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37456 discussions

MI Command in Script
by Reda Aouad 15 Mar '12

15 Mar '12

Hi, Can we call MI commands from the script ? I'm thinking about reloading dispatcher or hash table from database for example.. Thanks Reda

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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

3 3

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

3 6

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/

2 1

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

1 0

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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