:5060 parsed: POST / HTTP/1.0 User-Agent: Fiddler Host: 10.2.0.55:5060 any ideas?

2 1

Backing up htable to SQL table ...
by Krishna Kurapati 09 Mar '12

09 Mar '12

Hi, This is my first time attempt using htable module. How can I back up the htable content in the database incase the server restarts. I saw a htable in the MySQL database. My observation has been that htable module is not automatically backing up to database even though I added modparams to point to the database. Do I need to explicitly backup htable content to database using sqlops or some other mechanism? Thanks Krish Kura

3 6

Advice on placement of t_replicate (REGISTER)
by Asgaroth 09 Mar '12

09 Mar '12

Hi Guru's, I'd just like some advice on t_replace and its placement in the routing script. A lot of examples show the following setup: save("location"); t_replicate("backup_server"); Would it not be wiser to have something like: if (save("location")) { t_replcate("backup_server"); } Would there be any scenario where you would want to t_replicate if save("location") fails for some reason? The configuration I refer to is if the location database is in memory. I suspect you may want to replicate all if location is going to db core in case of external failure. But would it matter on the placement if the location was in memory? Thanks

2 1

kamailio with flowroute
by romon.zaman 09 Mar '12

09 Mar '12

hello room, i was trying to add sip provider(like. flowroute,vitelity) with user-pass authentication in kamailio to accept inbound calls. any help? thanks

3 2

no hangup when there is no network or when the UAC on the other side is power off
by Rabary 08 Mar '12

08 Mar '12

Hi everybody, We are using kamailio+callcontrol+cdrtool+freeradius+mysql to build a voip prepaid based system: . 2 SIP accounts created and one of them is prepaid . we can debit balance when we call with the SIP prepaid account, and when the balance is 0, the call hang up . if we hangup a call session from the prepaid SIP and the balance is not still 0, the call hangup and the system debit the appropriate cost . but when we call from the prepaid SIP account, and we disconnect the ethernet cable or the alimentation of the other UAC, the call don't hangup until the balance is set to 0 . if the phone is still ringing too, when we disconnect the other UAC, the caller UAC still ringing . the 2 uacs and kamailio are in the same subnet network the question is, how can we tell to kamailio to detect if the other side is down and then send bye immediately so thecalculating of the credit will be correct? Thank you. --

1 0

Rewrite $tU
by Lucas Alvarez 08 Mar '12

08 Mar '12

I want rewrite $tU but I'm not being able, I'm doing the following: remove_hf("To"); insert_hf("To: sip:$rU@$rd\r\n", "From");

4 8

Limiting registrations to known users without auth
by Pedro Antonio Vico Solano 08 Mar '12

08 Mar '12

3 4

default IP to forward TCP request
by Laurent Schweizer 07 Mar '12

07 Mar '12

Hello all, I have a linux server with different IP. eth0 -> 10.2.2.4 eth0:1 -> 10.2.2.5 I have kamailio listening on IP: listen=udp:10.2.2.5:5060 listen=tcp:10.2.2.5:5060 and in the script I do : rewritehostport("10.2.2.10:5060;transport=tcp"); if (!t_relay()) { sl_reply_error(); } exit; the problem is that t_relay is not using the IP 10.2.2.5 to forward this SIP request but the eth0 IP ( 10.2.2.4) I see that they is a function force_send_socket , I need to use this function to set the correct IP used to forward the request ? As I'm using only one IP 10.2.2.5 for openser, why he cannot always use this ip as default Ip ? Best regards Laurent

3 2

Re: [SR-Users] sr-users Digest, Vol 82, Issue 21
by Javier Gallart 07 Mar '12

07 Mar '12

Hello this is what I normally use: if (defined $var(x)) {...} Regards Javi > ------------------------------ > > Message: 4 > Date: Tue, 6 Mar 2012 20:05:32 -0600 > From: Krishna Kurapati <kkurapat(a)gmail.com> > Subject: [SR-Users] Improving Kamailo configuration parser ... > To: kamailio users <sr-users(a)lists.sip-router.org> > Message-ID: > <CAOzCjAfcaRQ+HT3r+mwckPyaVJH+EVFWNa_Hj6tAhGyD2DaMJA(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > I used > > if ($var(x) != null) { ... } > > in one of the routes, and tried to run Kamailio server. > The server exited with an error. I did not know why. > > Once I changed it to if ($var(x) != $null) {} the error went away. > I missed $ infront null. I think it is hard to trace such problems once a > large chunk > of logic is written since many of us come from C++ and scripting background > where $null > is not a common practice. > > Also if I tried to use if (!$var(x)) { ... } Kamailio tries to convert the > value of $var(x) to integer. > prints WARNINGS in the log when the $var(x) consists of a string. > > Krish Kura >

1 0

Improving Kamailo configuration parser ...
by Krishna Kurapati 07 Mar '12

07 Mar '12

I used if ($var(x) != null) { ... } in one of the routes, and tried to run Kamailio server. The server exited with an error. I did not know why. Once I changed it to if ($var(x) != $null) {} the error went away. I missed $ infront null. I think it is hard to trace such problems once a large chunk of logic is written since many of us come from C++ and scripting background where $null is not a common practice. Also if I tried to use if (!$var(x)) { ... } Kamailio tries to convert the value of $var(x) to integer. prints WARNINGS in the log when the $var(x) consists of a string. Krish Kura

2 1

msilo is failing to in db connection...
by Krishna Kurapati 06 Mar '12

06 Mar '12

Hi all, I am trying to use msilo module from modules_k. Unfortunately it if failing during DB initialization. Other modules are initializing fine. I am using Kamailio 3.2.2. Not sure why the connection is not found. I tried to increase the MAX mysql connections to 250. I see the same error after I increase the connections. The config file is pretty standard: loadmodule "msilo.so" modparam("msilo", "add_date", 0); modparam("msilo", "db_url", DBURL); # The DBURL works for other modules... Here is the Log output: 0(14115) DEBUG: <core> [cfg.y:1684]: loading module msilo.so 0(14115) DEBUG: <core> [sr_module.c:557]: load_module: trying to load </usr/local/kamailio-3.2.2/lib64/kamailio/modules_k/msilo.so> ..... 0(14117) DEBUG: <core> [sr_module.c:928]: DEBUG: init_mod: msilo 0(14117) DEBUG: <core> [sr_module.c:667]: find_export_record: found <db_bind_api> in module db_mysql [/usr/local/kamailio-3.2.2/lib64/kamailio/modules/db_mysql.so] 0(14117) DEBUG: <core> [db.c:212]: using db bind api for db_mysql 0(14117) DEBUG: <core> [db.c:299]: connection 0x7fc20d1ace60 not found in pool 0(14117) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/openser 0(14117) ERROR: db_mysql [km_my_con.c:109]: driver error: Access denied for user 'openser'@'localhost' (using password: YES) 0(14117) ERROR: <core> [db.c:303]: could not add connection to the pool 0(14117) ERROR: msilo [msilo.c:319]: failed to connect to the database 0(14117) ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module msilo (/usr/local/kamailio-3.2.2/lib64/kamailio/modules_k/msilo.so) ERROR: error while initializing modules Any help? Thanks

3 6

Re: [SR-Users] msilo is failing to in db connection...
by Daniel-Constantine Mierla 06 Mar '12

06 Mar '12

fwd to the list - reply all failed in first place ... > Hello, > > the module connects in mod init to check the table version, otherwise it does not do any message storage related operations with the database. It should work fine at runtime even if mod init does not connect to db. > > But it is strange of mod init fails but child init goes on. Do you have a firewall with policies related to active connections (rate of new connections)? > > Cheers, > Daniel > > On Sat, Mar 3, 2012 at 6:16 PM, Krishna Kurapati <kkurapat(a)gmail.com> wrote: > Further investigation into the issue revealed that the db_conn in the module_init is failing... When I commented out the code, > the module is initialized ok. Not sure if this is will fine. > > #if 0 > db_con = msilo_dbf.init(&ms_db_url); > if (!db_con) > { > LM_ERR("failed to connect to the database\n"); > return -1; > } > > if(db_check_table_version(&msilo_dbf, db_con, &ms_db_table, S_TABLE_VERSION) < 0) { > LM_ERR("error during table version check.\n"); > return -1; > } > if(db_con) > msilo_dbf.close(db_con); > db_con = NULL; > #endif > > Is the module not suppose to connect to database in the module init? > > Thanks > > > On Sat, Mar 3, 2012 at 10:13 AM, Krishna Kurapati <kkurapat(a)gmail.com> wrote: > Hi, > > Thanks for the response. I looked at the DBURL and it looked just fine. I wanted to test and see if the module is reading the config > param or not, I deliberately changed it to > > modparam("msilo", "db_url", "mysql://root:xxxxx@localhost/openser"); > > When I ran it, the logs shows the user as 'openser' not the 'root' that I gave in config param. May be somewhere in the code, > db_url is being overwritten by DEFAULT_DB_URL. > > 0(28806) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/openser > 0(28806) ERROR: db_mysql [km_my_con.c:109]: driver error: Access denied for user 'openser'@'localhost' (using password: YES) > 0(28806) ERROR: <core> [db.c:312]: could not add connection to the pool 0(28806) ERROR: msilo [msilo.c:365]: failed to connect to the database > > Thanks > Krish Kura > > On Sat, Mar 3, 2012 at 5:14 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > the error says the username, password or IP address are invalid to connect to mysql -- access denied. There does not seem like being other connections using same db_url, check for typos in db_url parameter. > > Cheers, > Daniel > > > > On 3/3/12 12:39 AM, Krishna Kurapati wrote: >> Hi all, >> >> I am trying to use msilo module from modules_k. Unfortunately it if failing during DB initialization. >> Other modules are initializing fine. I am using Kamailio 3.2.2. >> >> Not sure why the connection is not found. I tried to increase the MAX mysql connections to 250. I see the >> same error after I increase the connections. >> >> The config file is pretty standard: >> >> loadmodule "msilo.so" >> modparam("msilo", "add_date", 0); >> modparam("msilo", "db_url", DBURL); # The DBURL works for other modules... >> >> Here is the Log output: >> >> 0(14115) DEBUG: <core> [cfg.y:1684]: loading module msilo.so >> 0(14115) DEBUG: <core> [sr_module.c:557]: load_module: trying to load </usr/local/kamailio-3.2.2/lib64/kamailio/modules_k/msilo.so> >> >> ..... >> >> 0(14117) DEBUG: <core> [sr_module.c:928]: DEBUG: init_mod: msilo >> 0(14117) DEBUG: <core> [sr_module.c:667]: find_export_record: found <db_bind_api> in module db_mysql [/usr/local/kamailio-3.2.2/lib64/kamailio/modules/db_mysql.so] >> 0(14117) DEBUG: <core> [db.c:212]: using db bind api for db_mysql >> 0(14117) DEBUG: <core> [db.c:299]: connection 0x7fc20d1ace60 not found in pool >> 0(14117) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/openser >> 0(14117) ERROR: db_mysql [km_my_con.c:109]: driver error: Access denied for user 'openser'@'localhost' (using password: YES) >> 0(14117) ERROR: <core> [db.c:303]: could not add connection to the pool 0(14117) ERROR: msilo [msilo.c:319]: failed to connect to the database >> 0(14117) ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module msilo (/usr/local/kamailio-3.2.2/lib64/kamailio/modules_k/msilo.so) >> ERROR: error while initializing modules >> >> Any help? >> >> Thanks >> >> >> >> _______________________________________________ >> 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://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > > > -- > Daniel-Constantin Mierla > http://www.asipto.com

1 0

disptacher failover
by Klaus Darilion 06 Mar '12

06 Mar '12

Hi! Imagine a dispatcher list with 3 entries. I wonder what happens if one of the targets fail. Is the traffic moved evenly to the remaining 2 nodes or is the whole traffic of the broken node moved to one of the nodes? I took a quick look at the code and it seems that the whole traffic is moved to the next node in the list (i=i+1): while (ds_skip_dst(idx->dlist[i].flags)) { if(ds_use_default!=0 && idx->nr!=1) i = (i+1)%(idx->nr-1); else i = (i+1)%idx->nr; Is my assumption correct? Thanks Klaus

2 1

Question regarding... LCR Module
by kamailio Padawan 06 Mar '12

06 Mar '12

Hello, I have attempted to ask questions on IRC with no response. I really appreciate any help from anybody! I am using Kamailio 3.2.2. I am unable to get anything loaded in $avp(i:709). I am attempting to route between 2 media handling asterisk boxes. I am trying to load balance between them. I have the following entries in my tables... *lcr_gw* * id,lcr_id,gw_name,ip_addr,hostname,port,params,uri_scheme,transport,strip,prefix,tag,flags,defunct * 1, 1, media1, *1.1.3.123*, *1.1.3.123*, 5060, , 1, 1, 0, 0, 0 2, 1, media2, *1.1.4.123*, *1.1.4.123*, 5060, , 1, 1, 0, 0, 0 *lcr_rule* *id,lcr_id,prefix,from_uri,stopper,enabled* * * 1, 1, *[1][0-9][0-9][0-9][0-9]*, 0, 1 *lcr_rule_target* *id,lcr_id,rule_id,gw_id,priority,weight* 1, 1, 1, *1*, 1, 5 2, 1, 1, *2*, 1, 5 I was under the impression that, if my from URI was 15555, and I have 2 gateways, that I can call *load_gws("1","$fU")* and return the gateways that match the *from_uri* rule of 15555 , then look the rule_id up in * lcr_rule_target* and return the *2* gws gw_id[1] and gw_id[2], which are the 2 entries in the* lcr_gw* table..... and it would be weighted 5 and 5 so that it would load balance. Right now..... I get nothing loaded in $avp(i:709). I thought this would show the IP/Hostname of the media1 or media2 gws. The variable is null far before the call to *load_gws*. I'm loading it at the top in a modparam like *modparam("lcr", "gw_uri_avp", "$avp(i:709)")* *modparam("lcr", "ruri_user_avp", "$avp(i:500)")* When i get the result of calling the *load_gws*() function into a variable. The variable is *2*. Is that the number of rows returned? Why is that parameter [avp(i:709)] always empty? How can i get next_gw() to complete and not fail? Thank you so much.... I am very new to Kamailio and am very interested... Kamailio Padawan Learner

2 1

TOS value in configuration
by Olle E. Johansson 04 Mar '12

04 Mar '12

I've set tos=IPTOS_RELIABILITY in my configuration, which follows https://www.kamailio.org/wiki/cookbooks/3.2.x/core#tos which has the following examples: tos=IPTOS_LOWDELAY tos=0x10 tos=IPTOS_RELIABILITY but I get a configuration error: 0(45338) : <core> [cfg.y:3501]: parse error in config file /usr/local/etc/kamailio/kam-global.inc, line 29, column 5-21: syntax error 0(45338) : <core> [cfg.y:3501]: parse error in config file /usr/local/etc/kamailio/kam-global.inc, line 29, column 5-21: number expected Has this option been deprecated or is it a small bug? /O

2 1

Facebook page for Kamailio
by Olle E. Johansson 04 Mar '12

04 Mar '12

Just to make sure we're in the flow, I've started a Facebook page for Kamailio. http://www.facebook.com/kamailio Feel free to like it if you're a Facebook user. There's also a Google+ page. If you have news to share, you're welcome to add it on these pages. /O

2 1

Building Up a Kamailio Server
by Faisal Rehman 03 Mar '12

03 Mar '12

Hi Everyone, I am totally new to Kamailio & going to build up Kamailio server as an XMPP server with the following wanted functionalities: -Have XMPP client to register to my kamailio server - The XMPP clients should all be able to see each others presence - The XMPP clients should be able to IM other XMPP clients - The registered sip client will be able to just dial any other sip client. No need for any gateway or trunk for now I have a good previous background with SIP but its my first time to work with XMPP & Kamailio & that's why I am in need of a bit help & assistance. I have installed Kamailio on my CentOS box & now going to start to configure it in order to complete the above requirements, so any links or any refferals will be highly appreciated. Regards, Faisal Rehman

2 5

XMPP module not passing subscribes coming from xmpp side
by Dan-Cristian Bogos 02 Mar '12

02 Mar '12

Hey Guys, I was wondering if anyone having experience with using xmpp module as presence gateway between SIP and XMPP. I started evaluating it and till now got some weird results, not sure if it is just my setup which does not match the one widely used or the module is incomplete. Regarding my setup, I use OpenFIRE 3.7.1 on one side and Kamailio 3.2.0 out of debian packages on the other. The issue I got right now is that the subscribe coming from XMPP side produces no action/reply on SIP side. Bellow you can see the packet coming from OpenFIRE: # T 2012/03/02 09:49:57.743750 127.0.0.1:5275 -> 127.0.0.1:49965 [AP] <presence id="4h8F6-19" to="dan4(a)gw.mydomain.com" type="subscribe" from="dan(a)mydomain.com"><c xmlns="http://jabber.org/protocol/caps" ext="voice-v1 video-v1 camera-v1 " hash="sha-1" node="http://jitsi.org" ver="ohjz8WKq0ZQRiNWRNcapX4BBHLk="/></presence> """ Anyone here got this working somehow? Ta, DanB

2 1

iOS push notification using Kamailio
by Krishna Kurapati 02 Mar '12

02 Mar '12

Hi, I have been looking at the usrloc module's source code to see how the uuid that is sent in +sip.instance parameter in the Contact header. Apparently, the module ensures that only one contact per uuid exists. However the uuid is not stored in the database. The reason I need the uuid in the database is to access it when the iOS device is offline inorder to send a push notification. I am using pjsip which supports sending the uuid of the device ( w/ rfc 5626 support) in the REGISTER message. Has anyone implemented push notification using this or other methods using Kamailio? Thanks Krish Kura

2 1

automatically switch to TCP
by Klaus Darilion 01 Mar '12

01 Mar '12

Hi! Is there a feature somewhere (core, tm) to automatically switch from UDP to TCP? It may be useful with bodies > 64K. regards Klaus

2 1

Kamailio error code 500 (6/SL)
by Pavel Klochan 01 Mar '12

01 Mar '12

Hi. I'm a newbie to Kamailio (so do not judge strictly). I have experience only with Asterisk and now I'm trying to configure Kamailio. With default config all is working as expected, but I'm trying to configure it from zero with minimal functions, but using parts from default config (those parts that I figured out). With my present config local calls are working great, but I can't make outbound calls. I've attached config file, sip-trace and debug log. I think there is a route loop somewhere but I can't figure out where. I would be grateful if you pointed out to me where exactly my mistake or just point me in the right direction =) Thanks in advance

2 2

loose_route() with pre-loaded Route in ACK
by Andreas Granig 01 Mar '12

01 Mar '12

Hi, I ran into an interesting scenario when experimenting with kamailio 3.1 as a stateless proxy (no tm module used): There is an initial INVITE without to-tag and with pre-loaded Route header pointing to the external address of the lb. When I call loose_route for that, it returns false and the Route header is removed. That's what I expected, and it's documented in http://www.kamailio.org/docs/modules/3.1.x/modules_k/rr.html#loose-route-id. However if a 407 is relayed back from the registrar via the lb to the UA, it sends an ACK, again with pre-loaded Route header like in the initial INVITE, this time with to-tag. In that case, loose_route() returns true, but the $du is $null and the R-URI is still the same. So the Route is consumed, but the URI is not used. Since it's a stateless proxy, this hop-by-hop ACK needs to be routed the same way as the initial INVITE, so what I'm doing is something like that: if(loose_route()) { if($du == $null && $ru == $ou) { # skip loose-routing for such a request } } Is this how it's supposed to work? And is there some documentation of how loose_route() detects whether Route points to itself or not? I'm not using auto_aliases, nor do I set any aliases manually. Does it determine internally whether the Route points to its local ip and port? If not, how come the $du is not set although loose_route returns true? Andreas

2 9

DRouting cause Kamailio 3.2 to stop
by Fatima Chahrour~Vanrise Support 01 Mar '12

01 Mar '12

Hi, I am newly experiencing kamailio 3.2. I have installed kamailio and MySQL and tried the test call from 1 PC(x-lite) to another PC(x-lite) and ended successfully. Now moving further, I have configured dynamic routing module and applied needed parameters and scripts to finally be able to start kamailio without any issue. Also I set a testing scenario in the related database tables BUT my problem is that whenever am trying to call the predefined prefix (961) from x-lite am getting a 'calling' message on it and then request timeout where I find that kamailio has stopped running. I kamctrl start again and call 961 then get calling so I wait and kamailio stps again. Below is the script used in the routing logic and database record. Any help to achieve a successful call is highly appreciated. dr_gatways: 3, 1, '192.168.111.15', 0, '', '', 2, 'PC1' dr_groups: 2, '.*', '.*', 0, 'Anyone' dr_gw_lists: 2, '3', 'Test Call' dr_rules: 2, '0', '961', '', 0, '1', '#2', 'Lebanon Fix' ####### Routing Logic ######## # main request routing logic route{ if (is_method("INVITE")) { setflag(1); # do accounting route(4); # Call our dynamic route. } } .. .. .. route[4] { if (!do_routing("0")) { send_reply("503", "No Rules matching the URI"); exit; } if (is_method("INVITE")) { t_on_failure("4"); } route(RELAY); } .. .. .. failure_route[4] { if (use_next_gw()) { t_relay(); exit; } else { t_reply ("503", "Service not available"); exit; } } Thanks in Advance, F.Chahrour

4 8

memory allocation failure
by Robert R 29 Feb '12

29 Feb '12

Hi, The openser service died on my servers after "memory allocation failure". This issue has been occurred a couple of times on two different servers within the last two months. Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:core:do_action: memory allocation failure Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:core:pv_set_ruri_user: do action failed Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:core:do_assign: setting PV failed .... Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:core:insert_cond_lump_after: out of pkg memory Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:rr:build_rr: failed to insert lumps Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:rr:record_route: failed to insert inbound Record-Route Feb 29 08:31:59 kernel: [8806798.263792] openser[7115] general protection rip:7f6e17eae69c rsp:7fff21dac5b0 error:0 Feb 29 08:31:59 /usr/sbin/openser[7164]: CRITICAL:core:receive_fd: EOF on 22 It seems increasing memory size in /etc/default/openser to 1024 didn't fix the issue. # Amount of memory to allocate for the running OpenSER server (in Mb) MEMORY=1024 Any advice would be appreciated. Thanks, R

2 1

Presence between different domains
by Manuel Perez 29 Feb '12

29 Feb '12

Hello, I've used kamailio 3.2.2, i have rls, simple presence and xcap support working. I've replicated this server into another one in a different domain, but presence between domains is not working. I'm using for presence presence.so, presence_xml.so and presence_mwi.so. In route[WITHINDLG] i have the following lines: ... if (is_method("SUBSCRIBE|NOTIFY") && uri==myself) route(PRESENCE); ... When NOTIFY from user B at domain B arrives to the server at domain A, the message is not forwarded to the user A, How can i achieve this? Thank you.

2 3

dialog doesn't clear calls that were re-INVITEd shortly after ACK
by Andrew Pogrebennyk 29 Feb '12

29 Feb '12

Hi, I noticed dialog module showing calls that have already been terminated with state=4. The kamailio version is 3.1.5, I'm aware of certain limitations of dialog module in this version (must use stateless replies, create dialog after 407), but this is something rather new. In kamailio config I'm using dlg_manage() for initial INVITEs and BYE/CANCEL and in-dialog requests are loose-routed, so nothing finicky. The call scenario is: A calls B, B answers the call, A immediately after sending ACK for the 200 OK issues re-INVITE. After some seconds A hangs up, and the dialog stays in memory in state=4, e.g.: dialog:: hash=3304:623200556 state:: 4 ref_count:: 2 timestart:: 1329938667 timeout:: 78590167 callid:: MGExMjRiYzMzN2Q5YWU4YjRiMjFjYTYwYWZlMGFlMzY. from_uri:: sip:sipwise-user1@67.202.62.202 from_tag:: 4207e429 caller_contact:: sip:sipwise-user1@80.108.64.151:46420 caller_cseq:: 3 caller_route_set:: <sip:127.0.0.1:5060;lr;r2=on>,<sip:67.202.62.202:5060;lr;r2=on> caller_bind_addr:: udp:127.0.0.1:5062 callee_bind_addr:: udp:127.0.0.1:5062 to_uri:: sip:43991002@67.202.62.202 to_tag:: 6C8F89E9-4F4540EA0001F575-AC557700 callee_contact:: sip:sipwise-user2@127.0.0.1:5080 callee_cseq:: 11 callee_route_set:: When I set dlg_match_mode=1 the problem goes away. I would like however to understand why the dialog fails to clear in default mode (DID_ONLY), because the dialog cookie seems to be preserved by caller in all in-dialog requests properly. There is nothing in the debug log and trace attached that catches my eye. Do you see anything wrong there? In trace file the proxy address is 127.0.0.1:5062. You may notice that the actual signaling flow is a little bit more complicated, I will gladly answer any questions about it. For this test I've used two lines in the eyeBeam softphone, when one line answers the other is automatically put on hold which is what I want. Then I un-hold the callee and clear the call. The customer is reporting multiple stale calls when re-INVITE is sent by Cirpack PSTN gw immediately after call connect, which is basically the same call flow I am talking about, it is just not practical for me to run kamailio with debug=3 there all the time. I am going to try theredlg_match_mode=1 though and see if the problem goes away. Any ideas?

1 1

Re: [SR-Users] sr-users Digest, Vol 81, Issue 95
by Yufei Tao 29 Feb '12

29 Feb '12

Hi Thanks for your reply! I already set the accounting flag like you say. And all normal BYEs are inserted into the acc table fine. Just internally generated BYEs don't get inserted in acc table. I've done a bit googling and added event_route: event_route[tm:local-request] { if (is_method("BYE")) { xlog("L_INFO", "event_route: BYE: callid=$ci, cseq=$cs - internally generated\n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } } When I have two clients in a call, I disconnect the network for both clients, so no RTP either direction, which triggered mediaproxy to send dlg_bye_dlg to Kamailio, which sent BYE to both clients. I can see from the log that event_route did get executed, and BYE sent to both clients, but still BYE isn't inserted to acc table. The code inside event_route() above is copied from my route[WITHINDLG], which for client-generated BYEs operates correctly (inserts BYEs into acc table). But in event_route it didn't work. I also tried to use sql_query to insert BYE into acc table in event_route and it causes Kamailio to crash. Using sql_query to select * from acc also causes Kamailio to crash. Any idea where I've done wrong? I'm very much lost. Thank you! Yufei > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 27 Feb 2012 13:21:50 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] How to do accounting for Kamailio-generated > BYEs? > To: sr-users(a)lists.sip-router.org > Message-ID: <4F4BC9BE.8000704(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 02/27/2012 12:58 PM, Yufei Tao wrote: >> Hi >> >> I've got a mediaproxy that detects no media, and sends dlg_end_dlg to >> Kamailio via mi_datagram socket. Then Kamailio sends BYE to both calling >> parties. It all works fine. But my question is, how to do accounting for >> these Kamailio-generated BYEs? They don't appear in the acc table. > In the loose_route() section of the config script, set the accounting > flag for BYEs, e.g. > > if(is_method("BYE")) > setflag(xx); > > -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

1 2

Re: [SR-Users] CallControl and MediaProxy
by Reda Aouad 29 Feb '12

29 Feb '12

Ok thanks Daniel. I'll do what you suggested and we'll see how to proceed. Thanks again Reda On 27 févr. 2012, at 17:46, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, On 2/26/12 11:01 AM, Reda Aouad wrote: Daniel, I sent you the log file by private email, for privacy concerns. ok, no time yet to look over them, but as you say it does not call mediaproxy explicitely from config and callcontrol, maybe you should look inside mediaproxy module, might be automatically executed on dialog or tm module callbacks -- then the patch has to be done for mediaproxy module. I took a quick look at the log files, and didn't see any call to mediaproxy module. Though, the mediaproxy log file indicates a media session engaged, and CDRTool shows media information for the established session. The mediaproxy module is loaded, but no function from the module is called at all. It can be via some internal callbacks registered to some modules such as tm and dialog, or even script execution callbacks... so look over the code of mediaproxy module. You can run kamailio with debug=3 and from the logs you should get some hints about what is the way media proxy is executed. Cheers, Daniel Thanks Reda On Sat, Feb 25, 2012 at 18:58, Reda Aouad <reda.aouad(a)gmail.com> wrote: > I looked in detail in the source of the call control module and didn't > find anything related to the mediaproxy module neither.. > I'll send you the output of cfgtrace soon. > > Reda > > > > On Fri, Feb 24, 2012 at 23:46, Daniel-Constantin Mierla <miconda(a)gmail.com > > wrote: > >> A quick grep over the callcontrol sources didn't reveal a point where >> mediaproxy is engaged. >> >> Can you load debugger module and enable cfgtrace, then run such a >> scenario and send out the output from cfgtrace to see all the config >> actions executed? >> >> Cheers, >> Daniel >> >> >> On 2/24/12 7:56 PM, Reda Aouad wrote: >> >> It would be great Daniel if you could do it. >> I will be more than happy to test it. >> A function parameter would be more flexible than a module parameter. >> I expect it shouldn't affect the behavior of the external application >> CallControl. >> >> Thanks in advance :) >> >> Reda >> >> >> >> On Fri, Feb 24, 2012 at 08:09, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> I am not using mediaproxy at all (but nathelper/rtpproxy), neither the >>> call control module, but making an option (module parameter or function >>> parameter) for call control to bind to another module like media proxy, >>> should not be big deal if it is all you are looking for -- I can look over >>> it and send a patch if you are going to help testing it. I cannot do it >>> these days, though, being out of the office. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 2/23/12 8:59 PM, Reda Aouad wrote: >>> >>> First, I am posting about the wrong behavior of CallControl (or most >>> probably Kamailio modules) which leaves no option. I should be the only >>> one deciding about how to handle timeouts. If I decide to take some risk, >>> no module should oblige me to do otherwise. >>> >>> Mediaproxy detects ONLY RTP timeouts from BOTH parties, because linux >>> conntrack rules it uses are bi-directional. If a single party stops sending >>> RTP for whatever reason (connection lost, codec with silence detection >>> used, ....), mediaproxy doesn't care and doesn't act upon it. This is a >>> feature, and a wanted one, to mainly support voice-detecting codecs. Think >>> also about conferences for example, in which only a single person talks for >>> a long time while others are silent and don't send RTP. >>> >>> Single-side RTP timeout because of a real problem (loosing network >>> connection for example) should be handled with other methods, such as SIP >>> session timers. >>> >>> MY POINT IS : I don't see it practical to handle RTP flows for EVERY >>> call to handle the least probable scenario: an RTP timeout from both (or >>> all) parties. >>> >>> If I understood well, mediaproxy updates the CDR when it detects an >>> RTP timeout from both parties. CallControl can look in the CDR to debit the >>> correct balance, instead of attaching itself to the dialog module to detect >>> dialog termination. >>> >>> This is an extract from the call_control module : >>> >>> Even when mediaproxy is unable to end the dialog because it was not >>> started with engage_media_proxy(), the callcantrol application is still >>> able to detect calls that did timeout sending media, by looking in the >>> radius accounting records for entries recorded by mediaproxy for calls that >>> did timeout. These calls will also be ended gracefully by the callcontrol >>> application itself. >>> >>> >>> Unless there is something I miss.. >>> >>> I also opened a bug about the issue because call_control doesn't have >>> the same behavior with OpenSips. It doesn't force mediaproxy. >>> >>> Reda >>> >>> >>> >>> On Thu, Feb 23, 2012 at 20:00, Jeff Brower <jbrower(a)signalogic.com>wrote: >>> >>>> Reda- >>>> >>>> > It's clear but not necessary. It can look at radius records fixed by >>>> > mediaproxy on RTP timeout to debit the correct balance as well. And >>>> why >>>> > also force it on postpaid calls which it doesn't control at all ? >>>> >>>> I don't understand how you plan to tear down Kamailio calls that >>>> suffer RTP time-out? >>>> >>>> -Jeff >>>> >>>> > What happens is cost and performance issues for additional calls >>>> passing >>>> > through my mediaproxy server, which I didn't plan for at first. No >>>> audio >>>> > issue at all. >>>> > >>>> > Reda >>>> > >>>> > >>>> > >>>> > On Thu, Feb 23, 2012 at 11:58, Sammy Govind <govoiper(a)gmail.com> >>>> wrote: >>>> > >>>> >> Reading from the module docs its clear why it needs to engage >>>> media/rtp >>>> >> proxy to start,stop billing or timer of a call. so what happens when >>>> it >>>> >> engages mediaproxy on unwanted calls !? audio-issues? >>>> >> >>>> >> >>>> >> On Thu, Feb 23, 2012 at 1:21 PM, Reda Aouad <reda.aouad(a)gmail.com> >>>> wrote: >>>> >> >>>> >>> Thanks Sammy. I didn't get any reply yet. >>>> >>> >>>> >>> CallControl is an application used with CDRTool for prepaid calls. >>>> It >>>> >>> calculates the maximum call duration based on the user's balance. >>>> Once the >>>> >>> call's duration limit is reached, it sends BYE to both calling >>>> parties, >>>> >>> terminating the call. At the end of a prepaid call, terminated >>>> either by >>>> >>> the user or by CallControl, it debits the user's balance according >>>> to the >>>> >>> call's duration. >>>> >>> >>>> >>> The Call_Control module interfaces with this external application. >>>> >>> >>>> >>> call_control function is called in Kamailio's cfg to check if the >>>> user >>>> >>> has prepaid or postpaid account, and get the max call duration for >>>> prepaid >>>> >>> users. CallControl controls only prepaid calls, not postpaid ones. >>>> >>> >>>> >>> So call control and NAT traversal using mediaproxy are two >>>> differents >>>> >>> things which i can't link, since I don't want mediaproxy for every >>>> call. >>>> >>> And since the function call_control is called on every invite before >>>> >>> knowing if the user has a prepaid account or not, it engages >>>> mediaproxy for >>>> >>> every call. >>>> >>> >>>> >>> CallControl relies on mediaproxy to detect RTP timeouts and debit >>>> the >>>> >>> correct balance from a prepaid account based on the last instant the >>>> >>> mediaproxy saw an RTP packet. >>>> >>> >>>> >>> But why to force using mediaproxy with no choice? And why to force >>>> it for >>>> >>> every call, whether it falls under CallControl's control or not? >>>> >>> >>>> >>> I am using Kamailio 3.2. >>>> >>> >>>> >>> >>>> >>> Reda >>>> >>> >>>> >>> On 23 févr. 2012, at 07:21, Sammy Govind <govoiper(a)gmail.com> >>>> wrote: >>>> >>> >>>> >>> Hi, >>>> >>> I can see you posting multiple times on both proxies listings so >>>> I'm sure >>>> >>> you havent heard back from anyone.I am not at all familiar with your >>>> >>> functions in email but could it be possible for you to determine on >>>> which >>>> >>> calls you need to engage mediaproxy and on which not to, then on >>>> the base >>>> >>> of that flag use the call_control function ! >>>> >>> your problem is complicated for me atleast. I hope somebody could >>>> answer >>>> >>> you accurately and precisely. >>>> >>> >>>> >>> btw, what are you using in real? opensips or kamailio, which >>>> version? and >>>> >>> in what context you need to use the call_control function? >>>> >>> >>>> >>> Thanks, >>>> >>> Sammy >>>> >>> >>>> >>> On Thu, Feb 23, 2012 at 12:45 AM, Reda Aouad <reda.aouad(a)gmail.com >>>> >wrote: >>>> >>> >>>> >>>> Hi, >>>> >>>> >>>> >>>> When I use the function call_control( ) of the call_control >>>> module, it >>>> >>>> automatically engages mediaproxy if it finds the mediaproxy module >>>> loaded. >>>> >>>> If the mediaproxy module is not loaded, call_control doesn't even >>>> try to >>>> >>>> engage it. >>>> >>>> >>>> >>>> I need mediaproxy for NAT traversal in some cases, but don't want >>>> it to >>>> >>>> be engaged on every call. >>>> >>>> >>>> >>>> How can I disable this behavior? >>>> >>>> >>>> >>>> Thanks >>>> >>>> Reda >>>> >>>> >>>> >>>> _______________________________________________ >>>> >>>> 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 >>>> >>>> >>>> >>>> >>>> >>> _______________________________________________ >>>> >>> 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 >>>> >>> >>>> >>> >>>> >>> _______________________________________________ >>>> >>> 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 >>>> >>> >>>> >>> >>>> >> >>>> >> _______________________________________________ >>>> >> 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 >>>> >> >>>> >> >>>> > _______________________________________________ >>>> > 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 >>>> > >>>> >>>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>> >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda

2 4

obtaining htable array index
by Asgaroth 28 Feb '12

28 Feb '12

Hi All, Is it possible to asertain the array index of an htable entry based on its value? For example, say I have an htable called foo with the following entries: Entry:: 11 testing[0]:: hello-2 testing[1]:: 1 testing::size:: 2 is it possible to: [a] extract the array index based on a regex of the value, for example, something like value count: shtcv(foo=>hello-*) [b] if [a] is possible, how would I extract that value (from the associated array index returned in [a] above) from the htable using sht()? Is the above possible, or am I using the wrong tool for the job? I did have a look at the htable module docs, but i didnt see anything that stood out to me showing that i could achieve the above. I saw an shtval exported function, but i dont see it documented in the pseudo-var wiki page. I then tried it in the routing script but got an error message to the affect that it couldnt find the PV. I assume its not exported. All comments welcome and appreciated? Thanks

2 4

Dropping 183 and sending 180
by Fabian Borot 27 Feb '12

27 Feb '12

Hello We need to implement this scenario: client -> proxy --> provider The provider sends 183 with SDP and we need to drop it and send 180 instead. I managed to do it like this: 1- on_reply block, I catch the 183 if it comes, call a route block named "DROP_180" then drop it: if ($rs == "183") { route(DROP_180); drop(); } 2- on the new route block "DROP_180" I use the command send_reply(180, "Ringing") route[DROP_180] { xlog("L_INFO","mylog: On Route DROP_180."); send_reply(180,"Ringing"); } I had to do it on a separate block because the send_reply command could not be used on the same on_reply block, also I had to send the 180 first before the 183 is dropped because the drop() apparently stops the script execution and the 180 was not sent The idea works fine, BUT the To tag on the locally generated 180 is not the same as the one on the original 183 message. After the progress (183 message), it comes the 200 OK, which we forward downstream unaltered. Now, the client seems to ignore the 200 OK, (there is no ACK), we think that since the To tag on the original 183 and the 200 OK does not match the one on the 180 that is why the customer is processing the 200 OK. Any ideas here? txs a lot

3 4

XMLOPS xpath issues
by Olle E. Johansson 27 Feb '12

27 Feb '12

Friends, I'm having issues with the xpath support. If I run Daniel's example in the XMLOPS, xpath works as documented. But if I take the body of a PUBLISH from the and run xpath, I don't get expected results. Now, I'm no XPATH guru so I may be totally off the markup here... Here's the test script: $xml(pub=>doc) = $rb; $var(contact)=$xml(pub=>xpath:registration/contact@event); xlog("--- Publish contact $var(contact) \n"); xlog("--- Publish aor $xml(pub=>xpath:registration@aor) \n"); xlog("--- Publish aor: $xml(pub=>xpath://@aor) \n"); xlog("--- Publish reg state: $xml(pub=>xpath://reginfo@state) \n"); xlog("--- Publish uri: $xml(pub=>xpath:contact/uri/text()) \n"); I've tested with various paths, orginating from /reginfo or just picking an attribute. Here's the XML from Carsten's PUA_REGINFO module: <?xml version="1.0"?> <reginfo xmlns="urn:ietf:params:xml:ns:reginfo" version="0" state="full"> <registration aor="sip:oej@testnamn.se" id="0x80590a590" state="terminated"/> </reginfo> I wonder if it's something with using the request body that doesn't parse properly in the XMLOPS module? Error messages are a series of "XPath error : Invalid expression" /O

4 6

Using Voicemail strange behavior - git head 3.3
by David 27 Feb '12

27 Feb '12

I was testing the head version in GIT; I ran across a strange issue. The new config includes a "WITH_VOICEMAIL" definition I am not sure exactly how it is designed to work, but by default if the "usrloc" fails it creates a branch and goes to voicemail That part work correctly. It seems however if a phone is registered and the you get a busy, the failure route creates a branch but also retries the original RURI but I dont see a fork being created for the new branch to the voicemail server. Seems to just retry the original RURI. I am using the default config from the kamailio. repo. Any ideas. Thanks.

1 0

change DBURL without restarting kamailio?
by Uri Shacked 27 Feb '12

27 Feb '12

Hi, Is there a way to set a DBURL as a parametes that i can change in run time without restarting kamailio? It is usfull if i have a backup data base and i dont want to use mysql clustering... Cheers, Uri

2 1

CSEQ problem with SPA504G
by David 27 Feb '12

27 Feb '12

Hello, I am trying to connect to a kamailio server using a Cisco SPA504G phone using TCP instead of TCP. When in UDP, it works fine, but in TCP I get errors relating to CSEQ. I included the trace taking with ngrep after the email. I don't understand the CSeq error. I had a look at RFC3261, it says that the CSEQ number has to increase and that's what it has done. Can anyone tell me where the errorlies ? I tried changeing the check sanity to not check that CSeq is valid, but it screws up authentication as it mis detects the method type. Please advise, David interface: eth0 (SERVERIP/255.255.255.128) filter: (ip or ip6) and ( port 5060 and host 206.248.130.233 #### T 206.248.130.233:5075 -> SERVERIP:5060 [AP] REGISTER sip:SERVERNAME.TLD SIP/2.0. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-b03863b5;rport. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50446 REGISTER. Max-Forwards: 70. Contact: "104 Zone T" <sip:testzone.104@192.168.1.225:5075;transport=tcp>;expires=3600. Warning: 399 spa "STUN Server Not Reachable". User-Agent: Cisco/SPA504G-7.4.9c. Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE. Supported: replaces. . ## T SERVERIP:5060 -> 206.248.130.233:5075 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 192.168.1.225:5075;received=206.248.130.233;branch=z9hG4bK-b03863b5;rport=5075. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=a889ade7c63805944f01d11bb71a916b.6a29. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50446 REGISTER. WWW-Authenticate: Digest realm="SERVERNAME.TLD", nonce="4f4b87c31ea308c5fdf54c90cf2b30a400fa97a7". Server: OmniVigil 5.0a. Content-Length: 0. . ## T 206.248.130.233:5075 -> SERVERIP:5060 [AP] REGISTER sip:SERVERNAME.TLD SIP/2.0. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-dcfc09e;rport. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50447 REGISTER. Max-Forwards: 70. Authorization: Digest username="testzone.104",realm="SERVERNAME.TLD",nonce="4f4b87c31ea308c5fdf54c90cf2b30a400fa97a7",uri="sip:SERVERNAME.TLD",algorithm=MD5,response="77ddf2a1baafd33ed0c675fdc074bc03". Contact: "104 Zone T" <sip:testzone.104@206.248.130.233:5075;transport=tcp>;expires=3600. Warning: 399 spa "STUN Server Not Reachable". User-Agent: Cisco/SPA504G-7.4.9c. Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE. Supported: replaces. . ## T SERVERIP:5060 -> 206.248.130.233:5075 [AP] SIP/2.0 400 CSeq method does not match request method. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-dcfc09e;rport=5075;received=206.248.130.233. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=1feca66ab82b1f32a901c4cd8a4b7a05.e8e6. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50447 REGISTER. Server: kamailio (3.1.5 (x86_64/linux)). Content-Length: 0. .

2 1

How to do accounting for Kamailio-generated BYEs?
by Yufei Tao 27 Feb '12

27 Feb '12

Hi I've got a mediaproxy that detects no media, and sends dlg_end_dlg to Kamailio via mi_datagram socket. Then Kamailio sends BYE to both calling parties. It all works fine. But my question is, how to do accounting for these Kamailio-generated BYEs? They don't appear in the acc table. I'm using Kamailio 3.1.5. Thank you very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

2 1

CallControl and MediaProxy
by Reda Aouad 27 Feb '12

27 Feb '12

Hi, When I use the function call_control( ) of the call_control module, it automatically engages mediaproxy if it finds the mediaproxy module loaded. If the mediaproxy module is not loaded, call_control doesn't even try to engage it. I need mediaproxy for NAT traversal in some cases, but don't want it to be engaged on every call. How can I disable this behavior? Thanks Reda

4 12

Social networking event, London, March 6, 2012
by Daniel-Constantin Mierla 27 Feb '12

27 Feb '12

Hello, we are already several folks planing to go out after the first day of UC Expo and continue discussions about latest news of Kamailio/SER project and VoIP/Telecomunications. If you want to join, drop me a message to send you the details of the location. Looking forward to meeting many of you next week, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] How to do request URI rewrites using database tables
by Javier Gallart 27 Feb '12

27 Feb '12

Hello Ben dialplan module should help you: http://kamailio.org/docs/modules/3.2.x/modules/dialplan.html Regards Javi > ------------------------------ > > Message: 5 > Date: Fri, 17 Feb 2012 12:49:04 +1300 > From: Ben WIlliams <benwilliams(a)joobworld.com> > Subject: Re: [SR-Users] How to do request URI rewrites using database > tables? > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CANfYGQkeK3i=vBbdox+ekGeeKmjWhhgarUJcAo3oV=rrO=Wwig(a)mail.gmail.com > > > Content-Type: text/plain; charset=ISO-8859-1 > > I've managed to get dbaliases and lcr to do most of this. But the lcr > prefix does not allow regular expressions. Is there any other module > that allows you to store in the database a regular expression rewrite > rule? > > On Thu, Feb 16, 2012 at 1:21 PM, Ben WIlliams <benwilliams(a)joobworld.com> > wrote: > > Hi, can someone please recommend the most appropriate modules to > > rewrite R-URIs based on a database lookup table? I've read the > > documentation for lcr and carrierroute but not sure if they can do > > this. > > > > In most cases it will be a simple R-URI rewrite but I also need to > > rewrite based on From user. > > > > ie > > > > R-URI match ? ? ? ? ? ? From match ? ? ? ? ? ? ? ? ?new R-URI > > =========== ? ? ? ? ? ? ========== ? ? ? ? ? ? ? ? ?========= > > *97(a)example.com ? ? ? ? 6[0-9](a)example.com ? ? ? ? ?*97(a)10.0.0.1 > > *97(a)example.com ? ? ? ? 7[0-9](a)example.com ? ? ? ? ?*97(a)10.0.0.2 > > 6[0-9](a)example.com ? ? ?match any ? ? ? ? ? ? ? ? ? rewrite domain to > 10.0.0.1 > > abc(a)example.com ? ? ? ? match any ? ? ? ? ? ? ? ? ? def(a)example.com > > > > Thanks > > Ben > > > >

2 1

Have-my-domain! - host your own SIP domain at iptel.org
by SuperCel.mx.tc 26 Feb '12

26 Feb '12

Sirs: We cannot get our domain to configure to the iptel SIP Service (Have-my-domain! - host your own SIP domain at iptel.org). I have attached a copy of how we've tried setting the DNS, but we cannot find anywhere that gives us info on the "SIP" and/or "protocol UDP". Can you assist us so that we can use our own domain (http://supercel.tel) so that we can send SIP calls across this domain? Sincerely, Frank, CTO SuperCel.tel

1 0

Have-my-domain! - host your own SIP domain at iptel.org
by SuperCel.mx.tc 26 Feb '12

26 Feb '12

Sirs: We cannot get our domain to configure to the iptel SIP Service. I have attached a copy of how we've tried setting the DNS, but we cannot find anywhere that gives us info on the "SIP" and/or "protocol UDP". Can you assist us so that we can use our own domain (http://supercel.tel) so that we can send SIP calls across this domain? Sincerely, Frank Riley, CTO SuperCel.tel

1 0

Kamailio segmentation fault
by Fabian Borot 24 Feb '12

24 Feb '12

HelloOur kamailio proxy (version 1.5.2 ) crashed today due to a segmentation fault:Feb 22 16:52:10 AW-PROXY-01 kernel: kamailio[3667]: segfault at 000000000000000b rip 00002ab2b47bf93f rsp 00007ffff62dd270 error 6Apparently due to lack of private memory:[root@AW-PROXY-01 log]# cat proxy.log | grep ERROR | grep dbFeb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_convert_rows: no private memory left Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_convert_result: converting rows failed Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_store_result: failed to convert result Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:core:db_do_query: error while storing result Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_fetch_result: no private memory left Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:htable:ht_db_load_table: Error while fetching result I was hoping to see a core file somewhere but I haven't found one yet. I did not have the line "disable_core_dump=no" set on the config file because supposedly is set to no by default: >From the online guide for my version: //////////////////// Disable_core_dumpCan be 'yes' or 'no'. By default core dump limits are set to unlimited or a high enough value. Set this config variable to 'yes' to disable core dump-ing (will set core limits to 0).Default value is 'no'. ///////////////////How can I make sure that a core file is created after a situation like this? Where will it be created? How can I check the status of the private memory? I use the "kamctl fifo get_statistics all" but I don't know which one relates to "private memory": Example:shmem:total_size = 2147483648 shmem:used_size = 16860928 shmem:real_used_size = 17033968 shmem:max_used_size = 19174528 shmem:free_size = 2130449680 shmem:fragments = 1850I know that with so little information is hard to diagnose, but if anybody has an idea as to what else I can do I greatly appreciate it. Thank you in advancefborot

3 7

avp_delete problem [kamailio 1.5]
by Sven Knoblich 24 Feb '12

24 Feb '12

Hello all, i am currently confused by using the avpops function avp_delete. When i run avp_delete without the flag-value \g only the last value will be unset (like an undo). Is this the wanted behaviour? EXAMPLE: $avp(test) = "test" if( $avp(test)){xlog("L_NOTICE","result1:$avp(test)\n");} $avp(test) = "nooo"; if( $avp(test)){xlog("L_NOTICE","result2:$avp(test)\n");} avp_delete("$avp(test)"); if( $avp(test)){xlog("L_NOTICE","result3:$avp(test)\n");} RESULT: result1:test result2:nooo result3:test Could anybody helps me to understand that? thanks in advance, Sven

4 5

Adding parameter to headers
by Uri Shacked 24 Feb '12

24 Feb '12

Hi, I need to add a parameter to the contact header. suppose to look like: Contact: <sip:anonymous@domain:port;param1=XXX;param2=YYY> How do I add the parameters? Should i concatenate the $ct with the parameters? Or is there a better way? BR, Uri

2 1

fix_nated_sdp issue
by Ric Marques 24 Feb '12

24 Feb '12

Greetings, I'm not sure if I found a bug, or if I just have something completely misconfigured... I'm a total newb with Kamailio, working on a proof of concept design. Here's my configuration: provider -> nat firewall -> kamailio/rtpproxy -> asterisk For outbound calls from a phone registered to asterisk via kamailio, I'm trying to use fix_nated_sdp("2", "10.50.50.8") to rewrite the media ip address to resolve my audio issues, where 10.50.50.8 is the address outside my firewall. What I'm running into is the 'c=' line doesn't get re-written properly... it inserts the specified address in front of the existing address, and I end up with the following line in my INVITE: c=IN IP4 10.50.50.810.0.10.10 I have the fix_nated_sdp command under route[sipout], because I only want to use it on calls being sent outside the nat firewall. Here's the sip invite without the 'fix_nated_sdp' command: -------------------------------------------------------------------------------------------------------------- INVITE sip:19165551212@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes> Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060 Max-Forwards: 69 From: "1009" <sip:1009@10.0.10.11>;tag=as5498b77e To: <sip:19165551212@xxx.xxx.xxx.xxx> Contact: <sip:1009@10.0.10.11:5060> Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:06:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 309 P-hint: outbound v=0 o=root 604360056 604360056 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.0.10.10 t=0 0 m=audio 9702 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------- Here's the sip invite with the 'fix_nated_sdp' command: -------------------------------------------------------------------------------------------------------------- INVITE sip:19167828326@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes> Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK20d28324;rport=5060 Max-Forwards: 69 From: "1009" <sip:1009@10.0.10.11>;tag=as49e00c81 To: <sip:19167828326@xxx.xxx.xxx.xxx> Contact: <sip:1009@10.0.10.11:5060> Call-ID: 4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:18:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 347 P-hint: outbound v=0 o=root 1009117068 1009117068 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.50.50.8.10.0.10.10 t=0 0 m=audio 13540 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=oldmediaip:10.0.10.11 a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------- Is this a bug, or is it likely I have something else screwed up? Thank you in advance for your assistance - this list is an incredible resource! -Ric

2 3

Require Certificate
by Bruno Bresciani 22 Feb '12

22 Feb '12

Hi All, Does it possible on tls module require certificates only some hosts? My kamailio comunicate with several UA clients (Softphones and IP phones) and one SIP gateway. I want require certificate only gateway SIP, it's possible? The parameter *modparam("tls", "require_certificate", 1)* require certificate of all clients... Best Regards

2 4

Topoh hiding prefixes
by Efelin Novak 22 Feb '12

22 Feb '12

Hi folks, I have a topology like this: UA -> kamailio -> PSTN GW. When I'm placing calls to a PSTN I have to append prefixes so the PSTN GW knows how to route the call. So the request-uri is <prefix><number>@<gateway_ip> e.g. 999123456(a)192.168.1.1. When the UA calls to the GW a "200 OK" message contains a contact header with 999123456(a)192.168.1.1. This is a problem because I don't want my customers to see gateway prefixes. The Topoh module only hides IP addresses or the domain part. Is there any way to remove prefixes. This document (http://www.kamailio.org/events/2011-Cluecon/DCM-kamailio-security.pdf) stated "encoding IP and prefixes can be set via parameters". Can I remove prefixes from the contact in any other way than some contact header substitutions? Thanks Efelin

3 9

CallControl and MediaProxy
by Reda Aouad 22 Feb '12

22 Feb '12

Hi, I noticed that calling call_control( ) function forces engage_media_proxy( ). Is there a way to disable this behavior ? Thanks Reda

1 0

Kamailio v3.2 problem with dialog module.
by Phillman25 Kyriacou 21 Feb '12

21 Feb '12

Dear List After upgrading from V3.1.5 to V3.2. Kamailio crashes with the below error messages: Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core> [db.c:392]: invalid number of rows received: 4, dialog_vars Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core> [db.c:419]: querying version for table dialog_vars Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: dialog [dlg_db_handler.c:153]: error during dialog-vars version check. Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: dialog [dialog.c:646]: failed to initialize the DB support Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module dialog (/usr/local/lib/kamailio/modules_k/dialog.so) My dialog paramaters are as follows: # ----------------- DIALOG MODULE PARAMETERS----------------------------------# modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag", 4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "default_timeout", 21600) modparam("dialog", "dlg_extra_hdrs", "NULL") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 60) modparam("dialog", "db_fetch_rows", 500) modparam("dialog", "table_name", "dialog") modparam("dialog", "from_uri_column", "from_uri") modparam("dialog", "from_tag_column", "from_tag") modparam("dialog", "to_uri_column", "to_uri") modparam("dialog", "to_tag_column", "to_tag") modparam("dialog", "h_id_column", "hash_id") modparam("dialog", "h_entry_column", "hash_entry") modparam("dialog", "state_column", "state") modparam("dialog", "start_time_column", "start_time") modparam("dialog", "timeout_column", "timeout") modparam("dialog", "sflags_column", "sflags") modparam("dialog", "bridge_controller", "sip:controller@kamailio.org") modparam("dialog", "default_timeout", 7200) i realised that when i set the "modparam("dialog", "db_mode", 0)" kamailio starts fine, seems to be a problem connecting to the database. Does anyone know what is wrong? Thanking you in advance Phillip

2 3

UAC canceling a specific early dialog with BYE
by Andrew Pogrebennyk 21 Feb '12

21 Feb '12

Hello, we have a Kamailio proxy which gets the call from PSTN gw and does some call forwarding (serial forking) to several destinations through our sbc The call flow I am looking at is: - Kamailio sends INVITE to branch_1. - branch_1 sends 180 with to-tag*, proxy relays it to the gw * 180 meets the requirements for dialog creating 18x responses in sections 12.1, 12.1.1 because it contains to-tag, contact and mirrored record-route. - After some seconds Kamailio sends a CANCEL to branch_1. - And sends the INVITE to branch_2. - branch_1 replies 200 for the CANCEL and 487 for the INVITE. - branch_2 replies 180 and 200 for the INVITE. - When PSTN gw receives that it sees it still needs to cancel other early dialog established by 180 from branch_1. - The PSTN gw sends a BYE with to-tag of branch_1 to cancel this specific early dialog. SIP allows early dialogs to individually released while other dialogs continue, as written in RFC, section 15: "The BYE request is used to terminate a specific session or attempted session. In this case, the specific session is the one with the peer UA on the other side of the dialog. (...). The caller’s UA MAY send a BYE for either confirmed or early dialogs, and the callee’s UA MAY send a BYE on confirmed dialogs, but MUST NOT send a BYE on early dialogs." The BYE follows the loose routing path, proxy gets 481 from the sbc and forwards that response back to PSTN gw, which somehow breaks it. AFAICS it's not specified in RFC what should the behavior look like when getting both a 200 and error-class response for the INVITE (quotes are most welcome!). IMO it would be more correct to absorb BYE in proxy but I see a big problem here: branch_2 can even ring for 5 minutes and it's not feasible for proxy to have a wt-timer that long. Also it's not possible to inform the gw that early dialog has cleared as soon as we receive 200/487 from branch_1. So I'm not sure which party is at fault and if we can workaround that somehow in the Kamailio. Any thoughts?

1 1

Re: [SR-Users] Kamailio ims / rtpproxy
by trojak＠st.fri.uniza.sk 21 Feb '12

21 Feb '12

Hi All! We did litttle research: - both behind the NAT = all OK = straeming over the rtpproxy - caller behind NAT (other public) = all BAD = no change in SDP (Connection Address) - rtp stream sending on private addres - calle behind NAT (other public) = all BAD = SDP has changed but only private IP address... public IP unchanged = so rtpproxy do not ralay one way stream Can somebody (especialy IMS) please help us Thanks Best regards, Lubo > Hi Daniel, > > thank you for answer. I think that parts of kamailio default config > will not working with pcscf. (e.g. there is on pcscf no database - > it has hss) > > It partialy works with pcscf config -> it rewrites Connecition > Infromation IP in SDP: > - sometimes with IP of rtpproxy (more often when both clietnts are > behind NAT) > - sometimes with public IP of NATed client (we are not sure why) > - or unchanged (when one client is behind NAT and other is public) > > but this works only with Monster client, with Boghe IP is unchanged > (shouldn't it be client independent ?). > > Maybe someone from IMS help us. > > Thanks all for any advice. > > Best regards, > > Lubo > > Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Hello, >> >> On 1/23/12 10:43 AM, trojak(a)st.fri.uniza.sk wrote: >>> >>> Hi, >>> >>> I am trying to make kamailio PCSCF working with rtpproxy (for NAT >>> traversal). >>> I would like to ask, if there is any tutorial for this, or if you >>> can give me any advice. >> Maybe IMS guys can give other hints, what is on public so far was >> summarized at: >> * http://www.kamailio.org/w/2011/01/ims-extensions-available-for-testing/ >> >>> I also have installed rtpproxy package; #!define WITH_NAT, and >>> defined rtpproxy port in config file... >>> But this is not working. I noticed, that in default config file of >>> pcscf is nothing in routing logic with rtpproxy (or it is >>> unnecessary ?), so I try to copy route[RELAY] route[NAT] >>> route[RTPPROXY] modules from kamailio (no pcscf) config file... >>> but also with no success. >>> >>> Can you help me please ? What I am doing wrong ? Do you have any >>> configuration working with rtpproxy ? >> Be sure you can the nat handling routes for INVITES behind nat. You >> can follow the logic in the default config file and adapt it for >> you pcscf config. In summary should be like: >> - invite comes in, test if it is from caller behind nat >> - lookup location returns a branch flag if the calee is behind nat >> (you have to set that branch flag for register request coming from >> users) >> - if caller or callee are behind nat, engage rtpproxy for that call >> and set a onreply route to update sdp for 18x/200 replies >> - default config adds a Route header parameter for nated calls to >> detect easily the natted situation for re-invites or bye >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.com >> http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Dialogs not removed from memory, and occasionally persistent in DB as well
by Øyvind Kolbu 20 Feb '12

20 Feb '12

Hi, we use the dialog-module to keep track of concurrent calls, and then set a treshold for simultanious calls. After we upgraded from 1.5 to 3.2, we've been having a lot of dialogs which are stuck in both the database and memory. kamailio -V version: kamailio 3.2.1 (i386/linux) 918035-dirty flags: STATS: Off, USE_IPV6, 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: 918035 -dirty compiled on 10:00:10 Dec 20 2011 with gcc 4.1.2 Our configuration: modparam("dialog", "dlg_flag", 3) modparam("dialog", "default_timeout", 21600) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_url", VOIP_DATA4) modparam("dialog", "db_mode", 1) #!ifdef VOIP1 modparam("dialog", "table_name", "dialog1") modparam("dialog", "vars_table_name", "dialog_vars1") #!endif #!ifdef VOIP2 modparam("dialog", "table_name", "dialog2") modparam("dialog", "vars_table_name", "dialog_vars2") #!endif modparam("dialog", "profiles_with_value", "busy") To Kamailio instances, voip1 and voip2, share the same config, but define VOIP1 and VOIP2, respectively. Then in the main route: if !allow_trusted() { route(AUTH); } if (is_method("INVITE")) { setflag(3); dlg_manage(); if ($avp(s:f_uid) != $null) { set_dlg_profile("busy","$avp(s:f_uid)"); get_profile_size("busy", "$avp(s:f_uid)", "$var(dlg_busy)"); xlog("L_INFO", "BUSY: f_uid: $avp(s:f_uid), dlg_busy: $var(dlg_busy)\n"); } } $avp(s:f_uid) is set in route(AUTH), so it will catch calls originating from one of our users. Then later in the callforward route: # Testing dialog $var(dlg_busy) = 0; $var(dlg_busy1) = 0; $var(dlg_busy2) = 0; #!ifdef VOIP1 get_profile_size("busy", "$rU", "$var(dlg_busy1)"); sql_query("data4","select id from dialog2 where caller_contact like 'sip:$rU@%' or callee_contact like 'sip:$rU@%'", "dialog"); $var(dlg_busy2) = $dbr(dialog=>rows); #!endif #!ifdef VOIP2 get_profile_size("busy", "$rU", "$var(dlg_busy2)"); sql_query("data4","select id from dialog1 where caller_contact like 'sip:$rU@%' or callee_contact like 'sip:$rU@%'", "dialog"); $var(dlg_busy1) = $dbr(dialog=>rows); #!endif sql_result_free("dialog"); $var(dlg_busy) = $var(dlg_busy1) + $var(dlg_busy2); And then check if $var(dlg_busy) is above a users treshold. Finally when routing to a local user we mark it as busy: set_dlg_profile("busy","$rU"); As of writing I have 5 entries in the dialog1-table, all valid, but: # kamctl fifo dlg_list | grep -c "^dialog" 18 # kamctl fifo dlg_list | grep state | sort | uniq -c 13 state:: 1 5 state:: 4 on the other hand in the busy profile, which is the one we use: # kamctl fifo profile_list_dlgs busy | grep -c "^dialog" 11 # kamctl fifo profile_list_dlgs busy | grep state | sort | uniq -c 6 state:: 1 5 state:: 4 It seems as this happens when, for some reason, kamailio is trying to parse duplicate INVITEs. From the logs: # grep "^Jan 17" /var/log/messages | grep BUSY | grep -v "dlg_busy: 1" | cut -d\ -f8- |sort | uniq -c 1 BUSY: f_uid: 2367449, dlg_busy: 2 3 BUSY: f_uid: 2367453, dlg_busy: 2 1 BUSY: f_uid: 2367453, dlg_busy: 3 1 BUSY: f_uid: 2367453, dlg_busy: 4 1 BUSY: f_uid: 2574912, dlg_busy: 2 1 BUSY: f_uid: 2582270, dlg_busy: 2 1 BUSY: f_uid: ahmedma, dlg_busy: 2 1 BUSY: f_uid: josephim, dlg_busy: 2 1 BUSY: f_uid: perla, dlg_busy: 2 1 BUSY: f_uid: rosalinm, dlg_busy: 2 1 BUSY: f_uid: suresha, dlg_busy: 2 1 BUSY: f_uid: sveinb, dlg_busy: 2 I've full debugging logs available, if needed. -- Øyvind Kolbu

5 13

How to deal with diversion loops.
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, the maxfwd will not help me prevent the loop on the first round, it would only stop it a little sooner the X times. no? the dlg_var is interesting, but i didn't understand how to use it exactly. how do i match two dialogs? is the spiral detection on dialog module helps? thanks, Uri >Hi Uri, >for that you can easily count the forwards by using the maxfwd module >(mf_process_maxfwd_deader <your max>) or you can store all your spiraled >calls into a dialog-variable like >$dlg_var(forward-chain)=<B>;<C>|<C>;<B> and check whether the next >number is already stored or not. Bye Sven Am 20.02.2012 13:20, schrieb Uri Shacked: >* *>* Hi, *>* *>* The service i built in kamailio is simple - a caller dials an Access *>* number, my gateway send it to kamailio. *>* *>* kamailio finds the shadow number for this Access number and send an *>* invite to the same gateway with the shadow number as destination *>* (state full proxy). *>* *>* In kamailio i change only the RURI and the “To” and “From” stays *>* untouched. *>* *>* Now, when a client performs “diversion”, I can see that I can find *>* myself in an unwanted long loop…. *>* *>* Example : *>* *>* A calls B *>* *>* B diverted the call unconditionally to C. *>* *>* C is an Access Number that its shadow number is D. *>* *>* D diverted the call to B……. *>* *>* Well, that is a bad loop to have! *>* *>* It can be longer or shorter, but the idea I understood (I think). *>* *>* I thought about diversion check, counters, dialog “To” and “From” *>* check, etc…. *>* *>* Anyone here dealt with it? *>* *>* What will be the best way to check and see if the situation accrues? *>* *>* Thanks, *>* *>* Uri *>* *

1 0

integration kamailio-sems
by Uri Shacked 20 Feb '12

20 Feb '12

Thanks, did you tried to develop an IVR system based on SEMS? will it be better? harder? more flexible than Asterisk?

1 0

Kamailio as SBC
by Martin Schiøtz 20 Feb '12

20 Feb '12

Hi I'm new to be working with SIP. We have a SIP server and it handles SIP well but it does not handle far-end-nat. Can the Kamailio server be used as a SBC (Session Border Controller) to handle far-end-nat? Just forward all SIP packages to a SIP server and fix the far-end-nat problem? Cheers! Martin

2 2

How to deal with diversion loops.
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, The service i built in kamailio is simple - a caller dials an Access number, my gateway send it to kamailio. kamailio finds the shadow number for this Access number and send an invite to the same gateway with the shadow number as destination (state full proxy). In kamailio i change only the RURI and the “To” and “From” stays untouched. Now, when a client performs “diversion”, I can see that I can find myself in an unwanted long loop…. Example : A calls B B diverted the call unconditionally to C. C is an Access Number that its shadow number is D. D diverted the call to B……. Well, that is a bad loop to have! It can be longer or shorter, but the idea I understood (I think). I thought about diversion check, counters, dialog “To” and “From” check, etc…. Anyone here dealt with it? What will be the best way to check and see if the situation accrues? Thanks, Uri

2 1

integration kamailio-sems
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, I started reading a little on SEMS. Tried to look for the way Kamailio works has the frontend sip server and SEMS will be the media server (IVR, announcments etc). I didn't understand which kamailio module should i work with in order to open sockets with SEMS. Or do i get it all wrong? Uri

2 1

$time(year) gives 1900
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, I am using the $time(name) pv. On my lab machine (centos 5.7) i get the right value (2012). But, on my other machine (centos 6.2) i get 1900. what am i missing here?

2 2

Mtree VS Htable getting huge amount of data from DB not on local machine
by Uri Shacked 20 Feb '12

20 Feb '12

> > Hi, > > untill today i used the mtree module to get some data from a remote DB and > load it into memory. > the table in the data base has 5 million rows. prefix is a number and > value is some string. > > i moved to htable because i need the prefix exact match and not longest.... > > now, after 5-10 sec, i get the following error (that i never got in mtree) > - "lost connection to mysql during query" and kamailio does not start. > > any ideas how to solve it? > > cheers, > > Uri >

2 2

Registration replication
by Spencer Thomason 19 Feb '12

19 Feb '12

Hello, I'm currently working on a HA setup with geographically separated proxies. Each proxy is in a different data center with an MPLS link between them. The proxies access a locally connected PostgreSQL database that is replicated from a master in the primary data center. Note that the replicated dbs are read only. Everything works great for read only queries but registrations seem to pose a problem as they must use the primary db directly to deal with the writes. If I set userloc to db mode 2 the secondary proxies wipe out the entries in the database. If I use db mode 3 everything works great but all of the secondary proxies must use the primary database directly and a network link failure will cause a service disruption. Is there any way I can use separate read and write databases without p_userloc? Is there a better way to replicate registrations maintaining availability and scalability? And advice is appreciated. Thanks, Spencer Connected by DROID on Verizon Wireless

2 1

Segmentation Fault encountered in pv.so
by Simpson Chua 19 Feb '12

19 Feb '12

Hi, Since I am getting errors in from the usrloc module, I am trying another way to parse out the information I need using pv.so. I'm encountering similar segfaults issues. Here are snippets of my configs: modparam("sipcapture", "db_url", "mysql://xxx:xxx@localhost/test_db") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_socket_listen", "127.0.0.1:5060-9000") modparam("sipcapture", "raw_interface", "eth2") modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "promiscious_on", 1) route { if (method == "REGISTER") { if (avp_check("$si","eq/10.xxx.xxx.xxx")) { #save("location", "0x04"); # This currently results in segfault and is being addressed by Alexandr sip_capture(); xlog("The received IP is $Ri\n"); # Just to test if I am able to parse the received IP from the Via headers correctly drop; } drop; } } Feb 17 13:41:02 ubuntu kernel: [251382.760104] kamailio[9414]: segfault at 31 ip 00007fb4ef077611 sp 00007fff0c8a6468 error 4 in pv.so[7fb4ef06a000+39000] Core was generated by `kamailio -w /home/core'. Program terminated with signal 11, Segmentation fault. #0 pv_get_rcvip (msg=0x7fb4efdd7598, param=0x7fb4efdcef88, res=0x7fff0c8a64b0) at pv_core.c:647 647 || msg->rcv.bind_address->address_str.s==NULL) (gdb) bt #0 pv_get_rcvip (msg=0x7fb4efdd7598, param=0x7fb4efdcef88, res=0x7fff0c8a64b0) at pv_core.c:647 #1 0x000000000049b5ad in pv_get_spec_value (msg=0x7fb4efdd7598, sp=0x7fb4efdcef70, value=0x7fff0c8a64b0) at pvapi.c:1180 #2 0x000000000049b71d in pv_printf (msg=0x7fb4efdd7598, list=<optimized out>, buf=<optimized out>, len=0x7fff0c8a6558) at pvapi.c:1239 #3 0x00007fb4ed0d442e in xlog_helper (msg=<optimized out>, xm=0x7fb4efdceef0, level=-1, line=0) at xlog.c:168 #4 0x000000000041babb in do_action (h=0x7fff0c8a73b0, a=0x7fb4efdd02a0, msg=0x7fb4efdd7598) at action.c:1122 #5 0x000000000041abe2 in run_actions (h=0x7fff0c8a73b0, a=0x7fb4efdd0080, msg=0x7fb4efdd7598) at action.c:1610 #6 0x000000000041c05b in do_action (h=0x7fff0c8a73b0, a=0x7fb4efdd10d0, msg=0x7fb4efdd7598) at action.c:1107 #7 0x000000000041abe2 in run_actions (h=0x7fff0c8a73b0, a=0x7fb4efdd10d0, msg=0x7fb4efdd7598) at action.c:1610 #8 0x000000000041c05b in do_action (h=0x7fff0c8a73b0, a=0x7fb4efdd1350, msg=0x7fb4efdd7598) at action.c:1107 #9 0x000000000041abe2 in run_actions (h=0x7fff0c8a73b0, a=0x7fb4efdd1350, msg=0x7fb4efdd7598) at action.c:1610 #10 0x0000000000422fa2 in run_top_route (a=0x7fb4efdd1350, msg=0x7fb4efdd7598, c=<optimized out>) at action.c:1683 #11 0x00000000004b061e in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:207 #12 0x00007fb4edb538b0 in raw_capture_rcv_loop (rsock=5, port1=5060, port2=8933, ipip=0) at sipcapture.c:1518 #13 0x00007fb4edb53e1f in init_rawsock_children () at sipcapture.c:572 #14 0x00007fb4edb54033 in child_init (rank=<optimized out>) at sipcapture.c:542 #15 0x0000000000507a0f in init_mod_child (m=0x7fb4efdcb2b0, rank=0) at sr_module.c:886 #16 0x0000000000507974 in init_mod_child (m=0x7fb4efdcb668, rank=0) at sr_module.c:883 #17 0x0000000000507974 in init_mod_child (m=0x7fb4efdcc178, rank=0) at sr_module.c:883 #18 0x0000000000507974 in init_mod_child (m=0x7fb4efdcc4b0, rank=0) at sr_module.c:883 #19 0x0000000000507974 in init_mod_child (m=0x7fb4efdccaf0, rank=0) at sr_module.c:883 #20 0x0000000000507974 in init_mod_child (m=0x7fb4efdcd030, rank=0) at sr_module.c:883 #21 0x0000000000476149 in main_loop () at main.c:1661 #22 0x000000000041a940 in main (argc=<optimized out>, argv=0x7fff0c8a7be8) at main.c:2475 (gdb) p *msg $1 = {id = 82, pid = 9418, 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 = 43, u = {request = {method = { s = 0x7fb4edd57d2a "REGISTER sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCal"..., len = 8}, uri = { s = 0x7fb4edd57d33 "sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 225"..., len = 24}, version = { s = 0x7fb4edd57d4c "SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 2257e150-b49dfd8d-c8d924b2@1"..., len = 7}, method_value = 32}, reply = {version = { s = 0x7fb4edd57d2a "REGISTER sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCal"..., len = 8}, status = { s = 0x7fb4edd57d33 "sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 225"..., len = 24}, reason = { s = 0x7fb4edd57d4c "SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 2257e150-b49dfd8d-c8d924b2@1"..., len = 7}, statuscode = 32}}}, via1 = 0x7fb4efdd7ef0, via2 = 0x7fb4efdd8218, headers = 0x7fb4efdd7e50, last_header = 0x7fb4efdd87c0, parsed_flag = 18446744073709551615, h_via1 = 0x7fb4efdd80d8, h_via2 = 0x7fb4efdd8178, callid = 0x7fb4efdd19b0, to = 0x7fb4efdd1910, cseq = 0x7fb4efdd7c80, from = 0x7fb4efdd7e50, contact = 0x7fb4efdd84a0, maxforwards = 0x7fb4efdd8680, route = 0x0, record_route = 0x0, content_type = 0x0, content_length = 0x7fb4efdd87c0, authorization = 0x0, expires = 0x7fb4efdd8720, proxy_auth = 0x0, supported = 0x0, require = 0x0, proxy_require = 0x0, unsupported = 0x0, allow = 0x0, event = 0x0, accept = 0x0, accept_language = 0x7fb4efdd85e0, organization = 0x0, priority = 0x0, subject = 0x0, user_agent = 0x7fb4efdd8540, server = 0x0, content_disposition = 0x0, accept_disposition = 0x0, diversion = 0x0, rpid = 0x0, refer_to = 0x0, session_expires = 0x0, min_se = 0x0, sipifmatch = 0x0, subscription_state = 0x0, date = 0x0, identity = 0x0, identity_info = 0x0, pai = 0x0, ppi = 0x0, path = 0x0, privacy = 0x0, body = 0x0, eoh = 0x7fb4edd58036 "12963462\", realm=\"proxyApp\", nonce=\"proxyAppXgyrmj82aTujp3sdBW\", uri=\"sip:abc.def.com\", response=\"039891b3f5999b90a645d3e808c2147c\", algorithm=MD5, cnonce=\"l7/u88HtGJCOAmM\", qop=auth, nc="..., unparsed = 0x7fb4edd58036 "12963462\", realm=\"proxyApp\", nonce=\"proxyAppXgyrmj82aTujp3sdBW\", uri=\"sip:abc.def.com\", response=\"039891b3f5999b90a645d3e808c2147c\", algorithm=MD5, cnonce=\"l7/u88HtGJCOAmM\", qop=auth, nc="..., rcv = {src_ip = {af = 2, len = 4, u = {addrl = {140411255757578, 4294967291}, addr32 = {184916746, 32692, 4294967291, 0}, addr16 = {39690, 2821, 32692, 0, 65531, 65535, 0, 0}, addr = "\n\233\005\v\264\177\000\000\373\377\377\377\000\000\000"}}, dst_ip = {af = 2, len = 4, u = {addrl = {140414021193994, 5273972}, addr32 = { 2950353162, 32692, 5273972, 0}, addr16 = {53514, 45018, 32692, 0, 31092, 80, 0, 0}, addr = "\n\321Ú¯\264\177\000\000tyP\000\000\000\000"}}, src_port = 5060, dst_port = 5060, proto_reserved1 = 32692, proto_reserved2 = -266587208, src_su = {s = {sa_family = 2, sa_data = "\023\304\n\233\005\v\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 184916746}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 184916746, 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 = 0x1, proto = 1 '\001'}, buf = 0x7fb4edd57d2a "REGISTER sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCal"..., len = 782, new_uri = {s = 0x0, len = 0}, dst_uri = {s = 0x0, len = 0}, parsed_uri_ok = 1, parsed_uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = { s = 0x7fb4edd57d37 "abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 2257e15"..., len = 20}, 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}, 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}}, 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}, 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}}, add_rm = 0x0, body_lumps = 0x0, reply_lump = 0x0, add_to_branch_s = '\000' <repeats 57 times>, add_to_branch_len = 0, hash_index = 0, msg_flags = 0, flags = 0, set_global_address = {s = 0x0, len = 0}, set_global_port = {s = 0x0, len = 0}, force_send_socket = 0x0, path_vec = {s = 0x0, len = 0}} (gdb) p *msg->rcv.bind_address Cannot access memory at address 0x1 (gdb) Thanks, Simpson ________________________________ From: "sr-users-request(a)lists.sip-router.org" <sr-users-request(a)lists.sip-router.org> To: sr-users(a)lists.sip-router.org Sent: Friday, February 17, 2012 7:34 AM Subject: sr-users Digest, Vol 81, Issue 52 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: ndb_redis module fails after a while (Daniel-Constantin Mierla) 2. Re: Kamailio Exits With: Segmentation Fault Error 4 in usrloc.so (Alexandr Dubovikov) ---------------------------------------------------------------------- Message: 1 Date: Fri, 17 Feb 2012 14:30:44 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] ndb_redis module fails after a while To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: Javier Gallart <jgallartm(a)gmail.com> Message-ID: <4F3E5684.3090203(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, right, very dummy paste position for resetting the redis context, use the attached patch or exchange lines 228 and 229. in redis_client.c Thanks, Daniel On 2/17/12 1:13 PM, Javier Gallart wrote: > Hi Daniel, Andrew > > I've just tested the patch and kamailio crashes when the redis server > is stopped. This is what I could get: > -From the logs: > Feb 17 06:33:32 r-gate-test ./kamailio[23812]: ALERT: <core> > [main.c:751]: child process 23819 exited by a signal 11 > Feb 17 06:33:32 r-gate-test ./kamailio[23812]: ALERT: <core> > [main.c:754]: core was generated > Feb 17 06:33:32 r-gate-test ./kamailio[23812]: INFO: <core> > [main.c:766]: INFO: terminating due to SIGCHLD > > -An the backtrace: > (...) > Core was generated by `./kamailio -f ../etc/kamailio/kamailio.cfg'. > Program terminated with signal 11, Segmentation fault. > #0 redisFree (c=0x0) at hiredis.c:817 > 817 if (c->fd > 0) > (gdb) bt > #0 redisFree (c=0x0) at hiredis.c:817 > #1 0x00007f726f035461 in redisc_reconnect_server > (rsrv=0x7f7271c93ac0) at redis_client.c:229 > #2 0x00007f726f037240 in redisc_exec (srv=<value optimized out>, > cmd=0x7fffe98c6090, argv1=<value optimized out>, argv2=<value > optimized out>, argv3=<value optimized out>, > res=<value optimized out>) at redis_client.c:298 > #3 0x00007f726f034f7d in w_redis_cmd3 (msg=0x7f7271d7b018, > ssrv=<value optimized out>, scmd=<value optimized out>, > sres=0x7f7271d74b58 "h\341\313qr\177") at ndb_redis_mod.c:156 > #4 0x0000000000417025 in do_action (h=0x7fffe98c6570, > a=0x7f7271cc5248, msg=<value optimized out>) at action.c:1134 > #5 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #6 0x000000000041e8d4 in run_actions_safe (h=0x7fffe98c7610, > a=0x7f7271c93781, msg=0x7f7271c93780) at action.c:1662 > #7 0x00000000004b731d in rval_get_int (h=0x7fffe98c7610, msg=0x0, > i=0x7fffe98c6bd8, rv=0x3, cache=0x4) at rvalue.c:920 > #8 0x00000000004bb87c in rval_expr_eval_int (h=0x7fffe98c7610, > msg=0x7f7271d7b018, res=0x7fffe98c6bd8, rve=0x7f7271cc6768) at > rvalue.c:1914 > #9 0x0000000000417c7c in do_action (h=0x7fffe98c7610, > a=0x7f7271ccb9d0, msg=<value optimized out>) at action.c:1092 > #10 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #11 0x0000000000417cd7 in do_action (h=0x7fffe98c7610, > a=0x7f7271ccbb10, msg=<value optimized out>) at action.c:1111 > #12 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #13 0x000000000041795e in do_action (h=0x7fffe98c7610, a=<value > optimized out>, msg=<value optimized out>) at action.c:732 > #14 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #15 0x000000000041e862 in run_top_route (a=0x7f7271c94888, > msg=0x7f7271d7b018, c=<value optimized out>) at action.c:1683 > #16 0x0000000000498f36 in receive_msg ( > buf=0x8bb100 "INVITE sip:34661574758@79.170.68.215:5060 > <http://sip:34661574758@79.170.68.215:5060> SIP/2.0\r\nVia: > SIP/2.0/UDP 79.170.68.214:5060;branch=z9hG4bK-9451-1-0\r\nFrom: > 34661574758 <sip:34661574758@79.170.68.214:5060 > <http://sip:34661574758@79.170.68.214:5060>>;tag=9451SIPpTag001\r\nTo: > sut <sip:"..., len=<value optimized out>, rcv_info=0x7fffe98c7910) at > receive.c:207 > #17 0x0000000000525987 in udp_rcv_loop () at udp_server.c:544 > #18 0x00000000004635f4 in main_loop () at main.c:1585 > #19 0x0000000000465e62 in main (argc=3, argv=0x7fffe98c7c08) at > main.c:2475 > > > Regards > > Javi > > On Fri, Feb 17, 2012 at 11:39 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > thanks for testing, indeed it was an extra declaration left over. > Can you try with the new patch attached? > > Cheers, > Daniel > > > On 2/17/12 11:10 AM, Andrew Pogrebennyk wrote: > > Hi Daniel, > > On 02/17/2012 10:47 AM, Daniel-Constantin Mierla wrote: > > I made a patch for server reconnect -- I had no access to > a computer > with redis lib installed for the moment, hopefully it > compiles. If you > can try and tell the result, it would be great, I can > commit then. > > I may be able to test this patch as well. Currently > compilations bails > out on attempt to redeclare redisc_reconnect_server function > parameter: > > CC (gcc) [M ndb_redis.so] ndb_redis_mod.o > CC (gcc) [M ndb_redis.so] redis_client.o > redis_client.c: In function 'redisc_reconnect_server': > redis_client.c:206:19: error: 'rsrv' redeclared as different > kind of symbol > redis_client.c:202:46: note: previous definition of 'rsrv' was > here > make[1]: *** [redis_client.o] Error 1 > make: *** [modules] Error 1 > > _______________________________________________ > 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://linkedin.com/in/miconda -- http://twitter.com/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 -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

avp_db_query - closing the connection
by graham 19 Feb '12

19 Feb '12

Hi all, I have been using more and more avps for per-call queries populated by custom mysql queries. I would like to know how the connections of mysql queries are being closed. I tried to locate a command to do such, but unable to locate. Do they close on their own, after a certain timeout period? I am running kamailio 1.5.5, notls. The proxies are getting busier and I want to keep all the resources in check ... Thanks, -graham

2 1

question on getting RTP stats
by Nelson Pereira 19 Feb '12

19 Feb '12

Is there a way to get asterisk to publish RTP stats in the bye message? I would like to parse theses to DB and get stats from them. If so, how? Thanks Np

1 0

Kamailio or VOS3000
by Abid Naseem 17 Feb '12

17 Feb '12

How kamailio is different from VOS3000 or Nextone or PORTASwitch when we are talking about wholesale routing? Sent from my iPhone

2 1

Re: [SR-Users] Need example of Kamailio.cfg for a port mirror setup
by Nelson Pereira 17 Feb '12

17 Feb '12

I have libpcap, it's in /usr/lib64 but it's libpcap0.9.8 >:/usr/local/src/captagent # find / |grep libpcap /usr/share/doc/packages/libpcap-devel /usr/share/doc/packages/libpcap-devel/pcap.xml /usr/share/doc/packages/libpcap-devel/pcap.html /usr/share/doc/packages/libpcap-devel/pcap.txt /usr/share/doc/packages/libpcap0 /usr/share/doc/packages/libpcap0/LICENSE /usr/share/doc/packages/libpcap0/TODO /usr/share/doc/packages/libpcap0/CREDITS /usr/share/doc/packages/libpcap0/README.linux /usr/share/doc/packages/libpcap0/README /usr/share/doc/packages/libpcap0/CHANGES /usr/lib64/libpcap.a /usr/lib64/libpcap.so.0.9.8 /usr/lib64/libpcap.so /usr/lib64/libpcap.so.0 On Fri, Feb 17, 2012 at 10:38 AM, Max Mühlbronner <mm(a)42com.com> wrote: > This is a multi-part message in MIME format. > --------------070003040807050803020008 > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > Content-Transfer-Encoding: 8bit > > Hi, > > you are missing libpcap0.8 which is a dependency for the captagent. > > > BR > > Max M. > > Am 17.02.2012 16:31, schrieb Nelson Pereira: > > ok, tried to do the make install, getting the bellow: > > aurast:/usr/local/src/captagent # /usr/local/bin/captagent > > /usr/local/bin/captagent: error while loading shared libraries: > > libpcap.so.0.8: cannot open shared object file: No such file or directory > > > > Do i need to compile this? > > > > Also just tried running the captagent: > > aurast:/usr/local/src/captagent # ./captagent > > ./captagent: error while loading shared libraries: libpcap.so.0.8: > > cannot open shared object file: No such file or directory > > > > What am I doing wrong? > > > > This is a SUSE SLES11 64bit server running asterisk 1.8.6 > > > > Thanks > > > > > > > > On Fri, Feb 17, 2012 at 10:24 AM, Nelson Pereira <kitkat0981(a)gmail.com > > <mailto:kitkat0981@gmail.com>> wrote: > > > > from the atttached kamailio.cfg file, what should be changed to > > get captagent receiver working in kamailio instead of port mirroring? > > > > Thanks > > > > > > On Fri, Feb 17, 2012 at 7:29 AM, Alexandr Dubovikov > > <voip(a)start4.info <mailto:voip@start4.info>> wrote: > > > > > > The link has been not changed, you know it already: > > > > http://code.google.com/p/homer/wiki/HOWTO > > > > Homer Capture Agent (for all other unsupported system) > > > > if your SIP system (PBX, proxy) doesn't have a built-in > > capture agent with HEP or IPIP encapsulation support, you can > > use Homer Capture Agent (captagent). This is an external > > application (daemon) that collect SIP messages thru PCAPlib > > and send them to your capture server. The captagent support > > IPv4 and IPv6. > > > > preventing your next question: the captagent your can find in > > the git. > > > > http://code.google.com/p/homer/source/browse/ > > > > > > Wbr, > > Alexandr > > > > > > > > 17.02.2012 13:15, Nelson Pereira wrote: > >> Can captagent be installed on a asterisk server? If so, is > >> there a howto? > >> > >> Sent from my iPhone4 on IOS4 > >> > >> On Feb 17, 2012, at 7:07 AM, Alexandr Dubovikov > >> <voip(a)start4.info <mailto:voip@start4.info>> wrote: > >> > >>> Hi, > >>> > >>> start two captagents instance, each on own interface and > >>> duplicate all traffic to your kamailio node. (HEP protocol) > >>> > >>> Wbr, > >>> Alexandr > >>> > >>> 16.02.2012 17:51, Nelson Pereira wrote: > >>>> I would like to know if someone can show me an example of > >>>> there setup configuration in kamailio.cfg for capturing > >>>> packets comming in from a mirror port. > >>>> The reason I ask is that if I do a TCPDUMP on the server, I > >>>> see all traffic to and from different sources on my Voice > >>>> network. > >>>> > >>>> But when I use Homer SIPCAPTURE which uses Kamailio in the > >>>> background, I only see traffic sent FROM my PBX but not > >>>> seing the traffic comming back to it. > >>>> > >>>> I mean I only see one side of the call, yet TCPDUMP > >>>> confirmed I see everything.... > >>>> > >>>> What am I missing ? > >>>> > >>>> Thanks > >>>> > >>>> NP > >>>> > >>>> > >>>> _______________________________________________ > >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > >>>> sr-users(a)lists.sip-router.org <mailto: > sr-users(a)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 > >> > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > >> sr-users(a)lists.sip-router.org <mailto: > sr-users(a)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 > > > > > > > > > > > > _______________________________________________ > > 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 > > > -- > 42com Telecommunication GmbH > Straße der Pariser Kommune 12-16 / D-10243 Berlin > > E-mail: mm(a)42com.com > Homepage: www.42com.com > > Firmenangaben/Company information: > > Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B > Umsatzsteuer-ID/VAT-ID: DE223812306, Geschäftsführer/CEO: Thomas Reinig, > Alexander Reinig > > > Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese > sind möglicherweise vertraulich und ausschließlich für den Adressaten > bestimmt. Sollten Sie diese elektronische Nachricht irrtümlicherweise > erhalten haben, so informieren Sie uns bitte unverzüglich telefonisch oder > per E-Mail. > > This message is intended only for the use of the individual or entity to > which it is addressed. If you have received this message in error, please > delete the message and notify us immediately. > > > --------------070003040807050803020008 > Content-Type: text/html; charset=ISO-8859-1 > Content-Transfer-Encoding: 7bit > > <html> > <head> > <meta content="text/html; charset=ISO-8859-1" > http-equiv="Content-Type"> > </head> > <body bgcolor="#FFFFFF" text="#000000"> > Hi,<br> > <br> > you are missing libpcap0.8 which is a dependency for the captagent.<br> > <br> > <br> > BR<br> > <br> > Max M.<br> > <br> > Am 17.02.2012 16:31, schrieb Nelson Pereira: > <blockquote > cite=" > mid:CAC6cfSgZgOJQ0-X5ANsHEQnmt+AvV96404dU_nr53dTNdbcUoA@mail.gmail.com" > type="cite">ok, tried to do the make install, getting the bellow: > <div> > <div>aurast:/usr/local/src/captagent # > /usr/local/bin/captagent</div> > <div>/usr/local/bin/captagent: error while loading shared > libraries: libpcap.so.0.8: cannot open shared object file: No > such file or directory</div> > <div><br> > </div> > <div>Do i need to compile this?</div> > <div><br> > </div> > <div>Also just tried running the captagent:</div> > <div> > <div>aurast:/usr/local/src/captagent # ./captagent</div> > <div>./captagent: error while loading shared libraries: > libpcap.so.0.8: cannot open shared object file: No such file > or directory</div> > <div><br> > </div> > </div> > <div>What am I doing wrong?</div> > <div><br> > </div> > <div>This is a SUSE SLES11 64bit server running asterisk 1.8.6</div> > <div><br> > </div> > <div>Thanks</div> > <div><br> > </div> > <div><br> > </div> > <br> > <div class="gmail_quote"> > On Fri, Feb 17, 2012 at 10:24 AM, Nelson Pereira <span > dir="ltr"><<a moz-do-not-send="true" > href="mailto:kitkat0981@gmail.com">kitkat0981(a)gmail.com > </a>></span> > wrote:<br> > <blockquote class="gmail_quote" style="margin:0 0 0 > .8ex;border-left:1px #ccc solid;padding-left:1ex"> > from the atttached kamailio.cfg file, what should be changed > to get captagent receiver working in kamailio instead of > port mirroring? > <div><br> > </div> > <div>Thanks</div> > <div class="HOEnZb"> > <div class="h5"> > <div><br> > <br> > <div class="gmail_quote"> > On Fri, Feb 17, 2012 at 7:29 AM, Alexandr Dubovikov > <span dir="ltr"><<a moz-do-not-send="true" > href="mailto:voip@start4.info" target="_blank"> > voip(a)start4.info</a>></span> > wrote:<br> > <blockquote class="gmail_quote" style="margin:0 0 0 > .8ex;border-left:1px #ccc solid;padding-left:1ex"> > <div bgcolor="#FFFFFF" text="#000000"> <br> > The link has been not changed, you know it > already:<br> > <br> > <a moz-do-not-send="true" > href="http://code.google.com/p/homer/wiki/HOWTO" > target="_blank"> > http://code.google.com/p/homer/wiki/HOWTO</a><br> > <br> > Homer Capture Agent (for all other unsupported > system)<br> > <br> > if your SIP system (PBX, proxy) doesn't have a > built-in capture agent with HEP or IPIP > encapsulation support, you can use Homer Capture > Agent (captagent). This is an external > application (daemon) that collect SIP messages > thru PCAPlib and send them to your capture > server. The captagent support IPv4 and IPv6.<br> > <br> > preventing your next question: the captagent > your can find in the git.<br> > <br> > <a moz-do-not-send="true" > href=" > http://code.google.com/p/homer/source/browse/" > target="_blank"> > http://code.google.com/p/homer/source/browse/</a><br> > <br> > <br> > Wbr,<br> > Alexandr > <div> > <div><br> > <br> > <br> > 17.02.2012 13:15, Nelson Pereira wrote: > <blockquote type="cite"> > <div>Can captagent be installed on a > asterisk server? If so, is there a > howto?<br> > <br> > Sent from my iPhone4 on IOS4</div> > <div><br> > On Feb 17, 2012, at 7:07 AM, Alexandr > Dubovikov <<a moz-do-not-send="true" > href="mailto:voip@start4.info" > target="_blank">voip(a)start4.info</a>> > > wrote:<br> > <br> > </div> > <blockquote type="cite"> > <div> Hi,<br> > <br> > start two captagents instance, each on > own interface and duplicate all > traffic to your kamailio node. (HEP > protocol)<br> > <br> > Wbr,<br> > Alexandr<br> > <br> > 16.02.2012 17:51, Nelson Pereira > wrote: > <blockquote type="cite">I would like > to know if someone can show me an > example of there setup configuration > in kamailio.cfg for capturing > packets comming in from a mirror > port. > <div>The reason I ask is that if I > do a TCPDUMP on the server, I see > all traffic to and from different > sources on my Voice network.</div> > <div><br> > </div> > <div>But when I use Homer SIPCAPTURE > which uses Kamailio in the > background, I only see traffic > sent FROM my PBX but not seing the > traffic comming back to > it. </div> > <div><br> > </div> > <div>I mean I only see one side of > the call, yet TCPDUMP confirmed I > see everything....</div> > <div><br> > </div> > <div>What am I missing ?</div> > <div><br> > </div> > <div>Thanks</div> > <div><br> > </div> > <div>NP</div> > <br> > <fieldset></fieldset> > <br> > > <pre>_______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > <a moz-do-not-send="true" href="mailto:sr-users@lists.sip-router.org" > target="_blank">sr-users(a)lists.sip-router.org</a> > <a moz-do-not-send="true" href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a> > </pre> > </blockquote> > <br> > </div> > </blockquote> > <blockquote type="cite"> > > <div><span>_______________________________________________</span><br> > <span>SIP Express Router (SER) and > Kamailio (OpenSER) - sr-users > mailing list</span><br> > <span><a moz-do-not-send="true" > href="mailto: > sr-users(a)lists.sip-router.org" > target="_blank"> > sr-users(a)lists.sip-router.org</a></span><br> > <span><a moz-do-not-send="true" > href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > </a></span><br> > </div> > </blockquote> > <br> > <fieldset></fieldset> > <br> > > <pre>_______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > <a moz-do-not-send="true" href="mailto:sr-users@lists.sip-router.org" > target="_blank">sr-users(a)lists.sip-router.org</a> > <a moz-do-not-send="true" href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a> > </pre> > </blockquote> > <br> > </div> > </div> > </div> > <br> > _______________________________________________<br> > SIP Express Router (SER) and Kamailio (OpenSER) - > sr-users mailing list<br> > <a moz-do-not-send="true" > href="mailto:sr-users@lists.sip-router.org" > target="_blank">sr-users(a)lists.sip-router.org > </a><br> > <a moz-do-not-send="true" > href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br> > <br> > </blockquote> > </div> > <br> > </div> > </div> > </div> > </blockquote> > </div> > <br> > </div> > <br> > <fieldset class="mimeAttachmentHeader"></fieldset> > <br> > <pre wrap="">_______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > <a class="moz-txt-link-abbreviated" href="mailto: > sr-users(a)lists.sip-router.org">sr-users(a)lists.sip-router.org</a> > <a class="moz-txt-link-freetext" href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a> > </pre> > </blockquote> > <br> > <br> > <pre class="moz-signature" cols="72">-- > 42com Telecommunication GmbH > Straße der Pariser Kommune 12-16 / D-10243 Berlin > > E-mail: <a class="moz-txt-link-abbreviated" href="mailto:mm@42com.com"> > mm(a)42com.com</a> > Homepage: <a class="moz-txt-link-abbreviated" href="http://www.42com.com"> > www.42com.com</a> > > Firmenangaben/Company information: > > Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B > Umsatzsteuer-ID/VAT-ID: DE223812306, Geschäftsführer/CEO: Thomas > Reinig, Alexander Reinig > > > Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. > Diese sind möglicherweise vertraulich und ausschließlich > für den Adressaten bestimmt. Sollten Sie diese elektronische Nachricht > irrtümlicherweise erhalten haben, so informieren Sie uns bitte > unverzüglich telefonisch oder per E-Mail. > > This message is intended only for the use of the individual or entity to > which it is addressed. If you have received this message in error, please > delete the message and notify us immediately.</pre> > </body> > </html> > > --------------070003040807050803020008-- > > > _______________________________________________ > 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 1

Need example of Kamailio.cfg for a port mirror setup
by Nelson Pereira 17 Feb '12

17 Feb '12

I would like to know if someone can show me an example of there setup configuration in kamailio.cfg for capturing packets comming in from a mirror port. The reason I ask is that if I do a TCPDUMP on the server, I see all traffic to and from different sources on my Voice network. But when I use Homer SIPCAPTURE which uses Kamailio in the background, I only see traffic sent FROM my PBX but not seing the traffic comming back to it. I mean I only see one side of the call, yet TCPDUMP confirmed I see everything.... What am I missing ? Thanks NP

2 5

Re: [SR-Users] how to match exact string value in mtree (Uri Shacked)
by Javier Gallart 17 Feb '12

17 Feb '12

Hello Uri I had similar needs and I found the ndb_redis module more suited for that type of task. Instead of a tree you have a hash like this: tname tprefix tvalue. If you do a hget nts $avp(DID) and you get a not null value you have found your exact match. It works very well for me and the time it takes for that "query" is barely noticeable. Of course redis does not address items like persistence, etc the same way a rdbms does. Hope it helps. Regards Javi On Thu, Feb 16, 2012 at 12:00 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. how to match exact string value in mtree (Uri Shacked) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 16 Feb 2012 12:54:50 +0200 > From: Uri Shacked <ushacked(a)gmail.com> > Subject: [SR-Users] how to match exact string value in mtree > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAMMbDhTFNXAE-K88=AeMjO7AnA_QJV3Ajj3AH-AxemXN3ze6HQ(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > I am using Mtree to match prefix numbers, some of them starts with 0 or > characters like D for example. > so, the mtree param is like this: > > modparam("mtree", "db_url", CFGDB) > modparam("mtree", "mtree", "name=nts;dbtable=service_numbers_view;type=0;") > modpmodparam("mtree", "char_list", "0123456789*+#YMDabcdefgh") > modparam("mtree", "pv_value", "$avp(mtval)") > modparam("mtree", "pv_values", "$avp(mtvals)") > > The thing is, that i tried all: > > if(!mt_match("nts", "$avp(DID)","1")) > if(!mt_match("nts", "$avp(DID)","2")) > if(!mt_match("nts", "$avp(DID)","0")) > > and lets say i have the both prefix in the nts mtree: > 09555 > 09555333 > > And the prefix i search for is $avp(DID)=09555444 > > I allwasy get the 09555 because it is the longest match. > I need exact match..... > > how do i do that? >

3 7

ndb_redis module fails after a while
by Javier Gallart 17 Feb '12

17 Feb '12

Hi all I have started making some tests with the ndb_redis module. So far we have not stressed the module (no more than 5 HGET commands/second at maximum). It works well, but with at some point it starts failing. The failures are easily found because the logs always show this: INFO: <core> [main.c:811]: INFO: signal 13 received After that the redis value is always null. If I restart kamailio it starts working again. I've run kamailio with debug=4 but I haven't seen more useful information. On the redis side, I could find nothing in the logs either, the number of clientes connected is alway much less than the configured maximum, Any idea? On the other hand, if I restart redis we need to restart kamailio to restore the connections. Is the reconnection to redis on the roadmap? Thanks in advance Regards Javi

3 12

Kamailio Exits With: Segmentation Fault Error 4 in usrloc.so
by Simpson Chua 17 Feb '12

17 Feb '12

Hi, Kamailio is exiting with a segmentation fault when trying to do a save("location"). Here is some information I gathered. Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: <core> [db.c:435]: invalid parameter value Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: usrloc [udomain.c:564]: failed to use table location Feb 16 15:30:19 ubuntu kernel: [171540.056663] kamailio[25721]: segfault at 89 ip 00007fa1f9a41452 sp 00007fff971a3af0 error 4 in usrloc.so[7fa1f9a37000+1f000] Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: <core> [main.c:751]: child process 25720 exited by a signal 11 Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: <core> [main.c:754]: core was generated Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD Core was generated by `kamailio -w /home/core'. Program terminated with signal 11, Segmentation fault. #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 474 vals[11].val.str_val = _c->sock->sock_str; (gdb) bt #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 #1 0x00007f5f916d9649 in insert_ucontact (_r=<optimized out>, _contact=<optimized out>, _ci=<optimized out>, _c=0x7fff52bbeb88) at urecord.c:473 #2 0x00007f5f914b785f in insert_contacts (_m=0x7f5f93d7f458, _d=0x7f5f8f72f680, _a=0x7fff52bbec00) at save.c:428 #3 0x00007f5f914b8d10 in add_contacts (_mode=1, _a=0x7fff52bbec00, _d=0x7f5f8f72f680, _m=0x7f5f93d7f458) at save.c:737 #4 save (_m=0x7f5f93d7f458, _d=0x7f5f8f72f680, _cflags=4) at save.c:790 #5 0x000000000041ba87 in do_action (h=0x7fff52bbf5d0, a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1128 #6 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1610 #7 0x000000000041c05b in do_action (h=0x7fff52bbf5d0, a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1107 #8 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1610 #9 0x0000000000422fa2 in run_top_route (a=0x7f5f93d7fe08, msg=0x7f5f93d7f458, c=<optimized out>) at action.c:1683 #10 0x00000000004b061e in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:207 #11 0x00007f5f91b038b0 in raw_capture_rcv_loop (rsock=5, port1=5060, port2=8933, ipip=0) at sipcapture.c:1518 #12 0x00007f5f91b03e1f in init_rawsock_children () at sipcapture.c:572 #13 0x00007f5f91b04033 in child_init (rank=<optimized out>) at sipcapture.c:542 #14 0x0000000000507a0f in init_mod_child (m=0x7f5f93d7b2b0, rank=0) at sr_module.c:886 #15 0x0000000000507974 in init_mod_child (m=0x7f5f93d7b668, rank=0) at sr_module.c:883 #16 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c178, rank=0) at sr_module.c:883 #17 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c4b0, rank=0) at sr_module.c:883 #18 0x0000000000476149 in main_loop () at main.c:1661 #19 0x000000000041a940 in main (argc=<optimized out>, argv=0x7fff52bbfd68) at main.c:2475 Any idea why this is happening? Is there something in the REGISTER message that is causing this? Thanks, Simpson

2 2

Re: [SR-Users] how to match exact string value in mtree (Uri Shacked)
by Daniel-Constantin Mierla 17 Feb '12

17 Feb '12

Hello, On 2/17/12 1:04 PM, Javier Gallart wrote: > Hi Daniel, > > no, in order to find the best match we do a manual trick using > the s.prefixes transformation (I think you suggested that). could be, at least is the way I do prefix matching with mysql. Postgres, on the other hand, afaik, has an index plugin to match the longest prefix, might not be in default distro, but can be found on the net for sure. > We start with the full $rU and stop when we find a non-null value. > This kamailio is serving many thousands of calls per hour, and the > response time is pretty much the same as when we used the mtree > module. For us , using redis has the advantage of being able to load > the tree data faster than when we used postgres without the need to > perform a mtree reload. Loading huge amount of records from mysql takes time indeed. Maybe prefix matching will be added in redis in the future, or scripted server side in lua, as I understood it has such feature. Another option would be to add a custom command in kamailio like redis_cmd_pmatch() to do internally these prefix-based commands until a match, could simplify the config if it becomes something very popular. Also, I forgot about the other solution in kamailio - pdb module written by 1&1 for number portability: * http://kamailio.org/docs/modules/stable/modules/pdb.html Just mentioning it here for let it be known, if redis is working fine for you, makes no sense to look for other solutions. Cheers, Daniel > > Regards > > Javi > > > On Fri, Feb 17, 2012 at 10:25 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello Javier, > > one question about redis, can it do longest prefix matching or all > prefixes match? > > Cheers, > Daniel > > > On 2/16/12 2:51 PM, Javier Gallart wrote: >> Hello Uri >> >> I had similar needs and I found the ndb_redis module more suited >> for that type of task. Instead of a tree you have a hash like >> this: tname tprefix tvalue. If you do a hget nts $avp(DID) and >> you get a not null value you have found your exact match. It >> works very well for me and the time it takes for that "query" is >> barely noticeable. Of course redis does not address items like >> persistence, etc the same way a rdbms does. >> Hope it helps. >> >> Regards >> >> Javi >> >> On Thu, Feb 16, 2012 at 12:00 PM, >> <sr-users-request(a)lists.sip-router.org >> <mailto:sr-users-request@lists.sip-router.org>> wrote: >> >> Send sr-users mailing list submissions to >> sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> or, via email, send a message with subject or body 'help' to >> sr-users-request(a)lists.sip-router.org >> <mailto:sr-users-request@lists.sip-router.org> >> >> You can reach the person managing the list at >> sr-users-owner(a)lists.sip-router.org >> <mailto:sr-users-owner@lists.sip-router.org> >> >> When replying, please edit your Subject line so it is more >> specific >> than "Re: Contents of sr-users digest..." >> >> >> Today's Topics: >> >> 1. how to match exact string value in mtree (Uri Shacked) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Thu, 16 Feb 2012 12:54:50 +0200 >> From: Uri Shacked <ushacked(a)gmail.com >> <mailto:ushacked@gmail.com>> >> Subject: [SR-Users] how to match exact string value in mtree >> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router >> (SER) - >> Users Mailing List" <sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org>> >> Message-ID: >> <CAMMbDhTFNXAE-K88=AeMjO7AnA_QJV3Ajj3AH-AxemXN3ze6HQ(a)mail.gmail.com >> <mailto:AeMjO7AnA_QJV3Ajj3AH-AxemXN3ze6HQ@mail.gmail.com>> >> Content-Type: text/plain; charset="iso-8859-1" >> >> Hi, >> >> I am using Mtree to match prefix numbers, some of them starts >> with 0 or >> characters like D for example. >> so, the mtree param is like this: >> >> modparam("mtree", "db_url", CFGDB) >> modparam("mtree", "mtree", >> "name=nts;dbtable=service_numbers_view;type=0;") >> modpmodparam("mtree", "char_list", "0123456789*+#YMDabcdefgh") >> modparam("mtree", "pv_value", "$avp(mtval)") >> modparam("mtree", "pv_values", "$avp(mtvals)") >> >> The thing is, that i tried all: >> >> if(!mt_match("nts", "$avp(DID)","1")) >> if(!mt_match("nts", "$avp(DID)","2")) >> if(!mt_match("nts", "$avp(DID)","0")) >> >> and lets say i have the both prefix in the nts mtree: >> 09555 >> 09555333 >> >> And the prefix i search for is $avp(DID)=09555444 >> >> I allwasy get the 09555 because it is the longest match. >> I need exact match..... >> >> how do i do that? >>

2 1

get myself ip address on start
by Uri Shacked 17 Feb '12

17 Feb '12

Hi, i would like to get the kamailio server ip address on start up and save it in an $shv(). i tried to use "myself" but did manage to get the ip from it. any ideas? Uri

1 0

Re: [SR-Users] Kamailio Exits With: Segmentation Fault Error 4 in usrloc.so
by Daniel-Constantin Mierla 17 Feb '12

17 Feb '12

Hello, On 2/17/12 1:11 PM, Alexandr Dubovikov wrote: > Hi Daniel, > > he use raw_socket in the sipcapture module and save all REGISTER > messages to location. (something like fraud controling for > registration). I think in this case he have to ignore socket listener. not sure I understood -- it is on the server that saves to homer's database? It gets the traffic over a raw socket and then executes the config file as usual with functions from other modules? Cheers, Daniel > > Wbr, > Alexandr > > > > > 17.02.2012 09:19, Daniel-Constantin Mierla wrote: >> Hello, >> >> looks like an invalid listen socket structure, quite strange... what >> version of kamailio do you have? >> >> Also, in gdb, frame 0, can you send the output of: >> >> p *_c >> p *_c->sock >> >> You can replace sensitive data (like IP), if you wish. >> >> Cheers, >> Daniel >> >> On 2/16/12 10:52 PM, Simpson Chua wrote: >>> Hi, >>> >>> Kamailio is exiting with a segmentation fault when trying to do a >>> save("location"). Here is some information I gathered. >>> >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: DEBUG: >>> <core> [parser/msg_parser.c:103]: found end of header >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: >>> <core> [db.c:435]: invalid parameter value >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: >>> usrloc [udomain.c:564]: failed to use table location >>> Feb 16 15:30:19 ubuntu kernel: [171540.056663] kamailio[25721]: >>> segfault at 89 ip 00007fa1f9a41452 sp 00007fff971a3af0 error 4 in >>> usrloc.so[7fa1f9a37000+1f000] >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: >>> <core> [main.c:751]: child process 25720 exited by a signal 11 >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: >>> <core> [main.c:754]: core was generated >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: INFO: <core> >>> [main.c:766]: INFO: terminating due to SIGCHLD >>> >>> Core was generated by `kamailio -w /home/core'. >>> Program terminated with signal 11, Segmentation fault. >>> #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 >>> 474 vals[11].val.str_val = _c->sock->sock_str; >>> (gdb) bt >>> #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 >>> #1 0x00007f5f916d9649 in insert_ucontact (_r=<optimized out>, >>> _contact=<optimized out>, _ci=<optimized out>, _c=0x7fff52bbeb88) at >>> urecord.c:473 >>> #2 0x00007f5f914b785f in insert_contacts (_m=0x7f5f93d7f458, >>> _d=0x7f5f8f72f680, _a=0x7fff52bbec00) at save.c:428 >>> #3 0x00007f5f914b8d10 in add_contacts (_mode=1, _a=0x7fff52bbec00, >>> _d=0x7f5f8f72f680, _m=0x7f5f93d7f458) at save.c:737 >>> #4 save (_m=0x7f5f93d7f458, _d=0x7f5f8f72f680, _cflags=4) at save.c:790 >>> #5 0x000000000041ba87 in do_action (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1128 >>> #6 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1610 >>> #7 0x000000000041c05b in do_action (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1107 >>> #8 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1610 >>> #9 0x0000000000422fa2 in run_top_route (a=0x7f5f93d7fe08, >>> msg=0x7f5f93d7f458, c=<optimized out>) at action.c:1683 >>> #10 0x00000000004b061e in receive_msg (buf=<optimized out>, >>> len=<optimized out>, rcv_info=<optimized out>) at receive.c:207 >>> #11 0x00007f5f91b038b0 in raw_capture_rcv_loop (rsock=5, port1=5060, >>> port2=8933, ipip=0) at sipcapture.c:1518 >>> #12 0x00007f5f91b03e1f in init_rawsock_children () at sipcapture.c:572 >>> #13 0x00007f5f91b04033 in child_init (rank=<optimized out>) at >>> sipcapture.c:542 >>> #14 0x0000000000507a0f in init_mod_child (m=0x7f5f93d7b2b0, rank=0) >>> at sr_module.c:886 >>> #15 0x0000000000507974 in init_mod_child (m=0x7f5f93d7b668, rank=0) >>> at sr_module.c:883 >>> #16 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c178, rank=0) >>> at sr_module.c:883 >>> #17 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c4b0, rank=0) >>> at sr_module.c:883 >>> #18 0x0000000000476149 in main_loop () at main.c:1661 >>> #19 0x000000000041a940 in main (argc=<optimized out>, >>> argv=0x7fff52bbfd68) at main.c:2475 >>> >>> Any idea why this is happening? Is there something in the REGISTER >>> message that is causing this? >>> >>> Thanks, >>> Simpson >>> >>> >>> _______________________________________________ >>> 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://linkedin.com/in/miconda -- http://twitter.com/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 > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 3

Question on ("sipcapture", "raw_interface", "")
by Nelson Pereira 17 Feb '12

17 Feb '12

Is it possible to monitor 2 interfaces? I have 2 Asterisk servers and they are on 2 different switches so I need to capture ETH0 and ETH3 which is where the mirror ports are connected to from each switch. Thanks

3 3

CAcert root in Lumicall
by Daniel Pocock 17 Feb '12

17 Feb '12

I've been contemplating Daniel's earlier question about using the CAcert certificates with Lumicall sip5060.net should already accept mutual authentication from other Kamailio instances running with a CAcert certificate However, the Lumicall dialer itself will only connect to servers that are using a cert signed by a root CA trusted within Android. This applies to both the SIP and STUN over TLS support. It may be possible for me to embed the CACert root in the Lumicall package and create a custom trust manager - allowing Lumicall to work with any server using CAcert TLS certificates. Does anyone have any strong feelings about not doing something like that?

3 9

how to match exact string value in mtree
by Uri Shacked 17 Feb '12

17 Feb '12

Hi, I am using Mtree to match prefix numbers, some of them starts with 0 or characters like D for example. so, the mtree param is like this: modparam("mtree", "db_url", CFGDB) modparam("mtree", "mtree", "name=nts;dbtable=service_numbers_view;type=0;") modpmodparam("mtree", "char_list", "0123456789*+#YMDabcdefgh") modparam("mtree", "pv_value", "$avp(mtval)") modparam("mtree", "pv_values", "$avp(mtvals)") The thing is, that i tried all: if(!mt_match("nts", "$avp(DID)","1")) if(!mt_match("nts", "$avp(DID)","2")) if(!mt_match("nts", "$avp(DID)","0")) and lets say i have the both prefix in the nts mtree: 09555 09555333 And the prefix i search for is $avp(DID)=09555444 I allwasy get the 09555 because it is the longest match. I need exact match..... how do i do that?

2 1

image missing in xmlrpc module doc
by Andrew Pogrebennyk 17 Feb '12

17 Feb '12

Hi, Was reading through the xmlrpc docs, could anybody with an access check why Figure "RPC Example" is missing from http://kamailio.org/docs/modules/devel/modules/xmlrpc.html#fig.rpc_example as well as 3.1 and 3.2 module docs? Thanks, Andrew

1 0

Kamailio in VUC session, next Friday, Feb 17
by Daniel-Constantin Mierla 17 Feb '12

17 Feb '12

Hello, there will be a VoIP Users Conference (VUC) session dedicated to Kamailio SIP Server next week, Friday, February 17, starting with 17:00GMT. VUC is an online weekly audio conferencing event, you can connect via SIP and IRC to interact with the other participants. More details (including the ways to connect) are available at: * http://www.kamailio.org/w/2012/02/kamailio-at-voipuserconference-fri-feb-17… Join the event to learn about latest developments and future plans for the project, ask questions about your interested topics. Spreading the word about the event is also appreciated. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 2

mi_xmlrpc not compiling on Solaris
by Nathaniel L Keeling 17 Feb '12

17 Feb '12

Hello, I am trying to compile mi_xmlrpc on Solaris 10 64bit OS and I am getting this error: CC (gcc) [M mi_xmlrpc.so] abyss_xmlrpc_server.o abyss_xmlrpc_server.c:948:1: error: conflicting types for 'shutdownAbyss' abyss_xmlrpc_server.c:945:34: note: previous declaration of 'shutdownAbyss' was here abyss_xmlrpc_server.c: In function 'normalLevelAbyssRun': abyss_xmlrpc_server.c:1005:42: warning: passing argument 2 of 'xmlrpc_registry_set_shutdown' from incompatible pointer type [enabled by default] /opt/csw/include/xmlrpc-c/server.h:114:1: note: expected 'void (* const)(struct xmlrpc_env * const, void * const, const char * const, void * const)' but argument is of type 'void (*)(struct xmlrpc_env * const, void * const, const char * const)' gmake[1]: *** [abyss_xmlrpc_server.o] Error 1 gmake: *** [modules_k] Error 1 Thanks Nathaniel L Keeling

3 2

How to do request URI rewrites using database tables?
by Ben WIlliams 16 Feb '12

16 Feb '12

Hi, can someone please recommend the most appropriate modules to rewrite R-URIs based on a database lookup table? I've read the documentation for lcr and carrierroute but not sure if they can do this. In most cases it will be a simple R-URI rewrite but I also need to rewrite based on From user. ie R-URI match From match new R-URI =========== ========== ========= *97(a)example.com 6[0-9](a)example.com *97(a)10.0.0.1 *97(a)example.com 7[0-9](a)example.com *97(a)10.0.0.2 6[0-9](a)example.com match any rewrite domain to 10.0.0.1 abc(a)example.com match any def(a)example.com Thanks Ben

1 1

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