- sr-users - kamailio.org

Siremis v2.1.0 Released
by Elena-Ramona Modroiu 23 Nov '11

23 Nov '11

Siremis v2.1.0 has been released - this is an update to previous release v2.0.0, bringing several enhancements and new web pages to manage PUA and RLS. It is still compatible with Kamailio v3.1.x, the last of this kind, next one to be out in the near future will be compatible with Kamailio v3.2.x. You can find the news about this release, including links to download, screenshots and demos, at: * http://siremis.asipto.com/2011/11/23/siremis-v2-1-0-released/ Alternative download site (tarball or git pull) is from sourceforge project: * http://sourceforge.net/projects/siremis/ Regards, Ramona PS. Siremis v2.1.0 is working for most of the components with Kamailio 3.2.0, just the few that changed the database structure may not be fully functional (e.g., the modules with tables that have new columns, see http://www.kamailio.org/wiki/install/upgrade/3.1.x-to-3.2.0#sql_commands).

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releasing v3.2.1
by Daniel-Constantin Mierla 23 Nov '11

23 Nov '11

Hello, I plan to package v3.2.1 after mid of next week (most probably on Thursday) -- it is about 1 month and a half since 3.2.0 was released and branch 3.2 accumulated a bunch of patches. Anyone having comments about it? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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kamailio 3.1.0 crash on ssl-dos attack
by Jijo 23 Nov '11

23 Nov '11

Hi All, Kamailio is resetting when we do TLS renegotiation dos attack using the tool available at http://www.thc.org/thc-ssl-dos/. Anybody looked at this issue? How we could resolve it. Any idea? The core generated for 3 pid's as below Pid 1: Core was generated by `/usr/sbin/kamailio -u swrun -g sw -m 120 -f /etc/kamailio/kamailio.cfg'. Program terminated with signal 11, Segmentation fault. #0 atomic_inc_int () at atomic/atomic_x86.h:225 (gdb) bt #0 atomic_inc_int () at atomic/atomic_x86.h:225 #1 cfg_update_local () at cfg/cfg_struct.h:228 #2 timer_main () at timer.c:994 #3 0x080b0579 in main_loop () at main.c:1632 #4 0x080b1be4 in main (argc=9, argv=0xbfd61e54) at main.c:2446 Pid 2: Core was generated by `/usr/sbin/kamailio -u swrun -g sw -m 120 -f /etc/kamailio/kamailio.cfg'. Program terminated with signal 11, Segmentation fault. #0 0x0819bfe8 in qm_insert_free (qm=0xaf6c5000, p=0xb05eec30, file=0xb6fb4140 "tls: tls_init.c", func=0xb6fb4ce0 "ser_free", line=296) at mem/q_malloc.c:184 184 if (frag->size <= f->size) break; (gdb) bt #0 0x0819bfe8 in qm_insert_free (qm=0xaf6c5000, p=0xb05eec30, file=0xb6fb4140 "tls: tls_init.c", func=0xb6fb4ce0 "ser_free", line=296) at mem/q_malloc.c:184 #1 qm_free (qm=0xaf6c5000, p=0xb05eec30, file=0xb6fb4140 "tls: tls_init.c", func=0xb6fb4ce0 "ser_free", line=296) at mem/q_malloc.c:518 #2 0xb6f95404 in ser_free (ptr=0xb05eec30) at tls_init.c:296 #3 0xb732e9ba in CRYPTO_free (str=0xb05eec30) at mem.c:391 #4 0xb7330bee in int_new_ex_data (class_index=5, obj=0xbfd414f4, ad=0xbfd41574) at ex_data.c:440 #5 0xb7330443 in CRYPTO_new_ex_data (class_index=5, obj=0xbfd414f4, ad=0xbfd41574) at ex_data.c:575 #6 0xb73dfde3 in X509_STORE_CTX_init (ctx=0xbfd414f4, store=0xafd8b3d0, x509=0xafe08ff0, chain=0x0) at x509_vfy.c:2114 #7 0xb74b0f31 in ssl3_output_cert_chain (s=0xb0553a10, x=0xafe08ff0) at s3_both.c:349 #8 0xb74a4728 in ssl3_send_server_certificate (s=0xb0553a10) at s3_srvr.c:3034 #9 0xb74a5879 in ssl3_accept (s=0xb0553a10) at s3_srvr.c:353 #10 0xb74afa8f in ssl3_read_bytes (s=0xb0553a10, type=23, buf=0xb0ad44ec "", len=4095, peek=0) at s3_pkt.c:1266 #11 0xb74ac9c9 in ssl3_read_internal (s=0xb0553a10, buf=0xb0ad44ec, len=4095, peek=0) at s3_lib.c:3265 #12 0xb74c24a9 in SSL_read (s=0xb0553a10, buf=0xb0ad44ec, num=4095) at ssl_lib.c:954 #13 0xb6fad1c3 in tls_read_f (c=0xb0ad431c, flags=0xbfd619c4) at tls_server.c:1058 #14 0x08171c0e in tcp_read_headers (c=0xb0ad431c, read_flags=0xbfd619c4) at tcp_read.c:406 #15 0x08171db8 in tcp_read_req (con=0xb0ad431c, bytes_read=0xbfd619cc, read_flags=0xbfd619c4) at tcp_read.c:885 #16 0x08172f67 in handle_io (fm=<value optimized out>, events=1, idx=<value optimized out>) at tcp_read.c:1234 #17 0x0817583b in io_wait_loop_epoll (unix_sock=89) at io_wait.h:1092 #18 tcp_receive_loop (unix_sock=89) at tcp_read.c:1345 #19 0x0816e2e9 in tcp_init_children () at tcp_main.c:4867 #20 0x080affb1 in main_loop () at main.c:1646 #21 0x080b1be4 in main (argc=9, argv=0xbfd61e54) at main.c:2446 Pid 3: Core was generated by `/usr/sbin/kamailio -u swrun -g sw -m 120 -f /etc/kamailio/kamailio.cfg'. Program terminated with signal 11, Segmentation fault. #0 0xb76c9e7c in memmove () from /lib/libc.so.6 (gdb) bt #0 0xb76c9e7c in memmove () from /lib/libc.so.6 #1 0x081724e7 in tcp_read_req (con=0xb022c8f0, bytes_read=0xbfd619cc, read_flags=0xbfd619c4) at tcp_read.c:1026 #2 0x08172f67 in handle_io (fm=<value optimized out>, events=1, idx=<value optimized out>) at tcp_read.c:1234 #3 0x0817583b in io_wait_loop_epoll (unix_sock=93) at io_wait.h:1092 #4 tcp_receive_loop (unix_sock=93) at tcp_read.c:1345 #5 0x0816e2e9 in tcp_init_children () at tcp_main.c:4867 #6 0x080affb1 in main_loop () at main.c:1646 #7 0x080b1be4 in main (argc=9, argv=0xbfd61e54) at main.c:2446

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The requested URL /siremis/index.php/user/login was not found on this server.
by Jonny Klompas 23 Nov '11

23 Nov '11

Installation was proceeding successfully, until I reached the following stage: http://kb.asipto.com/images/siremis-2.0/siremis20-install05.png Then I receive above error. Thanks Jonny

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Kamailio best performance
by Ricardo Martinez 23 Nov '11

23 Nov '11

Hello list. I’m having some issues with the Kamailio versión 3.2.0. I want to ask if someone could give some hints how to optimize the performance of my kamailio server. For some reason and from time to time the kamailio process start to answer slower than usual, making calls fail and register expires. I’m still unable to detect the problem, but I want to know of maybe i´m running my kamailio not under the best conditions. This is part of my configuration : #!KAMAILIO #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 # ----------- global configuration parameters ------------------------ debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 children=16 port=5060 memdbg=9 memlog=9 listen=udp:10.0.10:5060 disable_tcp=yes server_signature=0 port=5060 # ----------------- setting module-specific parameters --------------- ## modparam("registrar", "received_avp", "$avp(s:rcv)") modparam("usrloc", "db_mode", 1) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("auth_db|permissions|uri_db|usrloc","db_url","mysql://openser:openserrw@localhost /openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("domain", "db_mode", 1) # ----- nathelper params ----- modparam("nathelper", "natping_interval", 20) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) modparam("nathelper","natping_interval", 20) ## modparam("nathelper","received_avp", "$avp(i:42)") modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock") modparam("mediaproxy", "signaling_ip_avp", "$avp(s:signaling_ip)") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") #modparam("tm", "fr_timer", 3) # ------ dialog params ------- modparam("dialog", "dlg_flag", 4) modparam("dialog", "profiles_with_value", "caller") # ------ pike params -------- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 25) modparam("pike", "remove_latency", 4) # ------ rr params -------- modparam("rr", "enable_full_lr", 1) # ----- misc_radius params ----- modparam("misc_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("misc_radius", "caller_service_type", 18) modparam("misc_radius", "callee_service_type", 19) modparam("misc_radius", "caller_extra", "Called-Station-Id=$ru") modparam("misc_radius", "callee_extra", "Called-Station-Id=$fu") # ---- htable param --------- modparam("htable", "htable", "a=>size=8;") modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=STATS") modparam("sqlops","sqlcon", "ca=>mysql://openser:openserrw@localhost/openser") #------ uac --------------- modparam("uac","rr_store_param","my_param") modparam("uac","from_restore_mode","auto") modparam("uac","auth_realm_avp","$avp(i:10)") modparam("uac","auth_username_avp","$avp(i:11)") modparam("uac","auth_password_avp","$avp(i:12)") Beside of this I have syslogd in asyn mode… This is the info of the kamailio –V version: kamailio 3.2.0 (x86_64/linux) 639f0a 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: 639f0a compiled on 11:35:43 Oct 28 2011 with gcc 4.5.1 I’m using pike to check flood alerts, but I have a white list stored in the “user_preference” table : I’m using it like this : route[REQINIT] { # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if( !(avp_db_load("pike", "$avp(ip_origen)") && avp_check("$avp(ip_origen)", "eq/$src_ip/gi")) ) #+---------+----------+--------+-----------+----------------+------+---------------------+ #| uuid | username | domain | attribute | value | type | modified | #+---------+----------+--------+-----------+----------------+------+---------------------+ #| pike | | | ip_origen | 10.0.0.44 | 0 | 2008-01-04 13:24:14 | #| pike | | | ip_origen | 10.0.0.66 | 0 | 2008-01-04 17:28:59 | { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); xlog("L_INFO","ALERT: pike blocking from $si\n"); $sht(ipban=>$si) = 1; exit; } } Is this check method too slow? Or intensive in mysql access request?? Hope someone can help me here. Thanks in advance.. Regards, *Ricardo Martinez.-*

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simple proxy
by Iurii Andamasov 23 Nov '11

23 Nov '11

Hello list. First of all sorry for dump question, i'm pretty new to kamailio so i hope someone point me in right direction currently i have one sip device which can handle sip request only from one ip, my goal is to put kamailio in front of it so i can proxy requests from several ips i added this to conf route{ if ( src_ip == 10.46.1.121 || src_ip == 10.46.1.122 ) { if(method == "INVITE") { $rd="10.46.29.10"; t_relay(); } } and calls seems connecting but signaling is completely broken i understand that i'm doing something wrong, but can't get what is wrong Please point me to solution Thanks

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strange behavior when trying to reach PSTN gateway via TCP
by Dmitry Petrakoff 23 Nov '11

23 Nov '11

Hi List! I'm running Kamailio for about 1 year without any serious problems. When it works with UDP only, it runs perfectly well for us. Without any major issues. But now we have a customer who wants to connect Lync to our network. Unfortunately, Lync does not support UDP for SIP signalling, so we tried to teach kamailio to send SIP messages via TCP. And getting strange error: 10(36629) DEBUG: tm [t_reply.c:659]: DEBUG: reply sent out. buf=0x832c7ac: SIP/2.0 100 trying -..., shmem=0x28d97c7c: SIP/2.0 100 trying - 10(36629) DEBUG: tm [t_reply.c:669]: DEBUG: _reply_light: finished 10(36629) DEBUG: <core> [forward.c:213]: DEBUG: get_out_socket: socket determined: 0x833ddc0 10(36629) DEBUG: <core> [msg_translator.c:457]: clen_builder: content-length: 553 (553) 10(36629) DEBUG: <core> [msg_translator.c:204]: check_via_address(10.200.245.200, 10.200.245.200, 0) 10(36629) DEBUG: <core> [tcp_main.c:1824]: tcp_send: no open tcp connection found, opening new one 10(36629) DEBUG: <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: 10.198.3.100 10(36629) DEBUG: <core> [tcp_main.c:1081]: tcpconn_new: on port 5060, type 2 10(36629) DEBUG: <core> [tcp_main.c:1382]: tcpconn_add: hashes: 0:0:0, 1 10(36629) ERROR: <core> [tcp_main.c:2748]: connect 10.198.3.100:5060 failed Socket is not connected 10(36629) ERROR: <core> [tcp_main.c:2754]: 10.198.3.100:5060: connect & send for 0x28d988ac failed: Socket is not connected (57) 10(36629) DEBUG: <core> [tcp_main.c:2955]: tcpconn_chld_put: destroying connection 0x28d988ac (1, -1) flags 0060 10(36629) ERROR: tm [../../forward.h:170]: msg_send: ERROR: tcp_send failed We tested two different variants: one when defined Lync in dr_gateways table as 10.198.3.100:5068;transport=tcp And then, tried global options udp_mtu=1300 udp_mtu_try_proto=TCP As soon as these options were enabled, all other gateways became unreachable with the same error as above. Could you please to point me out, where I'm wrong? fbsd-kam# /usr/local/sbin/kamailio -V version: kamailio 3.1.5 (i386/freebsd) ec672a 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, 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, PKG_SIZE 4MB poll method support: poll, select, kqueue. id: ec672a compiled on 11:02:58 Nov 15 2011 with gcc 4.2.1 FreeBSD fbsd-kam.dprs-consulting.com 8.2-RELEASE-p2 FreeBSD 8.2-RELEASE-p2 #0: Sun Jul 3 17:49:13 GMT 2011 root@fbsd-kam.dprs-consulting.com:/usr/obj/usr/src/sys/GENERIC i386 Routing script could be sent on request. Thank you very much! WBR Dimon

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parse 503 message
by Robert R 22 Nov '11

22 Nov '11

Hi, How can I set a filter for receiving 5xx messages, i.e. how can I parse 503 messages received by the proxy? I have tried the following and none works: t_check_status("503") is_method("503") Thanks, R

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(no subject)
by Charles Wang 22 Nov '11

22 Nov '11

http://functioncatering.co.za/modules/mod_wdbanners/blog.php?html143

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LCR weight parameter
by Fabian Borot 22 Nov '11

22 Nov '11

Hello On kamailio 1.5.2, I need to choose the gw from a group according to its weight. After trying unsuccessfully with different values I realized that the help guide says that the load_gw functions can return gateways according to "gateway's randomized weight within its group". So, I have these questions: 1- since it is "randomized", does this mean that the weight parameter actually can not guarantee that the weight value can decide how gateways are ordered? 2- once I have the gws on the avp, is it posible to re-order them? if so, how? thank you very much fborot

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Broadcasting message to users...
by Krishna Kurapati 22 Nov '11

22 Nov '11

Hi, During planned down time of Kamailio, I would like to send a Notice to all the users currently registered on the system. It would be nice if it appears in their Chat windows. I am thinking if I could loop through user_location table and send MESSAGE to individual user. What's the best way to accomplish this task? Thanks Krish Kura

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Re: [SR-Users] Core compiled with Sctp support
by Daniel-Constantin Mierla 22 Nov '11

22 Nov '11

Don't forget to keep the mailing list cc-ed -- private emails sent from mailing lists discussion are going to be discarded. If you have libsctp devel package installed, try compilation with: make SCTP=1 ... all Cheers, Daniel On 11/18/11 12:13 PM, Stefan Ljung wrote: > > So, its not. > > [test032@wslint90]# ./ser -V > > version: ser 3.1.2 (i386/linux) 2fe4d6 > > 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, PKG_SIZE 4MB > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > id: 2fe4d6 > > compiled on 11:31:57 Nov 18 2011 with gcc 4.1.2 > > This result came after an untar and then make command. > > Here's my machine : > > [linux]# uname -a > > Linux wslint90 2.6.23.1-21.fc7 #1 SMP Thu Nov 1 21:09:24 EDT 2007 i686 > i686 i386 GNU/Linux > > What can I do to force it in ? > > /Stefan > > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* den 18 november 2011 11:59 > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* Stefan Ljung > *Subject:* Re: [SR-Users] Core compiled with Sctp support > > Hello, > > when sctp support is available, if you do 'kamailio -V' or 'ser -V' > you should see USE_SCTP flag there. > > Not sure if sctp is enabled by default in listen sockets, you can do > it anyhow with global parameter: > > enable_sctp=yes > > Cheers, > Daniel > > On 11/18/11 11:49 AM, Stefan Ljung wrote: >> I've downloaded sip-router 3.1.2 and have been struggling with the >> make commands for configuring. Compiling and installing. >> Can't really say that I master it. >> But I have finally managed to do all steps and started a example ser.cfg. >> One of the features I want to play around eith is the sctp transport. >> From previous experiments with OpenSER I kearned that if I only >> specify a listen host address without transport protocol all three >> UDP TCP and SCTP sockets will be started. >> Ths time I only see UDP and TCP: >> Listening on >> udp: 127.0.0.1:5060 >> tcp: 127.0.0.1:5060 >> Aliases: >> tcp: localhost:5060 >> tcp: localhost.localdomain:5060 >> udp: localhost:5060 >> udp: localhost.localdomain:5060 >> I have read the c sources to verify that the sctp libs and header >> files are in the right places, and they are. >> What I could need is a tip on how to verify that my core has been >> compiled with sctp support. How can I see that ? >> In what files would there be a confimation of that the sctp feature >> will be compiled in? >> /Stefan >> >> >> _______________________________________________ >> 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 > Kamailio Advanced Training, Dec 5-8, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Trouble with SERVICE
by jermaine.boss＠gmail.com 22 Nov '11

22 Nov '11

Greetings: I am mailing around trouble with the product attaching service to a STANDARD sip phone outfit. My account user name is: call.8675309 The problem presents it self in a manner after login attemps where now a message around a NAT Router Cone problem exist. Please have the service to the account reset or otherwise further advise me. Sincerly J. Boss

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How long (in bytes) is an AVP which reads an Integer from DB?
by Iñaki Baz Castillo 22 Nov '11

22 Nov '11

Hi, I need to read from a MySQL table a 32 bits long Integer column. My question is: how many bytes does Kamailio use for storing integer values fetched from a DB query? does it depend on the 32/64 bits architecture? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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How to
by Sébastien Boutet 22 Nov '11

22 Nov '11

Hello, In the features of kamailio, I can see 'replication for High Availability (HA)' But I can't find any tutorials for HA install. Could you help me or did you have any book wich explain how to do that ? Thank's Best reagard, Sébastien BOUTET **

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New module: xhttp_rpc
by Ovidiu Sas 21 Nov '11

21 Nov '11

Hello all, A new module providing a web interface to kamailio/sip-router RPC interface is available in trunk: http://kamailio.org/docs/modules/devel/modules/xhttp_rpc.html http://sip-router.org/docbook/sip-router/branch/master/modules/xhttp_rpc/xh… The module is using the embedded web server provided by xhttp module: http://kamailio.org/docs/modules/devel/modules/xhttp.html A simple config that will enable the web interface is provide on the README file. Simply point your browser to http://<server_IP>:<TCP_port>/<xhttp_rpc_root> and start browsing available RPC commands. For more info about the module please see the README file. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

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kamctl lcr
by Jonny Klompas 21 Nov '11

21 Nov '11

HI, When I try and run this on ver 3.2 I receive the following error: ERROR: command disabled How do I install or configure lcr? Thanks Jonny

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New module: presence_profile
by MÉSZÁROS Mihály 21 Nov '11

21 Nov '11

Hello All, I've committed a new module named 'presence_profile' which provides UA config provisioniong according RFC6080. http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=cd… The README describes the module function: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=mo… The modul is very very simple in it's current stage. The modul added only the event 'ua-profile' handling. Best Regards, Mihály Mészáros

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GIT committing and backporting
by Daniel-Constantin Mierla 21 Nov '11

21 Nov '11

Hello, I put together a wiki page to help people porting patches between branches, exemplified with master and 3.2 branches -- the typical patch backporting these days: * http://www.kamailio.org/wiki/devel/backporting-to-3.2.x They are useful also for people with no write access, in case they want to try patches committed to devel branch in their local git repo clones. In additions, I added to kamailio wiki the page with guidelines for commits, trying to give the generic suggestions of how to do it in order to keep the coherence of the development: * http://www.kamailio.org/wiki/devel/git-commit-guidelines Updates and enhancements are welcome! If one has more suggestions or want to debate some of the proposed guidelines, feel free to start the discussion on the mailing list. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Limiting simultaneous calls
by Mino Haluz 21 Nov '11

21 Nov '11

Hi, I was using cdrtool (prepaid table) and callcontrol to limit concurrent calls. In fact this is only limiting the outbound calls, but I would like to use another mechanism which should limit the inbound calls too. So basically to limit voice channels. So is there some reliable method/module how to achieve this? Thanks, Mino

3 5

Dropped registrar bindings with 1.5.3
by Alex Balashov 19 Nov '11

19 Nov '11

I have a 1.5.3 installation functioning as a registrar that is exhibiting a very curious, if only infrequent set of behaviours. The host is CentOS 5.x, with PostgreSQL 9.0 backing usrloc, and db_mode = 1 (immediate write-through). I have a registration handler in my initial request route that begins with a log message: ... else if(is_method("REGISTER")) { xlog("L_INFO", "... Processing REGISTER from $si:$sp for AOR $tu\n"); route(2); exit; } I've got a registrant who consistently re-registers with an expiration time of 120, so in theory they ought to be re-registering every two minutes or so. Most of the time they do. But, occasionally, I'll see a gap of 5-8 minutes go by, in which the above log message does not appear, and their registration expires so that they can't receive calls. That's not what's interesting. What's interesting is that I did a packet capture on the proxy and caught one of these scenarios. Within that ~8 minute window when the registration expired and calls were failing, the UAC actually re-registered several times, on normal ~2 minute boundaries, AND the proxy successfully challenged the request and sent a 200 OK indicating that the binding was stored! I double-checked all the parameters to ensure that the 401s and 200 OKs corresponded to the right REGISTER flow, etc. - yes, it's all correct. Is there any imaginable set of circumstances or a known bug that would cause Kamailio to successfully authenticate and affirm a registration request, while neither logging its receipt, nor, evidently, storing it to the database? I can envision a number of database failure modes that would account for the binding not actually being in the 'location' table, but that doesn't explain why the request wouldn't even be logged. That's what has me baffled. Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

6 15

[Kamailio-Users] Converting rtpproxy recorded .rtp file to .wav file
by Vikram Ragukumar 18 Nov '11

18 Nov '11

Hello, I used Kamailio+rtpproxy to record a session and rtpproxy outputs the following files long_file_name.a.rtp, long_file_name.a.rtcp, long_file_name.o.rtp, long_file_name.o.rtcp http://www.rtpproxy.org/wiki/RTPproxy/FAQ From the Rtpproxy FAQ above, i tried to extract the audio using rtpbreak and sox. rtpbreak -W -r long_file_name.a.rtp rtpbreak -W -r long_file_name.o.rtp The above commands generate rtp.0.0.raw, rtp.1.0.raw. Then when i run sox using sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t wavpcm -s out.wav i get the following errors : sox: invalid option -- - sox: -c must be given a number Is there a switch/anything else that i am missing ? Thanks in advance, Regards, Vikram.

3 5

sqlops crash when connection does not exist
by Juha Heinanen 18 Nov '11

18 Nov '11

i misspelled connection argument in sql_query and got crash: Nov 10 17:19:15 sip /usr/sbin/sip-proxy[4182]: ERROR: sqlops [sqlops.c:266]: invalid connection [sip-proxy] Nov 10 17:19:15 sip /usr/sbin/sip-proxy[4182]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/sip-proxy/sip-proxy.cfg:281 Nov 10 17:19:15 sip kernel: [ 990.129668] sip-proxy[4182]: segfault at 786e6570 ip b769de38 sp bf960ffc error 4 in libc-2.11.2.so[b762b000+140000] (gdb) where #0 0xb769de38 in strcmp () from /lib/i686/cmov/libc.so.6 #1 0x080f54f9 in find_mod_export_record (mod=0xb750bb80 "db_mysql", name=0xb720d353 "db_bind_api", param_no=0, flags=0, mod_if_ver=0xbf96108c) at sr_module.c:657 #2 0x080f5756 in find_mod_export (mod=0xb750bb80 "db_mysql", name=0xb720d353 "db_bind_api", param_no=0, flags=0) at sr_module.c:722 #3 0xb720449c in db_bind_mod (mod=0xb6628944, mydbf=0x8) at db.c:209 #4 0xb66240d2 in ht_db_init_con () at ht_db.c:75 #5 0xb661b144 in destroy () at htable.c:225 #6 0x080f3684 in destroy_modules () at sr_module.c:782 #7 0x0809087c in cleanup (show_status=0) at main.c:564 #8 0x08091509 in shutdown_children (show_status=0, sig=<value optimized out>) at main.c:706 #9 0x080940f7 in main (argc=17, argv=0xbf961354) at main.c:2523 -- juha

2 3

Core compiled with Sctp support
by Stefan Ljung 18 Nov '11

18 Nov '11

I've downloaded sip-router 3.1.2 and have been struggling with the make commands for configuring. Compiling and installing. Can't really say that I master it. But I have finally managed to do all steps and started a example ser.cfg. One of the features I want to play around eith is the sctp transport. >From previous experiments with OpenSER I kearned that if I only specify a listen host address without transport protocol all three UDP TCP and SCTP sockets will be started. Ths time I only see UDP and TCP: Listening on udp: 127.0.0.1:5060 tcp: 127.0.0.1:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: localhost:5060 udp: localhost.localdomain:5060 I have read the c sources to verify that the sctp libs and header files are in the right places, and they are. What I could need is a tip on how to verify that my core has been compiled with sctp support. How can I see that ? In what files would there be a confimation of that the sctp feature will be compiled in? /Stefan

2 1

Problem trying to play recorded message before call
by David Candamil Santos 18 Nov '11

18 Nov '11

Hi there, guys. I hope you can help me. I use kamailio to comunicate some servers (each one with its own kamailio). When an user calls another, depending on the userID that will receive the call, the server redirects the call to the corresponding kamailio server. It works properly. Now, what I would like to do is to configure it so it plays a notification saying where the call is going to be forwarded. After reading some tutorials, I tried to get it working with rtpproxy, but after configuring it, it doesn't play the clips. I am testing it with a *.wav file. The path to the file is "/usr/local/etc/kamailio/test.wav" (it's in the kamailio directory). This is my software: Debian stable kernel 2.6.32-5-686 Kamailio 3.2.0 (GIT installation) rtpproxy 1.2.1-1 Kamailio and rtpproxy are running in the same machine. These are the log messages: -------------------kamailio.log------------------- (SEVERAL TIMES) Nov 16 12:58:10 debian-virtualbox kamailio[2076]: INFO: rtpproxy [rtpproxy.c:1415]: rtp proxy <udp:127.0.0.1:22222> found, support for it enabled (SEVERAL TIMES) Nov 16 12:58:12 debian-virtualbox kamailio[2069]: ERROR: <script>: Intentando reproducir audio -------------------------------------------------- This is the relevant configuration: --------------/etc/default/rtpproxy--------------- # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 # Additional options that are passed to the daemon. EXTRA_OPTS="-l 127.0.0.1" -------------------------------------------------- -----/usr/local/etc/kamailio/kamailio.cfg (just my important changes for this action)--------- loadmodule "rtpproxy.so" modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") request_route { rtpproxy_offer(); xlog("L_ERR","Intentando reproducir audio"); rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav","1"); (...default routes ommited...) route(RELAY); } -------------------------------------------------- I can play it with mplayer, so it's not a sound problem (or even a system codec problem). I would like to know if I can make it working with rtpproxy, or if there is an easier way to do it. Thanks for your time. (I include now the whole kamailio.cfg, just in case you need to check anything) ---------/usr/local/etc/kamailio/kamailio.cfg (complete) ---------- #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # ##-- # WITH_DEBUG # *** To enable mysql: # - define WITH_MYSQL # ##-- #!define WITH_MYSQL # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # ##-- #!define WITH_AUTH # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # ##-- #WITH_NAT # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #$banana_server="192.168.0.70" #$treasure_server="192.168.0.60" #$drake_server="192.168.0.60" #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif ##-- loadmodule "sdpops.so" loadmodule "textopsx.so" loadmodule "rtpproxy.so" ##- loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- ##-- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { #start_recording(); #force_rtp_proxy(); #rtpproxy_offer(); #rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav", "1"); #("fichero","numero de veces"); #formato alaw? #force_rtp_proxy(); rtpproxy_offer(); xlog("L_ERR","Intentando reproducir audio"); rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav","1"); if ($rU=~"^001788[0-3].*") { xlog("L_ERR", "Banana island"); $rd="192.168.0.70"; msg_apply_changes(); xlog("L_ERR","Redireccionando a $rd\n"); } if ($rU=~"^001788[4-5].*") { xlog("L_ERR", "Big treasure island"); $rd="192.168.0.60"; msg_apply_changes(); xlog("L_ERR","Redireccionando a $rd\n"); } if ($rU=~"^001788[6-7].*") { xlog("L_ERR", "Drake island"); $rd="192.168.0.60"; msg_apply_changes(); xlog("L_ERR","Redireccionando a $rd\n"); } if ($rU=~"^001788[8-9].*") { xlog("L_ERR", "Lost island"); sdp_keep_codecs_by_name("GSM"); #if para si true, bien, y si false mensaje y sale } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); ##-- #rtpproxy_offer(); #rtpproxy_stream2uas("/usr/local/etc/kamailio/test.wav", "1"); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } --------------------------------------------------------------

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Re: [SR-Users] mtree on db Berkeley DB?
by Javier Gallart 17 Nov '11

17 Nov '11

> > Hello > > coming back to the topic related to mtree, to be able to set values via > mi/rpc -- it won't be that difficult to add such functionality. Usually > with a tree is mainly reading, due to fast matching on tree indexing. > The question is how many times/often do you need to change values and > how many of them at the same time (more or less). > > I was also thinking about that: our application works in such a way that the full tree is recalculated every 15 minutes. Currently we have aound 60 branches with 150000 entries each. > I assume many times the changes will be somewhere down the tree, since > the first part of the number is usually the same (e.g., country code and > operator prefix). To update the tree at runtime, while there are reads > on it, there must be used a lock to be safe an consistent. If you do lot > of writes and very often, then you keep the tree structure locked a lot, > slowing the search. > That's the case, tname is rarely updated, it's tvalue the column that we normally update. > > Can you estimate the number of writes and how often they would be on a > daily basis? There might be other solutions to look at, if writes are > very often. > >From the numbers above, let's say that we need to update around 8M records every 15 minutes (we expect this number will keep steadily increasing...) As a side note, I've looked at how to implement the mt_match equivalent in redis and it does't look that hard using kamailio s.prefixed transformation (as you suggested) and redis sorted sets. I'll need to make more tests to check the performance, I'll share the results. Thanks! Javi > > Cheers, > Daniel > > > > Regards > > > > Javi >

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Re: [SR-Users] Problem with NAT and RTPProxy
by David Candamil Santos 16 Nov '11

16 Nov '11

Hi, I'm just using it because after reading lots of manuals and tutorials, it was the solution I thought it could work. The smartphone is outside the network. I am trying to use it over the Vodafone network. What do you suggest then? Thanks. 2011/11/14 Pavel Segeč <Pavel.Segec(a)fri.uniza.sk> > Hi, **** > > ** ** > > from your topology description I'm not sure why you are trying to use > RTPproxy when all your equipments are on private segments. RTPproxy should > have an public IP address**** > > ** ** > > palo**** > > >

2 1

mtree on db Berkeley DB?
by Javier Gallart 16 Nov '11

16 Nov '11

Hi list we've been happily using the mtree module for months now. Lately the size of the tree has grown a lot. The mtree table needs to be fully repopulated and reloaded several times a day, and we are looking for a fastest mechanism (for populating the table, I guess the reload time does not depend much on the db backend...). Does anyone tried with Berkeley DB? Is this combination mtree-berkeley actually feasible...? Thanks Javi

7 19

tls-modules in 3.2.0
by Mokhtar Bengana 15 Nov '11

15 Nov '11

Hello, I am trying to install the tls module in 3.2.0. I run apt-get update before apt-get install kamailio-tls-modules and still can't load the tls module and getting the following error. Kamailio can't find the tls module even though it's installed? I didn't get any errors when I run apt-get install kamailio-tls-modules. Everything else is defined in the config file. Has someone tried to install tls in the latest release and is having issues? Any help much appreciated. #!define WITH_TLS #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif root@Kamalio:/usr/local/lib/kamailio/modules_k# kamailio -c loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(12131) ERROR: <core> [sr_module.c:553]: ERROR: load_module: could not find module <tls> in </usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/> 0(12131) : <core> [cfg.y:3501]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 268, column 12-19: failed to load module 0(12131) ERROR: <core> [modparam.c:162]: set_mod_param_regex: No module matching <tls> found 0(12131) : <core> [cfg.y:3504]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 422, column 50: Can't set module parameter ERROR: bad config file (2 errors) root@Kamalio:/usr/local/lib/kamailio/modules_k# root@Kamalio:/usr/local/lib/kamailio/modules_k# apt-get install kamailio-tls-modules Reading package lists... Done Building dependency tree Reading state information... Done The following NEW packages will be installed: kamailio-tls-modules 0 upgraded, 1 newly installed, 0 to remove and 3 not upgraded. Need to get 0 B/399 kB of archives. After this operation, 848 kB of additional disk space will be used. Selecting previously deselected package kamailio-tls-modules. (Reading database ... 30034 files and directories currently installed.) Unpacking kamailio-tls-modules (from .../kamailio-tls-modules_3.2.0+squeeze1_i386.deb) ... Setting up kamailio-tls-modules (3.2.0+squeeze1) ... Thank you,

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Fwd: [Sip-implementors] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters
by marius zbihlei 15 Nov '11

15 Nov '11

-------- Original Message -------- Subject: [Sip-implementors] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters Date: Mon, 14 Nov 2011 20:28:53 +0100 From: Kevin P. Fleming <kpfleming(a)digium.com> To: sip-implementors(a)lists.cs.columbia.edu <sip-implementors(a)lists.cs.columbia.edu> Greetings, This is a call for presenters (with presentations!) for the telephony and communications devroom at FOSDEM 2012 - http://www.fosdem.org/. We will be holding a day full of presentations on development topics in the area of open source telephony and communications on Sunday, February 5th. The schedule allows for presentations from 9:00 to 17:00. The room we will be using will have a projector, wifi, and 100 seats. Please submit all proposals no later than 2011-12-09. Notification of accepted speakers will be provided by 2011-12-16. We will then work to have a schedule finalized by 2012-01-06. Talks should be submitted by subscribing to and then posting to the telephony-devroom mailing list hosted on http://lists.fosdem.org/. If you would like to contact the devroom organizer directly, please contact Kevin P. Fleming<kpfleming at digium.com>. Feel free to forward this along to any people or mailing lists that you think would be interested in this event. Thank you! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kfleming(a)digium.com | SIP: kpfleming(a)digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com& www.asterisk.org _______________________________________________ Sip-implementors mailing list Sip-implementors(a)lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

1 0

Re: [SR-Users] Compile kamailio modules only
by Daniel-Constantin Mierla 14 Nov '11

14 Nov '11

Hello, I think the problem was a comma at the end of sipcapture module name in kstandard group inside the Makefile. If you pull the branch 3.2, it should get the fix. The master was fixed by Ovidiu Sas few minutes ago when adding xhttp_rpc module in that list. Let us know if works now. Thanks, Daniel On 11/14/11 7:01 PM, Pavel Segeč wrote: > Hello, > > I'm trying to install homer capture agent on my Kamailio server. However for > the sipcapture module there is not precompiled package for debian 32 bit > neither 64 bit. Therefore I would like make module deb package and install > it after. > > I'm doing following steps: > > git clone --depth 1 git://git.sip-router.org/sip-router kamailio > cd kamailio/ > git checkout -b 3.2 origin/3.2 > make FLAVOUR=kamailio include_modules="sipcapture" cfg > cd pkg/kamailio/deb/ > mv debian debian-orig > ln -s lenny debian > cd ../../.. > make deb > > finaly after compilation I still have following packages only, > > kamailio > kamailio-postgres-modules_3.2.0_amd64.deb > kamailio_3.2.0_amd64.changes > kamailio-presence-modules_3.2.0_amd64.deb > kamailio_3.2.0_amd64.deb > kamailio-purple-modules_3.2.0_amd64.deb > kamailio_3.2.0.dsc > kamailio-python-modules_3.2.0_amd64.deb > kamailio_3.2.0.tar.gz > kamailio-radius-modules_3.2.0_amd64.deb > kamailio-berkeley-modules_3.2.0_amd64.deb > kamailio-snmpstats-modules_3.2.0_amd64.deb > kamailio-carrierroute-modules_3.2.0_amd64.deb > kamailio-sqlite-modules_3.2.0_amd64.deb > kamailio-cpl-modules_3.2.0_amd64.deb > kamailio-tls-modules_3.2.0_amd64.deb > kamailio-ldap-modules_3.2.0_amd64.deb > kamailio-unixodbc-modules_3.2.0_amd64.deb > kamailio-lua-modules_3.2.0_amd64.deb > kamailio-utils-modules_3.2.0_amd64.deb > kamailio-memcached-modules_3.2.0_amd64.deb > kamailio-xml-modules_3.2.0_amd64.deb > kamailio-mysql-modules_3.2.0_amd64.deb > kamailio-xmlrpc-modules_3.2.0_amd64.deb > kamailio-nth_3.2.0_amd64.deb > kamailio-xmpp-modules_3.2.0_amd64.deb > kamailio-perl-modules_3.2.0_amd64.deb > > without sipcapture module. I've tried to manually edit module.lst, manually > edit pkg/kamailio/deb/debian/rules, without results, still the same. What > I'm doing bad? > > thanks > > palo > > > > _______________________________________________ > 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 Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

TOS = 0
by marion deley 14 Nov '11

14 Nov '11

Hi, I'm trying to set tos value to zero but kamailio doesn't accept it. In fact, it's DSCP field which interest me but i can't set zero. Can you explain me why is it forbidden and how can i do that ? Thanks a lot Marion

3 2

Problem with NAT and RTPProxy
by David Candamil Santos 14 Nov '11

14 Nov '11

Hi, I am using Kamailio 3.2.0 (x86_64/linux) and since some days ago I am trying to call from a network with NAT to outside. This is the configuration: Softphone (192.168.0.5) <--> Kamailio (192.168.0.3) <--> Router (192.168.0.1) <--> Softphone over smartphone All the router ports are opened and redirected to 192.168.0.3. I have installed rtpproxy 1.2.1-1 with the following configuration: --------------------------/etc/defaults/rtpproxy-------------------------------- # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 # Additional options that are passed to the daemon. EXTRA_OPTS="-l candamil.dyndns.org" ---------------------------------------------------------------------------------------- and works properly. This is the log message: ---------------------------------------------------------------------------------------------- Nov 12 20:09:13 condor kamailio[7001]: INFO: rtpproxy [rtpproxy.c:1415]: rtp proxy <udp:127.0.0.1:22222> found, support for it enabled ----------------------------------------------------------------------------------------------- Both softphones are "linphone". The configuration of the network softphone is the following: SIP identity: sip:1001@192.168.0.3 SIP proxy: sip:192.168.0.3 Direct connection to the Internet. In the softphone of the smartphone I set as proxy and domain " candamil.dyndns.org", the DNS address of my router IP. In this case, these are the symptoms: Both softphones can autentify correctly. When calling, both softphones ring, but when I answer, the voice signal is not received. This is the log error: ---------------------------------------------------------------------------------------------- Nov 12 20:23:14 condor kamailio[6991]: ERROR: rtpproxy [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy ---------------------------------------------------------------------------------------------- The same happens if I set in the network softphone that it's behind NAT, and I set the router IP. The same happens if I set in the network softphone that it's behind NAT and with a STUN server (stunserver.org). In the three cases, in the softphone over smartphone, the caller is 1001(a)192.168.0.3, and in the network softphone, the caller is 1002(a)candamil.dyndns.org. If I set in the network softphone the proxy as sip:candamil.dyndns.org, the same happens. If what I do is setting the SIP identity as sip:1001@candamil.dyndns.org, when calling form inside to ouside the network, the softphone doesn't know that the call was answered, and there is the following error in the log: ---------------------------------------------------------------------------------------------------------- Nov 12 20:53:00 condor kamailio[7306]: ERROR: <core> [parser/parse_via.c:2600]: ERROR: parse_via: invalid port number <5060ranch=z9hG4bKc 50f.b4825246.0> Nov 12 20:53:00 condor kamailio[7306]: ERROR: <core> [parser/parse_via.c:2629]: ERROR: parse_via on: <SIP/2.0/UDP 192.168.0.3:5060ranch=z 9hG4bKc50f.b4825246.0;received=87.223.138.84#015#012Via: SIP/2.0/UDP 87.223.138.84:5060;rport=5060;branch=z9hG4bK1021772993#015#012From: <sip:1001@candamil.dyndns.org>;tag=783852345#015#012To: < sip:1002@candamil.dyndns.org>#015#012Call-ID: 1644787160#015#012CSeq: 21 INVITE# 015#012User-Agent: Linphone/3.4.0 (eXosip2/unknown)#015#012Content-Length: 0#015#012#015#012> ------------------------------------------------------------------------------------------------------------ If the call is from outside to inside the network, it happens the same than in the previous cases. This is the relevant kamailio configuration: -----------------------------------------kamailio.cfg---------------------------------------------------- #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_NAT ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="candamil.dyndns.org" /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } -------------------------------------------------------------------------------------------------------------- I hope you can help me. Thanks for your time.

2 1

Can the SBC (Kamailio 3.1.x and FreeSWITCH 1.0.6+) do Upper Registration?
by edson.gomes.leme 14 Nov '11

14 Nov '11

2 1

SnmpStats problem
by Adam Kuśmirek 14 Nov '11

14 Nov '11

Hi All I've compiled kamailio 3.2.0 with snmpstats module. When trying to start the module I received the following message: 0(28197) DEBUG: <core> [sr_module.c:515]: load_module: trying to load </usr/lib/kamailio/modules_k/ snmpstats.so> 0(28197) ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/lib/kamailio/modules_k/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: boot_DynaLoader I run it on CentOS 5.7 After digging Internet, I found the solution for the problem. I had to modify Makefile in snmpstats module's directory and recompile it. I had to change BUILDAGENTLIBS env param before change: BUILDAGENTLIBS =-L$(LOCALBASE)/lib -lnetsnmpmibs -lnetsnmpagent \ -lnetsnmphelpers -lnetsnmp after change: BUILDAGENTLIBS =-L$(LOCALBASE)/lib -lnetsnmpmibs -lnetsnmpagent \ -lnetsnmphelpers -lnetsnmp \ -Wl,-E -Wl,-rpath,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE \ -L/usr/local/lib /usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a \ -L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE \ -lperl -lresolv -lnsl -ldl -lm -lcrypt -lutil -lpthread -lc \ -lwrap \ -lsensors \ -lrpmdb -lrpm After that I successfully started snmpstats module. I don't know why I didn't have compilation errors before adding this linking info, but got problem when trying to run compiled module. Can you explain it ? Do you know another way to solve the problem ? Regards Adam

2 1

a2billing
by roman dmytriv 11 Nov '11

11 Nov '11

Hello folks, Any one tried using a2billing with Kamailio and Asterisk instead of Siremis? Can you share your experiens? Regards roman

3 6

tracking ACC for a 486
by Sven Knoblich 10 Nov '11

10 Nov '11

Hi, I have a short question regarding a tm callback. I want to track a local generated ACK during a cfa. scenario: A calls B with a cfa to C B response with a 486 Busy this will be confirmed with an ACK. I saw, that TMCB_ACK_NEG_IN will only be called when a new transaction is created (t_lookup.c:t_newtran). Unfortunately in this case an ACK will be send directly when a reply will be received (t_reply.c:reply_received). Any ideas? thanks in advance, Sven

2 2

segmentation fault in init_mod_child function
by Bruno Bresciani 10 Nov '11

10 Nov '11

Hi All, Kamailio 3.1.2 terminated with signal 11, segmentation fault, in the init_mod_child function at sr_module.c file. This problem occurs only when I try loading the loadbalance.so module, it was written by me. Someone Can help me to understand why this segmentation fault was generated? There are two other module written by me and this problem doesn't happen. Kamailio's trace: 9(27616) DEBUG: db_postgres [km_pg_con.c:80]: PQsetdbLogin(0x8db2878) 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): sl 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): auth_db 9(27616) DEBUG: <core> [db.c:294]: connection 0x82f4108 found in pool 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): usrloc 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): mi_fifo 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): registrar 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): nathelper 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): permissions 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): loadbalance 9(27616) DEBUG: loadbalance [loadbalance_mod.c:271]: loadbalance: child started! (5) 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): route 0(27582) ALERT: <core> [main.c:741]: child process 27585 exited by a signal 11 0(27582) ALERT: <core> [main.c:744]: core was generated 0(27582) INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD 8(27615) INFO: <core> [main.c:807]: INFO: signal 15 received 8(27615) DEBUG: <core> [main.c:818]: Memory status (pkg): 8(27615) DEBUG: fm_status: fm_status (0x82aefc0): 8(27615) DEBUG: fm_status: heap size= 4194304 8(27615) DEBUG: fm_status: used= 236720, used+overhead=264064, free=3930240 8(27615) DEBUG: fm_status: max used (+overhead)= 299824 8(27615) DEBUG: fm_status: dumping free list: 8(27615) DEBUG: fm_status: hash = 1 fragments no.: 27, unused: 0 bucket size: 8 - 8 (first 8) 8(27615) DEBUG: fm_status: hash = 44 fragments no.: 1, unused: 0 bucket size: 352 - 352 (first 352) 8(27615) DEBUG: fm_status: hash = 74 fragments no.: 1, unused: 0 bucket size: 592 - 592 (first 592) 8(27615) DEBUG: fm_status: hash = 77 fragments no.: 1, unused: 0 bucket size: 616 - 616 (first 616) 8(27615) DEBUG: fm_status: hash = 81 fragments no.: 1, unused: 0 bucket size: 648 - 648 (first 648) 8(27615) DEBUG: fm_status: hash = 84 fragments no.: 1, unused: 0 bucket size: 672 - 672 (first 672) 8(27615) DEBUG: fm_status: hash = 113 fragments no.: 51, unused: 0 bucket size: 904 - 904 (first 904) 8(27615) DEBUG: fm_status: hash = 2056 fragments no.: 1, unused: 0 gdb bt full trace: Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/route.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/route.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/rtpproxy.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/rtpproxy.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/tls.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/tls.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/perms_db.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/perms_db.so Reading symbols from /lib/libnss_files.so.2...done. Loaded symbols for /lib/libnss_files.so.2 Reading symbols from /lib/libnss_dns.so.2...done. Loaded symbols for /lib/libnss_dns.so.2 Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 11, Segmentation fault. #0 init_mod_child (m=0xa238, rank=2) at sr_module.c:827 827 sr_module.c: Arquivo ou diretório não encontrado. in sr_module.c (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) bt full #0 init_mod_child (m=0xa238, rank=2) at sr_module.c:827 No locals. #1 0x00000000 in ?? () No symbol table info available. (gdb) Best Regards

2 2

How to limit number of calls in Kamailio proxy
by Austin Einter 10 Nov '11

10 Nov '11

This is continuation with limiting number of calls per Kamailio proxy server. I modified kamailio.cfg as below. 1. loadmodule "dialog.so" 2. modparam("dialog", "enable_stats", 1) 3. modparam("dialog", "dlg_flag", 4) Next, as dialog module I can use either active_dialogs() or $DLG_count to find number of calls present at any given point of time. However if I use active_dialogs() I get parse error when I run kamailio proxy. Next I used $DLG_count , but always I am getting value of $DLG_count as 0. In *route* section, I added code as below. if (is_method("INVITE")) { if($DLG_count > 1) { sl_send_reply("503","RESOURCE UNAVAILABLE"); exit; } else { xlog("Dialog count = $DLG_count\n"); } } Everytime a new call starts , it prints $DLG_count as 0. Kindly let me know whats going wrong here. Regards Austin

5 15

Re: [SR-Users] Kamailio Dispatcher fails in front of FreeSwitch, help (again).
by Henrik Aagaard Sørensen 10 Nov '11

10 Nov '11

Hi Daniel. I just wanted to hear if you have had any time to look at the Kamailio Dispatcher <-> FreeSwitch problem? This is NOT to rush you. I've followed the mailing list and can see how busy you are and how much involved you are in all this, so I fully respect all the help you give to this mailing list etc. Best regards, Henrik 2011/10/20 Henrik Aagaard Sørensen <henrikaagaardsorensen(a)gmail.com> > Hi Daniel. > > I've just upgraded to 3.2.0. > > This is my NEW kamailio.cfg: http://pastebin.com/XzXjf1VY > This is my dispatcher.list: http://pastebin.com/9dLS8243 > > I've also attached a ngrep where it errors with "Too many hops" with the > new version 3.2.0. > > Once again, I really appreciate all the help you give! It's remarkable > with your effort in this. > > Best regards, > Henrik > > > 2011/10/17 Daniel-Constantin Mierla <miconda(a)gmail.com> > >> I will try to look over the sip traces you sent, although it is tight >> schedule for the 3.2.0 right now -- so it may take few days. >> >> Cheers, >> Daniel >> >> >> On 10/14/11 10:07 PM, Henrik Aagaard Sørensen wrote: >> >> Yes, pbx.my-domain.com is pointing to the IP of FreeSwitch. >> sip.my-domain.com is the domain name for Kamailio. >> >> 2011/10/14 Daniel-Constantin Mierla <miconda(a)gmail.com> >> >>> >>> >>> On 10/14/11 7:06 PM, Henrik Aagaard Sørensen wrote: >>> >>> My dispatcher.list currently just contains: >>> 1 pbx.my-domain.com:5060 >>> >>> Is pbx.my-domain.com pointing to the IP address of freeswitch? What is >>> the domain name for kamailio ip address? >>> >>> Daniel >>> >>> >>> >>> I will provide a ngrep trace within an hour or two. Thank you so much >>> for helping me. I really hope to get this solved. >>> >>> 2011/10/14 Daniel-Constantin Mierla <miconda(a)gmail.com> >>> >>>> Hello, >>>> >>>> >>>> On 10/14/11 4:48 PM, Henrik Aagaard Sørensen wrote: >>>> >>>> Hi Daniel (and everyone else) again. >>>> >>>> I've just tried to add: >>>> loadmodule "path.so" >>>> modparam("path", "use_received", 1) >>>> >>>> and: >>>> add_path_received() >>>> just before the "if(!t_relay())..." in the ROUTE part. >>>> >>>> That made both devices get the error "Too many loops". >>>> >>>> what is the content of the dispatcher list file? Maybe is better to >>>> use ds_select_domain(...) and ds_next_domain() -- cannot say it since I >>>> haven't seen full sip trace, but maybe the problem is that freeswitch is >>>> sending back if the r-uri address is kamailio's address. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> >>>> Moving the add_path_received() into the part: >>>> if (nat_uac_test("19")) { if (method=="REGISTER") { >>>> fix_nated_register(); add_path_received(); >>>> made the same issues as before loading the path.so module. >>>> >>>> 2011/10/14 Henrik Aagaard Sørensen <henrikaagaardsorensen(a)gmail.com> >>>> >>>>> Hi Daniel (and everyone else). >>>>> >>>>> This is a tshark trace (is that as good as ngrep?) from Kamailio when >>>>> there are too many loops: http://pastebin.com/yRjduB0s >>>>> In this trace I got the note that 253 packets was dropped (and >>>>> therefor not in the trace). >>>>> >>>>> And this is a trace when it times out: http://pastebin.com/KT0bJzEU >>>>> >>>>> I've actually looked at the PATH module, and tried several things. And >>>>> I know this type of question is rather newbee-like, but can you help me out >>>>> with the PATH module and how to integrate it correctly? >>>>> >>>>> I really do appreciate all the help! >>>>> >>>>> >>>>> 2011/10/14 Daniel-Constantin Mierla <miconda(a)gmail.com> >>>>> >>>>>> Hello, >>>>>> >>>>>> when using the load balancer in front of the registrar, you have to >>>>>> use PATH extensions for SIP (see path module in kamailio) in order to be >>>>>> able to cope with nat. >>>>>> >>>>>> Then, what would be helpful, will be a ngrep trace taken on kamailio >>>>>> server for failed calls. >>>>>> >>>>>> Cheers, >>>>>> Daniel >>>>>> >>>>>> >>>>>> On 10/14/11 4:13 PM, Henrik Aagaard Sørensen wrote: >>>>>> >>>>>> I've asked a lot of question about Kamailio Dispatcher in front of >>>>>> FreeSwitch in this mailing list. So I'm sorry that I'm still "spamming", >>>>>> but it still fails. >>>>>> >>>>>> This is my status so far. >>>>>> >>>>>> When connecting 2 devices directly to FreeSwitch everything works. >>>>>> When connecting to Kamailio, some calls fails with "Timeout" and some fails >>>>>> with "Too many loops". >>>>>> >>>>>> This is my kamailio.cfg: http://pastebin.com/KEuZmikh >>>>>> >>>>>> When connected directly to FreeSwitch where everything works, this is >>>>>> how my clients are registrered in FreeSwitch: >>>>>> http://pastebin.com/prkjkb7g >>>>>> >>>>>> When connected to Kamailio where nothing works, this is how my >>>>>> clients are registrered in FreeSwitch: http://pastebin.com/b5x6NZyv >>>>>> >>>>>> 215.197.145.132 (sip.my-domain.com) is my Kamailio. >>>>>> pbx.my-domain.com is my FreeSwitch. >>>>>> 96.224.14.164 is my IP at home, where my 2 devices are from. >>>>>> >>>>>> Can anyone help me out? I'm kinda lost at this point. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> 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.com >>>>>> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://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.com >>>> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://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.com >>> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://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.com >> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> >

2 2

uacreg issue
by Lucas Alvarez 09 Nov '11

09 Nov '11

Hi, after following the steps of this guide: http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html, poulating the uacreg table, loading the uac module and restarting kamailio I didn't see any REGISTER packet sent out to the sip provider, is something missing? How can I know if the values from the table are being loaded? I would appreciate any kind of help. Regards, Lucas Alvarez

1 0

negative www_authorize result codes
by Juha Heinanen 09 Nov '11

09 Nov '11

auth_db/README tells this about www_authorize() negative result codes: Negative codes may be interpreted as follows: * -1 (generic error) - some generic error occurred and no reply was sent out; * -2 (invalid password) - valid user, but wrong password; * -3 (invalid user) - authentication user does not exist. whereas in auth_radius i return: Negative result codes may be interpreted as follows: * -7 (internal error) - some internal error occurred (see syslog); * -6 (nonce reused) - nonce is used more than once; * -5 (no credentials) - credentials were not found in request; * -4 (stale nonce) - stale nonce; * -3 (authorization failed) - RADIUS responded with Access Reject which may be, for example, due to user not found or wrong password; * -1 (error) - some error occured during authorization (see syslog); since both modules depend on the same auth module, why does www_authorize return the full set of negative result codes or is its README out of date? -- juha

2 2

uacreg question
by Lucas Alvarez 08 Nov '11

08 Nov '11

Hi, I have properly populate the table "uacreg" as says here: http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html and I'm loading the uac module, is there anything I could be missing? Because I'm not seeing any register packet being sent to my sip provider. Thanks in advance, Lucas Alvarez

2 3

how to give integer avp name in pseudo variable?
by Juha Heinanen 08 Nov '11

08 Nov '11

i have not been able to figure out, how to give integer avp name in a pseudo variable. test $var(name) = "i:1000"; $avp($var(name)) = "value"; xlog("L_INFO", "avp i:1000 has value <$avp(i:1000)>\n"); xlog("L_INFO", "avp s:i1000 has value <$avp(s:i:1000)>\n"); produces to syslog Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp i:1000 has value <<null>> Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp s:i1000 has value <value> that is, $var(name) value i:1000 is interpreted as string name even when its syntax conforms to integer name. is this is a bug or how should value of $var(name) look like? -- juha

2 3

how to give integer avp name in pseudo variable?
by Juha Heinanen 08 Nov '11

08 Nov '11

i have not been able to figure out, how to give integer avp name in a pseudo variable. test $var(name) = "i:1000"; $avp($var(name)) = "value"; xlog("L_INFO", "avp i:1000 has value <$avp(i:1000)>\n"); xlog("L_INFO", "avp s:i1000 has value <$avp(s:i:1000)>\n"); produces to syslog Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp i:1000 has value <<null>> Nov 8 13:17:23 sip /usr/sbin/sip-proxy[8642]: INFO: avp s:i1000 has value <value> that is, $var(name) value i:1000 is interpreted as string name even when its syntax conforms to integer name. is this is a bug or how should value of $var(name) look like? -- juha

1 0

Conditional uacreg
by Alex Balashov 08 Nov '11

08 Nov '11

Is there some way by which credentials can be loaded into the uacreg table, but not all of them used at once? The problem I have is a need to make these registrations active and inactive periodically, whereas the registration facility that the uac module provides is fully managed/largely unattended. Using $uac_req(...) is not an option because I need to capture the failure replies (which is not supported in TM) in order to do digest challenge authentication. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 7

SQL Server
by Jonny Klompas 08 Nov '11

08 Nov '11

Can Kamailio send routes to and /or receive routes from SQL Server? Is there any way for Kamailio to connect to SQL Server in general? Thanks Jonny

3 2

TOS=0 in kamailio.cfg
by DELEY Marion 07 Nov '11

07 Nov '11

Hi, I'm trying to set tos value to zero but kamailio doesn't accept it. In fact, it's DSCP field which interest me but i can't set zero. Can you explain me why is it forbidden and how can i do that ? Thanks a lot Marion

1 0

Re: [SR-Users] Fwd: Kamailio as SBC / LB Media-Servers farm
by Klaus Darilion 05 Nov '11

05 Nov '11

On 04.11.2011 10:57, Sammy Govind wrote: > SORRY for PM but this is not getting through to the users List due to size. I have cc-ed the list. > > ---------- Forwarded message ---------- > From: *Sammy Govind* <govoiper(a)gmail.com <mailto:govoiper@gmail.com>> > Date: Fri, Nov 4, 2011 at 1:55 PM > Subject: Re: [SR-Users] Kamailio as SBC / LB Media-Servers farm > 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>> > > > Klaus: mhomed did work indeed - no need to force_send_socket() now :) Thanks > > I'm progressing abit. > > RTP-Proxy was run in bridged mode like this. > > *rtpproxy -l 109.XX.XXX.212/192.168.2.29 <http://192.168.2.29> -s > udp:localhost:22222 -F -d DBUG* See description of i,e flag: http://kamailio.org/docs/modules/3.2.x/modules/rtpproxy.html#id2535618 Thus in your setup 109.... is the "internal" interface and 192.... is the "external" interface. That's mixed. Thus I advice you to switch the order of IP addresses so that internal is 192... and external is 109... This is essential to use the i,e flags correctly. > I'm getting RTPstats published on Kamailio/RTPproxy syslog. > > I am using Echo() application in asterisk. So I've observed that my > audio went to asterisk and asterisk did indeed send audio back to the > RTPproxy assigned ports BUT the RTPproxy isn't forwarding the Audio back > to the client or may-be the audio is lost somewhere in between > RTPproxy<--->Client. /[See Wirshark Flow-Diagram taken > on-Kamailio/RTPproxy Server]/ In your trace you see, that RTP does not use the internal IP address, but only the external IP address. Thus I suspect that the RTP packets from Asterisk to the public rtpproxy interface do not even arrive at rtpproxy. > > The # DTMF signal sent from Client phone reached asterisk safely and > terminated the Echo() application. > > I did the ip_forwarding thingie too.-- somebody mentioned to do this if > using RTPproxy in bridged mode. This is bullshit. People seem to not understand IP at all. rtpproxy is forwarding in user mode - it receives on a UDP socket and sends on a UDP socket. Thus, kernel based forwarding is not involved when using rtpproxy. You can disable IP forwarding again. > #echo "1" >> /proc/sys/net/ipv4/ip_forward > #sysctl -p > > > Here are the corresponding Logs from Kamailio and rtpproxy. The thing is quite easy. Swap the IP addresses when starting rtpproxy and then configure Kamailio to call rtpproxy with the proper i,e flags (as described in my previous email). For example: if (has_body("application/sdp")) { if ($si == "192.168.2.75") { # message from internal to external rtpproxy_manage("ie"); } else { # message from external to internal rtpproxy_manage("ei"); } } Put above code block in the main route block and also in a reply route which will be activated (or in the default reply route). Put some xlog statements around the rtpproxy calls to verify the proper command is called. regards Klaus

2 1

Kamailio 3.2 Acc Runtime error
by Nathaniel L Keeling 05 Nov '11

05 Nov '11

I am getting this error when I start kamailio 3.2 on Solaris 10. Any direction on solving this would be greatly appreciated. ERROR: load_module: could not open module </opt/kamailio-3.2/lib64/kamailio/modules_k/acc.so>: ld.so.1: kamailio: fatal: relocation error: file /opt/kamailio-3.2/lib64/kamailio/modules_k/acc.so: symbol timersub: referenced symbol not found Thanks Nathaniel

2 2

Directory missing compiling dbtext module
by Pedro Antonio Vico Solano 05 Nov '11

05 Nov '11

3 2

CANCEL is not forwarded, if callee does not respond
by Mino Haluz 04 Nov '11

04 Nov '11

Hi, the scenario is like this: - user1 makes a call to user2 - proxy sends receives INVITE and sends it to user2 - user2 does not respond at all (firewall for example) - user1 after 5 seconds hangs up the call with CANCEL - proxy sends to user1 200 canceling CANCEL is not forwarded to user2. Ok this behaviour is maybe ok, but after 30 seconds Request timed out is sent to user1 (fr_inv_timer possibly hit). So if I am thinking well, first CANCEL did not match the transaction ? Why ? It this behaviour ok or do I have to search for something wrong in the config? Thanks, Mino

2 1

Kamailio as SBC / LB Media-Servers farm
by Sammy Govind 04 Nov '11

04 Nov '11

Hello List, I've successfully configured Kamailio with a farm of asterisk servers in L/B | F/O mode. The topology is something like below: User<-->Kamailio<-->Asterisk-Servers Both kamailio and Asterisk serevrs are on public IPs *The problem:* Kamailio and Asterisk Servers need to be on Public IPs in order to fully handle NAT/media related issues. Whereas what I require is to have two interfaces at kamailio server, one is Public other is Private and All the Asterisk Servers should be on LAN/Private IPs. I've tried making a scenario as per my requirement (attached image) but there are some serious NAT and media issues. Can anyone help me how to handle NAT'd Asterisk servers behind Kamailio? Thanks and Best Regards. Sammy

4 7

Kamailio as SBC / LB Media-Servers farm
by Sammy Govind 04 Nov '11

04 Nov '11

Hello List, I've successfully configured Kamailio with a farm of asterisk servers in L/B | F/O mode. The topology is something like below: User<-->Kamailio<-->Asterisk-Servers Both kamailio and Asterisk serevrs are on public IPs *The problem:* Kamailio and Asterisk Servers need to be on Public IPs in order to fully handle NAT/media related issues. Whereas what I require is to have two interfaces at kamailio server, one is Public other is Private and All the Asterisk Servers should be on LAN/Private IPs. I've tried making a scenario as per my requirement (attached image) but there are some serious NAT and media issues. Can anyone help me how to handle NAT'd Asterisk servers behind Kamailio? Thanks and Best Regards. Sammy

1 0

Issue with version table
by Lucas Alvarez 03 Nov '11

03 Nov '11

Hi, I've enabled xcap server in kamailio 3.1.3 and I'm stuck with this error: 0(5073) ERROR: <core> [db.c:408]: invalid version 0 for table subscriber found, expected 6 (check table structure and table "version") I've already checked the version table and the table_name = 'subscriber' has table_version = 6. Any help will be appreciated. Regards. Lucas

2 3

Dispatcher Confusion (v3.2.0)
by Asgaroth 03 Nov '11

03 Nov '11

Hi All, Just been doing some testing with Kamailio 3.2 and the dispatcher module, and have some strange behaviour, can we confirm if it is expected behaviour or a bug of some sort. It appears that I can only set the dspatcher state via fifo command once, and not reset it again, here is an example: version: kamailio 3.2.0 (i386/linux) 94d3b8 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: 94d3b8 compiled on 12:34:43 Oct 20 2011 with gcc 4.1.2 Here we start with the dispatcher list as it was pulled from the database and currently loaded in memory. # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Now I try to set one gateway inactive: # kamctl fifo ds_set_state i 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= All normal so far, now if I try to set the same gateway active again, the state still stays inactive, i dont seem to be able to change the state back to active. The same thing happens if I attempt to set this state to disabled and then try to set it back to active afterwards. The state does not change from what the first attempt was. # kamctl fifo ds_set_state a 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Now I can try to set the state to inactive probing, that works. # kamctl fifo ds_set_state ip 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IP priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Now if I try to disable the same destination: # kamctl fifo ds_set_state d 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IP priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= State still stays in Inactive/Probing mode. Now if I try to set the same gateway to active/probing mode, I cant: # kamctl fifo ds_set_state ap 1 sip:1.1.1.1:10001 # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=IP priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= The only way for me to get the state back to active after the above is to do a reload on the dispatch table: # kamctl dispatcher reload # kamctl dispatcher dump SET_NO:: 2 SET:: 1 URI:: sip:1.1.1.1:10001 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10000 flags=AX priority=0 attrs= SET:: 2 URI:: sip:1.1.1.1:10003 flags=AX priority=0 attrs= URI:: sip:1.1.1.1:10002 flags=AX priority=0 attrs= Is the dispatcher expected to behave as above? I'm still trying to understand the different states assuming the following options are set in the configuration file: modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_probing_mode", 0) AX = Active/No Probing (*) Gateway is used in ds_select_*, ds_next_* (*) No OPTIONS/INFO messages are sent to destination to check alive state This appears to be the normal operating state with no issues detected. AP = Active/Probing (*) Gateway is NOT used in ds_select_*, ds_next_* (*) OPTIONS/INFO messages ARE sent to destination to check alive state This appears to be the state taken when a gateway times out and ds_mark_dst("p") is set in failure route, the gateway is "pinged" while down and set into AX mode once a response is recieved. IX = Inactive/No Probing (*) Gateway is NOT used in ds_select_*, ds_next_* (*) No OPTIONS/INFO messages sent to destination This appears to be the state taken when a gateway times out and if ds_mark_dst("i") is set in failure route, this gateway is set inactive and no "pinging" is performed. IP = Inactive/Probing (*) Gateway is NOT used in ds_select_*, ds_next_* (*) OPTIONS/INFO messages ARE sent to destination I'm not sure when this state would be reached unless it is set by fifo command. The ds_mark_dst function only allows "a","i" or "p". So the probing method would only become active is ds_probing_mode = 1. In my case it is 0, so I'm not sure if I would ever reach this state unless I forced it. What would happen here is the gateway came back alive once in "IP" state, would it go back to "AX" state? DX = Disabled/No Probing (*) Gateway is administratively set to down This is where manual intervention has occured and the destination has been set to administrativly down, for example, work being carried out on the destination gateway. All these states make sense, for the most part, I'm a little unsure of how IP/IX is meant to work unless you manually set it to "inactive" in your failure route. The main concern though is the the fifo commands dont appear to be working as expected, I can only set the state for a destination manually once, and not again after that, unless I perform a dispatcher reload. Has anyone else experienced this behaviour? Any comments/suggestions explanations are most welcome. Thanks

3 32

Kamailio dinner next week in Cape Town, South Africa
by Daniel-Constantin Mierla 03 Nov '11

03 Nov '11

Hello, it happens to come across Cape Town for few days and I was wondering if there are folks using our project or doing VoIP in general that want to meet up for a dinner. The date for dinner is planned for next Tuesday, November 8, 2011. Place will be decided once I know if there are people coming and how many. Likewise with such events done in the past, it is going to be a dinner open for everyone that wants to join, each participant paying the drinks and food for himelf/herself. The goal is to meet and discuss latest trends in IP communication and what's up lately around our project. Here are some pictures from two past similar events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ So, if there are people on these mailing lists in the area of Cape Town next Tuesday and want to meet, please reply to me till Friday evening -- I will follow up privately to each person that is going to join with details about location and start time. Looking forward to meet some of you very soon! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

TOS=0 in kamailio.cfg
by DELEY Marion 03 Nov '11

03 Nov '11

Hi, I'm trying to set tos value to zero but kamailio doesn't accept it. In fact, it's DSCP field which interest me but i can't set zero. Can you explain me why is it forbidden and how can i do that ? Thanks a lot Marion

1 0

is there help on mi commands in sercmd?
by Juha Heinanen 03 Nov '11

03 Nov '11

sercmd has help that lists the commands, one of which is mi. is there a way to list all mi sub-commands? i have tried ip-proxy_ctl> help mi Execute MI command sip-proxy_ctl> mi help error: 500 - command not available -- juha

2 3

extra _ctl in ctl file
by Juha Heinanen 03 Nov '11

03 Nov '11

i just noticed that my sip-proxy_ctl (alias kamailio_ctl) command has stopped working. it now adds an extra _ctl to the socket file name: # sip-proxy_ctl ERROR: connect_unix_sock: connect(/tmp/sip-proxy_ctl_ctl): No such file or directory [2] any idea where that comes from or it is only me, who has this problem? i have not changed anything the command used to work fine earlier. -- juha

2 4

Dialog / BYE
by Brandon Armstead 02 Nov '11

02 Nov '11

Hello, I am attempting to tear down a call with a BYE packet generated externally (kind of similar to Kamailio fifo dlg_end_dlg). Let me describe what I am trying to do in more depth and then I will continue to tell you the problem I think I am experiencing. [PSTN SIP Proxy] -> [CORE SIP Proxy] -> [REGISTRAR] -> [UAC] So the above layout is the normal call flow / structure of calls (incoming when originating from pstn) (outgoing when originating from uac). I then have an "external" host - I am attempting to generate a BYE to [CORE SIP Proxy] and have it go both directions [PSTN] + [UAC]. So far I am able to get the call to tear down in only a single direction (only kill call with PSTN) or (only kill call with UAC). I have not been able to kill both legs of the call. I have tried sending a single BYE and using some dlg_bye("all") magic. - does not work, only kills one leg of the call. I have tried sending a BYE message to SIP Proxy for each leg (two BYE). - does not work, only kills one leg of the call. If I alternate the later method's BYE packet and only send A or B and not both Leg A or B will be killed... I think I am running into a race condition where the dialog is destroyed before the 2nd BYE packet is processed. My Question - what am I doing wrong - or what is the best method to tackle this task? Thank you for all of your help and time in advance. Sincerely, Brandon Armstead

2 9

Kamailio3.2 debian repo fails
by Benjamin Henrion 02 Nov '11

02 Nov '11

Hi, I have the following /etc/apt/sources.list: ======================================================== $ cat /etc/apt/sources.list deb http://ftp.debian.org/debian squeeze main contrib non-free deb-src http://ftp.debian.org/debian squeeze main contrib non-free deb http://deb.kamailio.org/kamailio32 squeeze main deb-src http://deb.kamailio.org/kamailio32 squeeze main ======================================================== Which generates the following error: ======================================================== $ apt-get update Ign http://deb.kamailio.org squeeze Release.gpg Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en_US Ign http://deb.kamailio.org/kamailio32/ squeeze/main Translation-en_US.UTF-8 Ign http://deb.kamailio.org squeeze Release Ign http://deb.kamailio.org squeeze/main Sources Hit http://ftp.debian.org squeeze Release.gpg Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en_US Ign http://deb.kamailio.org squeeze/main amd64 Packages Ign http://deb.kamailio.org squeeze/main Sources Ign http://ftp.debian.org/debian/ squeeze/contrib Translation-en_US.UTF-8 Ign http://ftp.debian.org/debian/ squeeze/main Translation-en Ign http://ftp.debian.org/debian/ squeeze/main Translation-en_US Ign http://ftp.debian.org/debian/ squeeze/main Translation-en_US.UTF-8 Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en_US Ign http://ftp.debian.org/debian/ squeeze/non-free Translation-en_US.UTF-8 Hit http://ftp.debian.org squeeze Release Ign http://deb.kamailio.org squeeze/main amd64 Packages Err http://deb.kamailio.org squeeze/main Sources 404 Not Found Err http://deb.kamailio.org squeeze/main amd64 Packages 404 Not Found Hit http://ftp.debian.org squeeze/main Sources Hit http://ftp.debian.org squeeze/contrib Sources Hit http://ftp.debian.org squeeze/non-free Sources Hit http://ftp.debian.org squeeze/main amd64 Packages Hit http://ftp.debian.org squeeze/contrib amd64 Packages Hit http://ftp.debian.org squeeze/non-free amd64 Packages W: Failed to fetch http://deb.kamailio.org/kamailio32/dists/squeeze/main/source/Sources.gz 404 Not Found W: Failed to fetch http://deb.kamailio.org/kamailio32/dists/squeeze/main/binary-amd64/Packages… 404 Not Found E: Some index files failed to download, they have been ignored, or old ones used instead. ======================================================== Who is responsible for the debian packaging? Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-4148403 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

3 6

Re: [SR-Users] Siremis: cannot enter domain in "New Subscriber"window
by Uwe Andersen 02 Nov '11

02 Nov '11

Hi Hugh, it works now. Thank you! Uwe > -----Ursprüngliche Nachricht----- > Von: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org]Im Auftrag von > Hugh Waite > Gesendet: Mittwoch, 2. November 2011 12:52 > An: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > UsersMailing List > Betreff: Re: [SR-Users] Siremis: cannot enter domain in "New > Subscriber"window > > > Hi Uwe, > > Domains are configured in Server Services->Domain List. > These are the > domains which the kamailio server is responsible for. > When they have been entered there, they will appear in the drop-down > list when adding users. > > Regards, > Hugh > > On 02/11/2011 11:33, Uwe Andersen wrote: > > Hi, > > > > I want to add a new user. So I choose "add" from the > "Subscriber Management" page. > > On the "New Subscriber" page I can enter username, > password, email-address, etc. > > But the line for "domain" seems to be an empty dropdown entry. > > I cannot choose, add or edit anything. > > As a result I cannot save the new user because Siremis > complains about the empty domain field. > > > > After adding a user by command line("kamctl add ...") the > user appears on the management page. > > If I edit that user via Siremis the line for domain is not > a dropdown but a "normal" text field. > > > > Can somebody explain this? > > > > BR, > > Uwe > > > > P.S.: I'm using Siremis 2.0 and Kamailio 3.2.0. > > > > _______________________________________________ > > 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 > > > -- > -- > Hugh Waite > Senior Design Engineer > Crocodile RCS Ltd. > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

1 0

Siremis: cannot enter domain in "New Subscriber" window
by Uwe Andersen 02 Nov '11

02 Nov '11

Hi, I want to add a new user. So I choose "add" from the "Subscriber Management" page. On the "New Subscriber" page I can enter username, password, email-address, etc. But the line for "domain" seems to be an empty dropdown entry. I cannot choose, add or edit anything. As a result I cannot save the new user because Siremis complains about the empty domain field. After adding a user by command line("kamctl add ...") the user appears on the management page. If I edit that user via Siremis the line for domain is not a dropdown but a "normal" text field. Can somebody explain this? BR, Uwe P.S.: I'm using Siremis 2.0 and Kamailio 3.2.0.

2 1

FW: App_python Module
by Muhammad Nasir Bhutta 02 Nov '11

02 Nov '11

Hi, I am a new user of Kamailio, and I am working on enabling the App_Python module in Kamailio.. While searching on google, I came acroos your email on the forum where you people are trying to patch the App_Python module.. I am wandering whether you can please help a bit .. 1- How to do the configuration to enable the App_python module ? .. At the moment, I am just adding the following line in the configuration file, loadmodule "app_python.so" and just correcting the path to handler.py to the right directory in the code for script path. 2- What output we should expect when we enable the app_python module ? .. I have seen the handler.py script written in python which is printing some sip message information ? .. 3- How the SIP Message structure is being passed to handler.py ? ... How we are accessing the Router module in handler.py script and LM_ERR function from Kamailio ? .. Any help in this regards shall be really helpful .. In any case, whether you can solve this issue or not, please reply the email.. With many thanks.. Kind Regards, Nasir

2 3

sdpops and dynamic codec ids
by Juha Heinanen 01 Nov '11

01 Nov '11

i made call sdp_keep_codecs_by_name("PCMU,PCMA,speex"); on an invite request and it didn't keep speex although it was in the sdp: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): sems 1 1 IN IP4 192.98.102.10 Session Name (s): sems Connection Information (c): IN IP4 192.98.102.10 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 40000 RTP/AVP 101 102 8 0 3 2 9 96 97 98 99 100 Media Attribute (a): rtpmap:101 iLBC/8000 Media Attribute (a): rtpmap:102 speex/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:2 G721/8000 Media Attribute (a): rtpmap:9 g722/8000 Media Attribute (a): rtpmap:96 telephone-event/8000 Media Attribute (a): rtpmap:97 G726-32/8000 Media Attribute (a): rtpmap:98 G726-24/8000 Media Attribute (a): rtpmap:99 G726-40/8000 Media Attribute (a): rtpmap:100 G726-16/8000 it is so that sdpops module does not support codecs, whose ids are dynamic, but only codecs with static ids? -- juha

2 14

enabling app_python module in Kamailio
by Muhammad Nasir Bhutta 01 Nov '11

01 Nov '11

Hi, I am trying to enable the app_python module in Kamailio to run a simple python script externally parallel with Kamailio proxy, While I try to load the module "app_python.so" in configuration file, the Kamailio server fails to start.. I can't see what is the problem .. The steps, I am following are that: - Script written in python saved in home directory. - Loadmodule "app_python.so" statement in configuration file. - Modparam("app_python", "script_name", "location of script") - Python_exec("function", "parameters list") .. I have also tried to check the Kamailio configuration option, It gives following warning, 0(2942) WARNING: <core> [sr_module.c:584]: /usr/local/lib64/kamailio/modules/app_python.so: exports dlflags interface is deprecated and it will not be supported in newer versions; consider using mod_register() instead I am wandering if any one can please help in this matter, I shall be very thankful .. With many thanks .. Kind Regards, Nasir Muhammad Nasir Bhutta Solutions Architect Britannic Technologies Ltd t: +44 1483 544812 f: +44 845 0501001 mbhutta(a)btlnet.co.uk<mailto:mbhutta@btlnet.co.uk> www.btlnet.co.uk<http://www.btlnet.co.uk> [cid:image001.jpg@01CC881F.1EA78300]<http://www.linkedin.com/company/britannic-technologies/products>[cid:image002.jpg@01CC881F.1EA78300]<http://www.twitter.com/BritannicTech>[cid:image003.jpg@01CC881F.1EA78300]<http://www.facebook.com/pages/Britannic-Technologies/202957249732198>[cid:image004.jpg@01CC881F.1EA78300]<http://www.youtube.com/user/BritannicTech> ________________________________ [cid:image007.jpg@01CC881F.4A995BF0]<http://www.btlnet.co.uk/> Watch the Telegraph Business Club video about Britannic at our educational TV site - britannictech.tv<http://britannictech.tv> Our 8th Annual Convergence Summit is being held at Silverstone on 17th November 2011. Book your place now<http://www.btlnet.co.uk/news--events/2011/convergence-summit-2011/register> and discover the world of Hosting and Cloud Services. ________________________________ This email and any files transmitted with it are confidential and intended for the addressee only. If you are not the intended recipient please notify the sender and delete this email. The views expressed in this email are personal, and do not constitute a commitment by Britannic Technologies Ltd unless specified by a separate, written agreement. Registered Office: Britannic House, Merrow Business Park, Guildford, Surrey GU4 7WA. Registered in England no. 2090797. ________________________________

3 4

app_python module
by Muhammad Nasir Bhutta 01 Nov '11

01 Nov '11

Hi, I have successfully installed the app_python module for the default example of handler.py .. Now, I need to see the output of handler.py,.. I have following questions: 1- What is the output of this module ? does it prints the specific message information extracted from SIP message ? 2- How to write modparam and python_exec in configuration file ? I have tried to write modparam("app_python", "mod_init") but, it gives error and says to use mod_register function instead and I don't know more about mod_register function. 3- I have also tried to use python_exec("child_init", 2) , The Kamailio starts successfully.. When, I connect a soft client to Kamailio, then nothing happens ? How to see the output of handler.py script . Any help in this regards shall be really appreciated .. With thanks. Kind Regards, Nasir

1 0

script local flags
by Uri Shacked 01 Nov '11

01 Nov '11

Hi, I am using the script local flags. i set a flag like this "setsflag(6)" in order to use it afterwards. i notice that this flag is relevent only for the request i am dealing with right now. because after the reply the flag seems to reset without me doing that doing "issflagset(6)" and getting nowhere :-) am i right or am i using it wrong?

2 1

FW: App_python Module
by Muhammad Nasir Bhutta 01 Nov '11

01 Nov '11

Hi, I am a new user of Kamailio, and I am working on enabling the App_Python module in Kamailio.. While searching on google, I came acroos your email on the forum where you people are trying to patch the App_Python module.. I am wandering whether you can please help a bit .. 1- How to do the configuration to enable the App_python module ? .. At the moment, I am just adding the following line in the configuration file, loadmodule "app_python.so" and just correcting the path to handler.py to the right directory in the code for script path. 2- What output we should expect when we enable the app_python module ? .. I have seen the handler.py script written in python which is printing some sip message information ? .. 3- How the SIP Message structure is being passed to handler.py ? ... How we are accessing the Router module in handler.py script and LM_ERR function from Kamailio ? .. Any help in this regards shall be really helpful .. In any case, whether you can solve this issue or not, please reply the email.. With many thanks.. Kind Regards, Muhammad Nasir Bhutta Solutions Architect Britannic Technologies Ltd t: +44 1483 544812 f: +44 845 0501001 mbhutta(a)btlnet.co.uk<mailto:mbhutta@btlnet.co.uk> www.btlnet.co.uk<http://www.btlnet.co.uk> [cid:image001.jpg@01CC9895.3EC504B0]<http://www.linkedin.com/company/britannic-technologies/products>[cid:image002.jpg@01CC9895.3EC504B0]<http://www.twitter.com/BritannicTech>[cid:image003.jpg@01CC9895.3EC504B0]<http://www.facebook.com/pages/Britannic-Technologies/202957249732198>[cid:image004.jpg@01CC9895.3EC504B0]<http://www.youtube.com/user/BritannicTech> ________________________________ [cid:image005.jpg@01CC9895.3EC504B0]<http://www.btlnet.co.uk/> Watch the Telegraph Business Club video about Britannic at our educational TV site - britannictech.tv<http://britannictech.tv> Our 8th Annual Convergence Summit is being held at Silverstone on 17th November 2011. Book your place now<http://www.btlnet.co.uk/news--events/2011/convergence-summit-2011/register> and discover the world of Hosting and Cloud Services. ________________________________ This email and any files transmitted with it are confidential and intended for the addressee only. If you are not the intended recipient please notify the sender and delete this email. The views expressed in this email are personal, and do not constitute a commitment by Britannic Technologies Ltd unless specified by a separate, written agreement. Registered Office: Britannic House, Merrow Business Park, Guildford, Surrey GU4 7WA. Registered in England no. 2090797. ________________________________

1 0

flags and masks in carrierroute
by Uri Shacked 01 Nov '11

01 Nov '11

Hi, I did not understand the explanation of the use of flags and mask. I understand flags are for matching and mask is doing something. But, what exactly does the mask do? How do i see or use it? And regarding the flags... i did "setflag(16)" before the CR function. Set the DB col as needed, and no match was found... Any help here?

1 0

siremis: login error / http error 404
by Uwe Andersen 01 Nov '11

01 Nov '11

Hi Ramona, You hit it! After enabling "rewrite" it is working now. Thank you very much! Uwe > > I have installed Siremis 2.0.0 and Kamaillio 3.2.0 on a > OpenSuse 11.4 system. > > After the Siremis Web Installation Wizard finished the > setup the login failed with a http error 404. > > > > The Apache2.0 error_log shows: > > [Mon Oct 31 11:09:49 2011] [error] [client ::1] File does > not exist: /var/www/siremis-2.0.0/siremis/system, referer: > http://localhost/siremis/index.php/user/login > > > > The Apache2.0 access_log shows: > > ::1 - - [31/Oct/2011:11:09:36 +0100] "GET > /siremis/bin/antispam_image.php?form=user.form.LoginForm&name= > antispam&length=6&level=2?rand=1768032447 HTTP/1.1" 200 240 > "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 > (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" > > ::1 - - [31/Oct/2011:11:09:48 +0100] "POST > /siremis/bin/controller.php?F=RPCInvoke&P0=[user.form.LoginFor > m]&P1=[Login]&__this=btn_login:onclick&jsrs=1 HTTP/1.1" 200 > 214 "http://localhost/siremis/index.php/user/login" > "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" > > ::1 - - [31/Oct/2011:11:09:49 +0100] "GET > /siremis/system/general_default HTTP/1.1" 404 1331 > "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 > (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" > > > > Is there anybody how can help me out of this? > > do you have mod_rewrite enabled in your apache2 server?

1 0

Memory Leak on DB Errors?
by Klaus Darilion 31 Oct '11

31 Oct '11

Hi! I recently had a problem with Kamailio 3.1.4 (provided Debian packages): I had some DB problems (missing tables). Thus, the timer module failed to insert the statistics (for siremis): ERROR: db_mysql [km_dbase.c:120]: driver error on query: Table 'kamailio.statistics_tmx' doesn't exist ERROR: <core> [db_query.c:130]: error while submitting query ERROR: sqlops [sql_api.c:217]: cannot do the query This happened for some time (weeks?), other DB queries were unaffected. Then, suddenly Kamailio ran out of memory: ERROR: <core> [sip_msg_clone.c:506]: ERROR: sip_msg_cloner: cannot allocate memory ERROR: tm [t_lookup.c:1338]: ERROR: new_t: out of mem: ERROR: tm [t_lookup.c:1478]: ERROR: t_newtran: new_t failed ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It do not think it is a load problem as the server is more or less idle. May it be a memory leak due to wrong error handling? Where there any fixes recently? Thanks Klaus

3 16

current shm memory status
by Klaus Darilion 31 Oct '11

31 Oct '11

Hi! Is there a way to get the status of shm memory? (used memory vs. free memory) Thanks Klaus

2 4

siremis: login error / http error 404
by Uwe Andersen 31 Oct '11

31 Oct '11

Hi, I have installed Siremis 2.0.0 and Kamaillio 3.2.0 on a OpenSuse 11.4 system. After the Siremis Web Installation Wizard finished the setup the login failed with a http error 404. The Apache2.0 error_log shows: [Mon Oct 31 11:09:49 2011] [error] [client ::1] File does not exist: /var/www/siremis-2.0.0/siremis/system, referer: http://localhost/siremis/index.php/user/login The Apache2.0 access_log shows: ::1 - - [31/Oct/2011:11:09:36 +0100] "GET /siremis/bin/antispam_image.php?form=user.form.LoginForm&name=antispam&length=6&level=2?rand=1768032447 HTTP/1.1" 200 240 "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" ::1 - - [31/Oct/2011:11:09:48 +0100] "POST /siremis/bin/controller.php?F=RPCInvoke&P0=[user.form.LoginForm]&P1=[Login]&__this=btn_login:onclick&jsrs=1 HTTP/1.1" 200 214 "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" ::1 - - [31/Oct/2011:11:09:49 +0100] "GET /siremis/system/general_default HTTP/1.1" 404 1331 "http://localhost/siremis/index.php/user/login" "Mozilla/5.0 (X11; Linux i686; rv:7.0.1) Gecko/20100101 Firefox/7.0.1" Is there anybody how can help me out of this? Regards, Uwe Andersen

2 1

Re: [SR-Users] uac module - error searching pvar "uac_req"
by Daniel-Constantin Mierla 31 Oct '11

31 Oct '11

On 10/31/11 2:09 PM, Vladimir Broz wrote: > ah, that was the problem :( ok. Don't forget to keep the mailing list always cc-ed, so people reading the archive for similar issues will find the conclusion, whether the hints were solving or not the problem. Cheers, Daniel > > Thanks Daniel! > -Vlada > > On 10/31/2011 01:37 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> be sure you load uac module from modules_k, not from modules_s. >> >> Cheers, >> Daniel >> >> >> On 10/31/11 12:36 PM, Vladimir Broz wrote: >>> ignore the "sip:b@@test.com" -> same result with the correct (only one >>> "@") value >>> >>> Thanks, >>> -Vlada >>> >>> On 10/31/2011 12:21 PM, Vladimir Broz wrote: >>>> Hi, >>>> >>>> I compiled 3.2 (master) with FLAVOUR=ser and I'm trying to use "uac" >>>> module. >>>> >>>> my config looks like: >>>> ... >>>> $uac_req(method)="NOTIFY"; >>>> $uac_req(ruri)="sip:server.test.com"; >>>> $uac_req(furi)="sip:a@test.com"; >>>> $uac_req(turi)="sip:b@@test.com"; >>>> ... >>>> >>>> and I'm getting this config file error >>>> >>>> 0(26125) ERROR: <core> [pvapi.c:706]: error searching pvar "uac_req" >>>> 0(26125) ERROR: <core> [pvapi.c:910]: wrong char [m/109] in >>>> [$uac_req(method)] at [9 (5)] >>>> 0(26125) : <core> [cfg.y:3501]: parse error in config file >>>> /root/vbroz/ser-srrs-bpi.cfg, line 388, column 2-17: unknown script >>>> pseudo variable $uac_req(method) >>>> ERROR: bad config file (1 errors) >>>> >>>> What's wrong? >>>> >>>> Any help is appreciated... >>>> >>>> Thanks in advance, >>>> -Vlada >>>> >>>> _______________________________________________ >>>> 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 >> -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

uac module - error searching pvar "uac_req"
by Vladimir Broz 31 Oct '11

31 Oct '11

Hi, I compiled 3.2 (master) with FLAVOUR=ser and I'm trying to use "uac" module. my config looks like: ... $uac_req(method)="NOTIFY"; $uac_req(ruri)="sip:server.test.com"; $uac_req(furi)="sip:a@test.com"; $uac_req(turi)="sip:b@@test.com"; ... and I'm getting this config file error 0(26125) ERROR: <core> [pvapi.c:706]: error searching pvar "uac_req" 0(26125) ERROR: <core> [pvapi.c:910]: wrong char [m/109] in [$uac_req(method)] at [9 (5)] 0(26125) : <core> [cfg.y:3501]: parse error in config file /root/vbroz/ser-srrs-bpi.cfg, line 388, column 2-17: unknown script pseudo variable $uac_req(method) ERROR: bad config file (1 errors) What's wrong? Any help is appreciated... Thanks in advance, -Vlada

2 2

Re: [SR-Users] how to retrieve the contact username from 3xx reply?
by Uri Shacked 30 Oct '11

30 Oct '11

Thanks! works great.

1 0

how to retrieve the contact username from 3xx reply?
by Uri Shacked 30 Oct '11

30 Oct '11

---------- Forwarded message ---------- From: Uri Shacked <uri.shacked(a)gmail.com> Date: Sun, Oct 30, 2011 at 12:16 PM Subject: how to retrieve the contact username from 3xx reply? To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> I need to store the user i get from the 301 reply to an AVP and use it afterwards in another route. How can i retrieve the contact from the 3xx reply? uac_redirect does not have this option and t_load_contacts does not work for me as well.... Anyone has an idea?

2 1

module docs broken
by Timo Reimann 28 Oct '11

28 Oct '11

Hey, documentation on the dialog module seems to be broken: The master branch directory content is empty http://sip-router.org/docbook/sip-router/branch/master/modules_k/dialog/ 3.1 shows close to nothing http://sip-router.org/docbook/sip-router/branch/3.1/modules_k/dialog/dialog… Moreover, 3.0 gives me 404 on the entire modules_k path: http://sip-router.org/docbook/sip-router/branch/3.0/modules_k/ Help. :) Cheers, --Timo

2 2

T1 timer
by Vitaliy 28 Oct '11

28 Oct '11

Hello, How can i change T1 timeout for transaction created by the UAC module (uac_req_send). Do i have to call t_set_retr before uac_req_send and then t_reset_retr() before t_relay ? I use uac_req_send to send accounting to the remote server and i want to increase retransmit timeout for such signalling messages.

4 5

Fwd: [RAI] SIPit 29 Summary
by Klaus Darilion 28 Oct '11

28 Oct '11

FYI -------- Original Message -------- Subject: [RAI] SIPit 29 Summary Date: Fri, 28 Oct 2011 09:40:51 +0200 From: Robert Sparks <rjsparks(a)nostrum.com> To: rai(a)ietf.org (also available at https://www.sipit.net/SIPit29_summary) SIPit 29 was hosted by Etsi Plugtests in Monte Carlo, Monaco the week of October 24-27, 2011. There were 34 attendees from 17 companies visiting from 12 countries. We had 25 distinct implementations. The roles represented (some implementations act in more than one role) 20 endpoints 5 proxy/registrars The b2buas attending this event were not attempting to appear to be proxies. Implementations using each transport for SIP messages: UDP 92% TCP 96% TLS 88% (24% server-auth-only) SCTP 8% DTLS 0% 44% of the implementations present supported IPv6. There were three RFC4474 Identity implementations present. For DNS we had support for: Full RFC3263 : 76% SRV only : 16% A records only : 08% no DNS support : 0% Support for various items in the endpoints: 75% replaces 50% ice 40% 5389stun 35% turn 25% sip/stun multiplexing 30% gruu 25% history-info (there were no implementations of 4244bis) 20% outbound 20% path 20% diversion 15% 3489stun 10% join 5% service-route 0% sigcomp Support for various items in the proxy/registrars: 100% path 80% sip/stun multiplexing 60% gruu 40% outbound 40% diversion 20% service-route 0% history-info The endpoints and b2buas implemented these methods: 100% INVITE, CANCEL, ACK, BYE 95% REFER 95% NOTIFY 80% REGISTER 80% INFO 75% OPTIONS 75% UPDATE 75% PRACK 60% SUBSCRIBE 35% MESSAGE 30% PUBLISH 100% of the implementations sent RTP from the port advertised for reception (symmetric-rtp). 30% of the implementations required the other party to use symmetric-rtp. 90% of the UAs present both sent RTCP and paid attention to RTCP they received. 80% of the endpoints present supported SRTP using sdes. There were no dtls-srtp implementations at this SIPit. 35% of the endpoints supported multipart/MIME. There were no implementations present with S/MIME support. 36% of the implementations present followed RFC6026 (corrections to the INVITE transaction) 36% followed RFC4320 (corrections to the non-INVITE transaction) Not counting implementations that support events only for REFER: There were 3 SIP Event Server implementations There were 5 SIP Event Client implementations These event packages were supported: Server Client 1 2 presence 1 1 presence.winfo 1 4 message-summary 1 1 dialog 1 1 reg 1 1 conference 1 1 xcap-diff 0 2 kpml These packages were supported with the eventlist extension: Server Client 0 1 presence Two of the proxies present still rely only on max-forwards and one that implemented 3261 loop detection. There were two implementations of fork-loop-fix, but no implementations of max-breadth. Multiparty tests (Notes provided by several test participants, including Eoin McLeod, Trond Andersen, Olle Johansson, and Adrian Georgescu) ** IPv6 Focused Tests The first multiparty session of the event focused specifically on IPv6. We used a basic forking scenario with endpoints registered with several proxies at the bottom of the tree. We had a mix of single and dual-stack endpoints, many supporting video. None of the participant's code would put both v4 and v6 candidates into ICE. Rather, it made assumptions about which candidates to use based on what was happening with the SIP signaling. We ran tests using SRV configurations stating that v6 should be tried first and v4 tried if the v6 failed (and a second configuration that tested v4 first) to ensure that implementations would fail forward and try both address families, with mixed results. We confirmed that one UA configured to use IPv6 only was able to request and use an IPv4 turn allocation and succeeded in making a call between an IPv4 UA and IPv6 ua. Most UAs that supported dual-stack had a configuration to tell the application to ignore any returned AAAAs due to issues encountered in deployments where endpoints autoconfigured IPV6 that didn't actually work. We successfully tested calls going though a mix of v4 and v6 hops (accruing Via and Route/Record-Route headers with addresses in both families. There was an instance of UA registering using sip-outbound (RFC5626) that used IPv4 and IPv6 flows simultaneously. We set up automated tests of using SRV records to indicate preference for IPv4 or IPv6. Two ipv4 only clients proved that they connect to a domain indicating preference for IPv6. ** Forking There are still UAs that do not sensibly handle multiple 200 OKs to a forked INVITE. When testing handling multiple 200 OKs on a simple two branch fork, we encountered an interesting failure. As the proxy handled the 200 OK from branch 1, it sent a CANCEL to the other branch. The UA on branch 2 had already responded with a 200 OK, so it responded to the CANCEL with a 200, but rather than ignoring it, it decided to send a BYE on the dialog. This surprised the calling UA which had chosen the 200 OK from branch 2 to keep, having already sent a BYE to the dialog from branch 1. The better thing for the UA to have done is ignore the CANCEL after replying to it. ** TLS Most of the implementations at the event supported TLS, and all the multiparty tests exercised it. One session focused on proper certificate verification. We tested a certificate at a server with several SAN URI fields, none of which matched the URI used for the server, but a CN that did. Only one UA failed to connect because of this, the rest of the UAs connected happily (or not at all, but because of other issues). Most of the implementations were "before 5922" code, which meant that there was a lot of checking of the Subject/CN. ** Identity There were several SIP Identity implementations present. We tested creating and verifying the Identity assertions at proxies. It wasn't immediately obvious to some implementers that the resource pointed to by the Identity-Info URI had to be DER encoded (a result of the requirement to use application/pkix-cert). Unfortunately, the tar encoded at the end of 4474 has .cer files in it that are PEM encoded. ** STUN/TURN/ICE and Outbound We spent a significant fraction of the multiparty test time (three sessions) focusing on STUN/TURN/ICE and Outbound. There are still clients using RFC 3489 STUN. The participants noted operator pushback against deploying ICE because when things go wrong, it's very hard to diagnose when, where and why they went wrong. If the endpoints always sent a reINVITE (or UPDATE) when concluding ICE, even if they used the default candidates, this diagnosis of failures would be much easier. Examples noted by the participants: * Without a mandatory conclusion, is not possible to know how long it took until end-points had successful media path established * Without a mandatory conclusion, is not always possible to know if different than default candidates have been chosen as lack of INVITE does not tell deterministically what happened * Without a mandatory conclusion, it is not possible for the callee to restart ICE if it does not agree with the conclusion We successfully exercised endpoints setting up and managing multiple outbound flows through separate edge proxies to a proxy/registrar (with different implementations taking turns in the proxy registrar role). We forced the network to break the flow in use in each case, verifying that subsequent dialog-forming requests would take an alternate flow. We discovered an issue with the recommendation in 5626 to have edge proxies record-route that prevents subsequent in-dialog requests from using alternate flows. ** SRTP All implementations supported SDES for SRTP key exchange. Good interop with many different combinations of endpoints. Some calls were left up for an extended period so that sequence number wrapping and roc increments were tested. Once the roc had changed the call was put on hold and resumed with success. In this situation new RTP streams with different SSRCs were created and the sequence number/roc combination re-generated thus proving that support for multiple SRTP contexts exists. Debate surrounding how to signal 'best-effort crypto' continues. Some implementations choose to advertise RTP/AVP with a=crypto attributes and then go crypto if answer also has them and non-crypto if they do not. RTP/SAVP is used to signal the desire for mandatory crypto. Another approach observed was to repeat the same media line twice, once for crypto and once for non-crypto and rely on the receiving end to only accept one of them and port reject the other. In this case both m-lines had the same port number and payload number descriptions so if a receiving end accepted both then the originator would have no clue as to whether it was receiving crypto or non-crypto traffic. ** Unusual messages We tried several kinds of valid, but unusual messages. Very few implementations did the right thing with a request-URI with an unknown scheme - some phones would answer an INVITE with that in the request-URI. ** Specification questions / comments RFC 3263 currently says "If no SRV records are found, the client SHOULD use TCP for a SIPS URI" - it should probably say TLS over TCP. In the spirit of Happy Eyeballs, there should be some guidance on what to do when the RFC 3263 process results in both A and AAAA records and the client is capable of using both. There were implementations of SIP over DTLS, but the specification of that usage seems not to have completed. There were questions on whether internationalized domain names could be used in SIP URIs. For use of SDP with ICE, there was an argument about whether a=rtcp was mandatory if rtcp was on the next port up. RFC 5245 says "If the agent is utilizing RTCP, it MUST encode the RTCP candidate using the a=rtcp attribute as defined in RFC 3605 [RFC3605].", and 3605 says "when that port is not the next higher (odd) port number following the RTP port described in the media line" to motivate why the attribute was created to begin with. Any updates to 5245 should clarify that adding the a=rtcp attribute is not optional when RTCP is being sent on the next higher port. _______________________________________________ RAI mailing list RAI(a)ietf.org https://www.ietf.org/mailman/listinfo/rai

1 0

How to limit maximum number of calls
by Austin Einter 28 Oct '11

28 Oct '11

Hi I am running kamailio proxy (1.5) as an intermediate proxy, where all my SIP signalling packets are passing through. I want to limit maximum number of calls kamailio proxy server can handle at a time. How can I do this, please give me some pointer in this regard. Regards Austin

4 7

new email contact
by Timo Reimann 28 Oct '11

28 Oct '11

Hey all, this is just to let everybody know that my email contact has changed to sr(a)foo-lounge.de . Of course, I'm still on the lists but if anyone aware of my former email address wishes to contact me directly, please refer to the new address in the future. Thanks a lot and cheers, --Timo

1 0

Asterisk 1.8 and Kamailio 1.5 issue
by Rowell Rufino 27 Oct '11

27 Oct '11

Hi, We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk. Below is the sip trace... I am also attaching a tcpdump. Please help what we can do. Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060 ;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 ------------------------------ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ------------------------------ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060 ;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ------------------------------ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ------------------------------ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 *X-Asterisk-HangupCause: Protocol error, unspecified *X-Asterisk-HangupCauseCode: 111 Content-Length: 0 Regards, Rowell

2 1

Asterisk 1.8 and Kamailio 1.5 issue
by Rowell Rufino 26 Oct '11

26 Oct '11

Hi, We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk. Below is the sip trace... I am also attaching a tcpdump. Please help what we can do. Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 ________________________________ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ________________________________ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ________________________________ Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 ________________________________ Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 Regards, Rowell

1 0

Who is at Astricon?
by JR Richardson 26 Oct '11

26 Oct '11

I am. On 10/25/11, sr-users-request(a)lists.sip-router.org <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. Re: Dispatcher Confusion (v3.2.0) (Daniel-Constantin Mierla) > 2. Re: Dispatcher Confusion (v3.2.0) (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Oct 2011 05:44:06 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Dispatcher Confusion (v3.2.0) > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <4EA78206.4060203(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > On 10/25/11 5:52 PM, Asgaroth wrote: >> Hi Daniel, >> >> Thanks for the clarrification, >> >> On 25/10/2011 16:30, Daniel-Constantin Mierla wrote: >>> >>> >>> 4.6. |ds_mark_dst("s")| >>> >>> Mark the last used address from destination set as inactive >>> ("i"/"I"/"0"), active ("a"/"A"/"1") or probing ("p"/"P"/"2"). With >>> this function, an automatic detection of failed gateways can be >>> implemented. When an address is marked as inactive or probing, it >>> will be ignored by 'ds_select_dst' and 'ds_select_domain'. >>> >> >> Above is the part that is a little misleading, it says that >> >> "When an address is marked as inactive or probing, it will be ignored >> by 'ds_select_dst' and 'ds_select_domain'." >> >> This, to me, means that if a gateway is in probing mode >> (Active-Probing) then it wont be selected in the destination set >> because it is in probing mode, if this is not the case then maybe the >> above line should read: >> >> "When an address is marked as inactive or inactive-probing, it will be >> ignored by 'ds_select_dst' and 'ds_select_domain'." >> >> This brings me to my next question then, how would I set the state of >> a destination to Inactive-Probing (state: IP) from the routing script. >> The ds_mark_dst() function only allows one of "a", "i", "p". >> >> If I do a ds_mark_dst("i"), then the state changes to "IX", Inactive >> with no probing to tell when gateway is back up. >> >> If I do a ds_mark_dst("i") and then right after ds_mark_dst("p"), a >> log is printed saying that you cannot put a destination into probing >> state when it is marked as inactive. > are you sure you run the devel version? There is no such case in the > sources -- send me exactly the log message you get. Only when the > destination is disabled the probing cannot be set, but not the same case > of inactive. > >> >> Is it possible to set the state of a gateway to inactive-probing from >> the routing script? > > Yes, it should be, no constraint in master branch source code. > > Daniel > >> >> 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 > Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > >

2 1

sip-router listen address
by caio 25 Oct '11

25 Oct '11

Hello, Is there a problem or anything to consider when sharing the same ip/interface between sip proxy and asterisk? I have asterisk on port 5080 and SR at 5060 in the same interface with the same IP address and have a missing BYE when call is over. However, moving asterisk service to different interface and ip in the same box, the issue is solved. I appreciate your comments. Regards, Claudio

3 4

Kamailio.cfg file woes
by Peter Schrock 24 Oct '11

24 Oct '11

I am wondering two things. 1. is there a place on the web that I may find a basic cfg file to at least get Kamailio running. 2. is there a resource that tells you what to include for what you want to use the system for. At this point, it seems like I should just include everything because I don't know what I might not need. I looked at the references on the website and found the information a little overwhelming with all the options that are available. I think smaller bit sizes would be helpful at this time for me. Peter

3 3

onsend_route and replies
by Asgaroth 24 Oct '11

24 Oct '11

Hi All, Does the onsend route get executed for all messages? I'm just performing some xlog's in the onsend_route, but I'm not seeing it being called for replies. Currently my onsend_route looks like: onsend_route { xlog("onsend_route : $rm : Recieve Details $si:$sp -> $Ri:$Rp"); xlog("onsend_route : $rm : Destination Details $snd(af) $snd(proto):$snd(ip):$snd(port)"); } I can see it being executed for initial messages coming in but I dont see it executed for any replies. Thanks

1 1

kamctl dialplan addrule
by Arthur Alkhasyan 24 Oct '11

24 Oct '11

Hi all, I'm new to the list and new to the Kamailio. We are using Kamailio with Mysql to route calls between clients asterisks and SIP PSTN gateways. We have two inbound gateways and one outbound. I need to add outbound gateway and a few new features: 1. Call routing between carriers based at prefixes added by asterisks. I would like to route all calls with prefix 99101 to gateway 1 and prefix 99102 to gateway 2. Prefix have to be striped out before sending call to carrier. 2. Remote Call Forwarding. I have to setup RCF at Kamailio. All calls to the particular DID number have to be forward to another number. I found description for dialplan module and believe the module can work for me. I recompiled Kamailio with dialplan module support and added to config # ----- dialplan params ----- modparam("dialplan", "db_url", "mysql://openser:openserrw@localhost/openser") I did search and found this thread related to my problem: http://lists.sip-router.org/pipermail/sr-users/2009-November/025416.html I try to use kamctl to add rule to the MySQL and can't find proper syntax kamctl dialplan addrule 10 100 1 ^123+ 0 ^(123.+) 0030\1 -bash: syntax error near unexpected token `(' any help is greatly appreciated. Thank you.

3 3

Re: [SR-Users] module version mismatch for carrierroute
by Uri Shacked 24 Oct '11

24 Oct '11

Hi, i found out the mistake i made... didn't add the carrierroute to the include modules when making cfg file. thanks

1 0

module version mismatch for carrierroute
by Uri Shacked 24 Oct '11

24 Oct '11

Hi, i installed 3.2 from git. when starting kamailio i get the following: "ERROR: module version mismatch for /usr/local/lib/kamailio/modules/carrierroute.so; core: kamailio 3.2.0 (i386/linux); module: kamailio 3.1.4 (i386/linux)" any ideas what to do? thanks, Uri

3 2

Daily tarballs not being created anymore?
by Timo Reimann 24 Oct '11

24 Oct '11

Hi all, something caught my attention today: Apparently, tarballs provided from http://sip-router.org/tarballs/sr/ do not seem to be updated automatically anymore (the latest files are from March) although the Download page states that this should happen: "Tarballs for master (development) and several branches are made available daily (only if there were changes in the sources from previous tarball)" Is there a reason why tarball creation has ceased? If so, I think the Download page information should be corrected. I'd be prefer to have automatic tarballs again, however, as it's easier to point people not involved into development at tarballs instead of a set of git instructions. Thanks and cheers, --Timo

3 2

Kamailio 3.2 compile error on Solaris 10
by Nathaniel L Keeling 22 Oct '11

22 Oct '11

I am trying to compile the new version of Kamailio, 3.2 on Solaris in a 64bit environment and I am getting this error. I have been searching the internet but have not been able to find any solutions. Any help would be appreciated. CC (gcc) [M xmlrpc.so] xmlrpc.o In file included from ../../parser/msg_parser.h:63:0, from ../../sr_module.h:62, from xmlrpc.c:41: ../../parser/../ip_addr.h:86:23: error: field 'sin6' has incomplete type ../../parser/../ip_addr.h: In function 'ip_addr_loopback': ../../parser/../ip_addr.h:283:3: warning: implicit declaration of function 'htonl' [-Wimplicit-function-declaration] ../../parser/../ip_addr.h:286:3: warning: implicit declaration of function 'IN6_IS_ADDR_LOOPBACK' [-Wimplicit-function-declaration] ../../parser/../ip_addr.h: In function 'sockaddr2ip_addr': ../../parser/../ip_addr.h:353:50: error: dereferencing pointer to incomplete type ../../parser/../ip_addr.h: In function 'su_getport': ../../parser/../ip_addr.h:399:4: warning: implicit declaration of function 'ntohs' [-Wimplicit-function-declaration] ../../parser/../ip_addr.h: In function 'su_setport': ../../parser/../ip_addr.h:418:4: warning: implicit declaration of function 'htons' [-Wimplicit-function-declaration] xmlrpc.c: In function 'rpc_scan': xmlrpc.c:1454:6: warning: variable 'fmt_len' set but not used [-Wunused-but-set-variable] gmake[1]: *** [xmlrpc.o] Error 1 gmake: *** [modules] Error 1 Thanks Nathaniel

4 3

Force signalling??
by Skyler 21 Oct '11

21 Oct '11

Hi, I'm trying to setup an inbound proxy (Proxy-1) using dispatcher to relay between 2 other proxies (proxy-2 & 3) depending on DID, but I need to force all signalling through Proxy-1. so #2 & 3 only communicate with #1 and #1 only communicates up and down stream. My trouble is, the INVITES contain the original $ru $fu $tu & $ct info when passing down to Proxy-2 or 3. The result is, only partial signalling is being sent to Proxy-1. After the ACK it seems that Proxy-1 is removed from the communication path entirely as $ru becomes $rU@proxy-2 in the ACK from upstream so Proxy-1 is out of the picture from then on. This means BYE's are not working as Proxy-2 & 3 do not have authentication upstream. I suspect that I need to re-write $ru $fu $tu & $ct info on the original INVITE to reflect Proxy-1 information before dispatching. Is that what I'm missing? If so, how would I restore the original INVITE info to send upstream again on replies etc.? Any help is greatly appreciated. Tia, Skyler

3 6

New developer: Mészáros Mihály
by Daniel-Constantin Mierla 20 Oct '11

20 Oct '11

Hello, just to let everyone know that a new person got today write access to GIT repository: Mészáros Mihály (cc-ed), having the user id misi. Welcome to development team and looking forward to your contributions! Involved in the mailing lists in the past, he submitted the first new module to be part of next major release: presence_profile, implementation of an extension about sip user agent configuration framework, specified in RFC6080: * http://tools.ietf.org/html/rfc6080 The code of the new module is now in GIT branch misi/ua-profile, to be merge soon to master after initial review are done. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Build version in custom server_header
by Asgaroth 20 Oct '11

20 Oct '11

Hi All, Is it possible to have the build/git version added in a custom server_header/useragent_header? For example, my kamailio version is: version: kamailio 3.2.0 (i386/linux) 94d3b8 if I had the following config options: server_header="Server: Proxy" useragent_header="User-Agent: Proxy" Is it possible to do something like: server_header="Server: Proxy ($buildver)" useragent_header="User-Agent: Proxy ($buildver)" Where "$buildver" is the build/git release (94d3b8)? I'm looking at the core cookbook but I dont see anything that jumps out at me, is there a variable that can do this? Thanks

2 2

ACK routing loop
by Antanas Masevicius 20 Oct '11

20 Oct '11

Hello, after troubleshooting ACK routing loops i came up with a thread: http://www.openser.org/pipermail/users/2009-March/004123.html In short, loops formed when i entered IP address to 'domain' table of a host which was remote. Call flow: UAC INVITE -> Kamailio (with remote domain entered) -> UAS (host whose IP matches that remote domain) It looks like after loose_route() check ACK to 200 OK gets routed to local host. Besides this issue, IP address of remote host in 'domain' table did no harm at all. Question is, is that normal ACK routing behaviour or loose_route() should still check if given IP in request is really bound to local interface? Domain module documentation states that "A “local” domain is one that the proxy is responsible for", but id does not tell you that you cannot set remote IP address as a local domain. Please advice. Best regards, Antanas Masevicius

2 4

Dispatcher Confusion Redeux
by Asgaroth 20 Oct '11

20 Oct '11

Hi All, I'd like to clear something up with the diapatcher module and it's probing parameters: First things first: ./kamailio -V version: kamailio 3.1.5 (i386/linux) 76fff5 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, PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 76fff5 compiled on 16:21:02 Oct 19 2011 with gcc 4.1.2 I think I am suffering from the same confusion as Klaus in the following archive message: http://web.archiveorange.com/archive/v/jZFTG61DlcEr9LJbS8Gq I have the following parameters set for the dispatcher module: modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshhold", 1) modparam("dispatcher", "ds_probing_mode", 0) When I first start kamailio all gateways are active, no probing occurs. If I then set a gateway inactive (ds_set_state i 1 sip:...), no probing occurs. If I then set same gateway active (ds_set_state a 1 sip:...), no probing occurs. If I mark same gateway as probing (ds_mark_dst("p") after 1 attempt as specified in module parameter), then probing starts. If I then set same gateway inactive (ds_set_state i 1 sip:...) probing still occurs. According to the docs if I set the state of a gateway to inactive then no probing should occur. From my understanding of the docs, if I set ds_probing_mode to 0 then only gateways marked as probing will be "pinged". Can someone with more experienced than myself please clarify how this method is supposed to work. Thanks

3 3

Re: [SR-Users] Kamailio v3.2.0 Released - new major version is
by Javier Gallart 20 Oct '11

20 Oct '11

Hello congratulations to all the people who has contributed! I'm looking forward to try it out, some of the new features look really promising... Regards Javi > ------------------------------ > > Message: 2 > Date: Tue, 18 Oct 2011 20:30:22 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: [SR-Users] Kamailio v3.2.0 Released - new major version is > out > To: kamailio users <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org > Message-ID: <4E9DC5BE.8080003(a)gmail.com> > Content-Type: text/plain; charset=windows-1252; format=flowed > > Kamailio (OpenSER) SIP Server v3.2.0 is out ? a major release with a > very large number of new features and improvements. > > v3.2.0 release is a result of more than 10 months of development and 2 > months of testing from the teams of Kamailio (OpenSER) and SIP Express > Router (SER) projects. In the year of the 10th celebration of our > project (http://sip-router.org/10-years-ser/), this version comes with > 12 brand new modules in addition to a lot of fresh features in core and > old modules. > > Full release notes, including links for download and documentation, are > available at: > > * http://www.kamailio.org/w/kamailio-openser-v3-2-0-release-notes/ > > Many thanks to all developers and community members that made possible > this release. > > Enjoy SIP routing in a more flexible and easier way with Kamailio v3.2.0! > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > > > > >

2 1

Kamailio v3.2.0 RPM packages
by Daniel-Constantin Mierla 20 Oct '11

20 Oct '11

Hello, forgot to announce explicitly that the RPM packages for CentOS, RedHat, Fedora and OpenSUSE were built quickly after the release. They can be downloaded from OpenSUSE build system portal: Links to download packages for each OS are available in the wiki: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

kamctl version
by caio 20 Oct '11

20 Oct '11

Hello, Is kamctl utility version independent of the kamailio binary version? # kamailio -V version: kamailio 3.2.0 (x86_64/linux) 94d3b8 # kamctl version /usr/sbin/kamctl 3.1.0 Just a typo i guess. Regards, Claudio

2 1

kamailio and high-available db backend
by Juha Heinanen 19 Oct '11

19 Oct '11

mino, instead of ndb cluster, you can use drbd together with heartbeat and pacemaker to provide high available mysql/sip server solution. -- juha

1 0

very basic handling for ua-profile event
by MÉSZÁROS Mihály 19 Oct '11

19 Oct '11

Hi, Please find attached a new _very_ basic presence modul to make available user profile provisioning as it stated in RFC6080. Best Regards, Misi

2 3

kamailio and high-available db backend
by Mino Haluz 19 Oct '11

19 Oct '11

Hi, I was using ndb cluster as the dabatase backend for my kamailio setup. The thing is, mysql ndb cluster is difficult to maintain for me and it crashed multiple times (desynchronization, datamemory limit reached, and other strange reasons) and it is not that well documented (the user-base is quite narrow). My question is, what mysql architecture would you recommend me, if it has to be high available, well documented with various examples wihitn the community, thus easy to maintain. DB has 5% writes, 100queries/sec. and its size is 100MB. Mino

1 0

ok
by abdelazize rhosni 19 Oct '11

19 Oct '11

ok

2 1

Kamailio Dispatcher fails in front of FreeSwitch, help (again).
by Henrik Aagaard Sørensen 18 Oct '11

18 Oct '11

I've asked a lot of question about Kamailio Dispatcher in front of FreeSwitch in this mailing list. So I'm sorry that I'm still "spamming", but it still fails. This is my status so far. When connecting 2 devices directly to FreeSwitch everything works. When connecting to Kamailio, some calls fails with "Timeout" and some fails with "Too many loops". This is my kamailio.cfg: http://pastebin.com/KEuZmikh When connected directly to FreeSwitch where everything works, this is how my clients are registrered in FreeSwitch: http://pastebin.com/prkjkb7g When connected to Kamailio where nothing works, this is how my clients are registrered in FreeSwitch: http://pastebin.com/b5x6NZyv 215.197.145.132 (sip.my-domain.com) is my Kamailio. pbx.my-domain.com is my FreeSwitch. 96.224.14.164 is my IP at home, where my 2 devices are from. Can anyone help me out? I'm kinda lost at this point.

3 11

master branch is now open for commits of new features to be in v3.3.0
by Daniel-Constantin Mierla 18 Oct '11

18 Oct '11

Hello, this is a short notice to let everyone know that the GIT master branch is open for commits of new features -- to be part of next major release v3.3.0, as v3.2.0 has been officially released few moments ago. From this moment on, master and 3.2 branches will diverge, 3.2 taking in only bug fixes. So if you want to use stable release of existing features, switch to branch 3.2. You can find guidelines of installing branch 3.2 in the wiki site: * http://www.kamailio.org/wiki/install/3.2.x/git I spotted already a new module by one of the community members submitted to the mailing list in the last days, related to presence extensions, therefore looks like is going to be another great development cycle until 3.3.0 will be out. Happy coding SIP routing extensions in SER and Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Kamailio v3.2.0 Released - new major version is out
by Daniel-Constantin Mierla 18 Oct '11

18 Oct '11

Kamailio (OpenSER) SIP Server v3.2.0 is out – a major release with a very large number of new features and improvements. v3.2.0 release is a result of more than 10 months of development and 2 months of testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. In the year of the 10th celebration of our project (http://sip-router.org/10-years-ser/), this version comes with 12 brand new modules in addition to a lot of fresh features in core and old modules. Full release notes, including links for download and documentation, are available at: * http://www.kamailio.org/w/kamailio-openser-v3-2-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy SIP routing in a more flexible and easier way with Kamailio v3.2.0! -- Daniel-Constantin Mierla -- http://www.asipto.com

1 0

branch 3.2 was created
by Daniel-Constantin Mierla 18 Oct '11

18 Oct '11

Hello, short notice to let everyone know that the branch for 3.2.x release series was created with the name 3.2. For example, downloading sources from GIT and building Kamailio flavour form this branch can be done with: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio git checkout -b 3.2 origin/3.2 make FLAVOUR=kamailio cfg make all make install The official release of 3.2.0 will follow just a bit later today. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

YACC error during Compile of Kamailio 3.1.5
by Nathaniel L Keeling 18 Oct '11

18 Oct '11

I am building Kamailio 3.1.5 on Solaris 10 64bit environment and I am getting this error. yacc -d -b cfg cfg.y "cfg.y", line 104: fatal: invalid escape, or illegal reserved word: expect gmake: *** [cfg.tab.h] Error 1 I tried compiling with bison, but got errors immediately after getting thru the yacc (I just issued ln -s /opt/csw/bin/bison /opt/csw/bin/yacc and recompiled) flex cfg.lex CC (gcc) [kamailio] lex.yy.o In file included from ip_addr.h:50:0, from globals.h:37, from cfg.lex:91: ut.h: In function 'str2s': ut.h:195:17: warning: variable 'init' set but not used [-Wunused-but-set-variable] cfg.lex: In function 'yylex': cfg.lex:561:11: error: 'yylval' undeclared (first use in this function) cfg.lex:561:11: note: each undeclared identifier is reported only once for each function it appears in cfg.lex:561:40: error: 'FORWARD' undeclared (first use in this function) cfg.lex:562:40: error: 'FORWARD_TCP' undeclared (first use in this function) cfg.lex:563:40: error: 'FORWARD_TLS' undeclared (first use in this function) Thanks Nathaniel L Keeling

1 0

Make all error
by Peter Schrock 17 Oct '11

17 Oct '11

I am doing a "make all" and am getting this error: error: mysql/mysql.h: No such file or directory I have mysql installed and know that kamailio is having trouble finding mysql header files. Is there a configuration I can set up to help kamailio find it? Is there a resource I can be directed to find the info? Peter

3 5

preparing to release v3.2.0
by Daniel-Constantin Mierla 17 Oct '11

17 Oct '11

Hello, tomorrow is the day to release the new major version 3.2.0. Since involves creation of a new branch on GIT repository and some adjustments to the packaging, the plan is to start in the morning with the preparations. If any developer has something left to commit, please do it before 9:00am GMT. Afterwards it is very likely that the branch 3.2 will be created and there will be need for backport (which is fine as well). Then by 14:00 GMT the 3.2 branch should be frozen completely to start packaging the release. From than moment on until the release is announced, please send an email to sr-dev in case anyone wants to commit to branch 3.2. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Organization in SIP header
by Skyler 17 Oct '11

17 Oct '11

Hello all, Is there a way to set Organization: within SIP header? I know how to change user_agent_header and server_header but I'd rather not do that. Also I know I could do append_hf, but is there an organization_header? Thanks, Skyler

3 6

Problem with REGISTER retransmissions
by Alejandro Mingo 17 Oct '11

17 Oct '11

Hi, Since we installed last version of Kamailio (3.1.4) we have been experiencing a big problem with REGISTER retransmissions. When the server receives a retransmitted REGISTER it removes the binding and the UAC remains unregistered until next refreshing period. I include a PCAP capture that shows this behaviour. I am not 100% sure that this problem never happened with older versions. Has anybody experienced the same problem? Thanks in advanced. Alejandro.

3 9

Re: [SR-Users] OPENSER MIB
by Daniel-Constantin Mierla 17 Oct '11

17 Oct '11

Hello Stefan, On 3/14/11 11:03 AM, Stefan Tiedje wrote: > Thanks for the answer. > Maybe I have some older versions of the OPENSER-MIB and the other > related MIB's since I could not find the counter you pointed at. I'm > using a MIB browser for reading the MIB's. > Is the suggested counter "expired dialogs" added in a specific release > of Kamailio? Which? We use Kamailio 3.0.2. I used Kamailio and recommend using it sine it has the latest commits for stability. However, what I wrote before is pretty much not related to the version. There is a counter that tracks the processed dialogs, but seems it is not exported by default through snmpstats module. The statistics counter is named "processed_dialogs", implemented by dialog module. You can dump all internal statistics through kamctl or via xmlrpc command, but probably to export it through snmpstats you may need to extend the mibs and the code of the module. I just grepped the sources of snmpstats module to see what dialog statistics it is exporting: $ grep -n _dialogs modules_k/snmpstats/* | grep get_statistic modules_k/snmpstats/alarm_checks.c:83: num_dialogs = get_statistic("active_dialogs"); modules_k/snmpstats/snmpObjects.c:404: int result = get_statistic("active_dialogs"); modules_k/snmpstats/snmpObjects.c:424: get_statistic("active_dialogs") - modules_k/snmpstats/snmpObjects.c:425: get_statistic("early_dialogs"); modules_k/snmpstats/snmpObjects.c:443: int result = get_statistic("early_dialogs"); modules_k/snmpstats/snmpObjects.c:459: int result = get_statistic("failed_dialogs"); modules_k/snmpstats/snmpObjects.c:508: int num_dialogs = get_statistic("active_dialogs"); Perhaps when the snmpstats was developed the dialog module didn't export the statistics counter of "processed_dialogs" and then it was not updated. Now, what I tried to say is that if the "processed_dialogs" counter is not available through snmpstats (and it is not now after grepping the sources) you can get its value from another application through "kamctl get_statistics all" or XMLRPC command for all of the existing kamailio releases. Upcoming one we will look to implement the export through snmpstats as well. If you have time to do it and send us a patch, we will gladly commit it to source tree in our GIT repository. Cheers, Daniel > Do you have the MIB name for the "expired dialogs" counter. I will > look for that in my version of OPENSER MIBS. > Important, do you have a link to where MIB files can be downloaded for > Kamailio 3.0.2? > Below follows an excerp from one of the MIB's. Is it old, I don't know? > > -- *********************************************************************** > > -- OPENSER-MIB: OPENSER MIB > > -- > > -- Date of Creation: Januay 2006 > > -- > > -- This MIB provides information related to the OpenSER SIP Router. > > -- > > -- Copyright (c) The Internet Society (2006) > > -- Ammendments (c) Soma Networks, Inc. (2006) > > -- > > -- All rights reserved. > > -- ***************************************************************** > > /Stefan > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* den 14 mars 2011 10:16 > *To:* Stefan Tiedje > *Cc:* sr-users(a)lists.sip-router.org > *Subject:* Re: [SR-Users] OPENSER MIB > > Hello, > > On 3/14/11 9:42 AM, Stefan Tiedje wrote: >> Hi, >> In the Kamailio OPENSER-MIB there is the counter >> "openserTotalNumFailedDialogSetups". This is a Counter32. >> The description is: >> "The total number of calls that failed with an error. The >> following codes define a failed call:" >> *Question:* >> >> * I'm looking for the corresponding counter to >> "openserTotalNumFailedDialogSetups" who counts successful >> Dialog setups of Counter32 type. Does it exist? >> * If not, does it exist a work around? >> * Where in the code can the new suggested counter be added? >> * Something else???? >> > > the dialog module counts the number of processed dialogs, see: > http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2966360 > > There is no counter currently inside dialog module exporting exactly > the number of successfully setup dialogs, it should not be hard to do > it, though. Using the above and the number of failed and expired > dialogs, you can actually get the number of successful dialogs. > > Dialog module being the one that tracks SIP dialogs, therefore being > able to count them, now I don't know if snmpstats module exports all > the counters from dialog module. I setup snmpstats just few weeks ago > and works perfect on Ubuntu/Debian servers, but I had no need to check > dialog module counters. > > Note that you can get the list of all internal statistics via kamctl: > - kamctl fifo get_statistics all > > Or via XMLRPC if you need them remotely in another application. > > Another option is to define your statistics with statistics module. > Knowing that in SIP a successful call dialog means 200ok reply to an > INVITE transaction, you can count it in the onreply_route[abc] that > you arm for relayed transactions with t_on_reply("abc"). > > Hope these help you, > Daniel > >> Suggestion for the new counter is a name like: >> "openserTotalNumSucceededDialogSetups". It has a counter32. >> Description: "The total number of calls that succeeded" >> I know that there are the counters openserCurNumDialogs, >> openserCurNumDialogsInProgress and openserCurNumDialogsInSetup but >> these are of Gauge type who only reflects the current situation. >> These Gauge counters can't be used together with a Counter32 counter. >> That don't mix. The calculation done for the counter >> "openserCurNumDialogsInProgress" should be used where every new >> dialog setup is added to the new suggested counter. A counter of 32 >> should cover a great deal of connections. These counters are usually >> read, if used, every 15 minutes or 1 hour. >> *Rationale:* >> The reason for the new counter is that a calculation between >> succeeded and failed dialog setups can be done and be used for SLA >> agreements. Without this, its hard to make any customer versus >> provider agreements. >> /Stefan >> PS. Ask if anything is unclear and I need an answer rapidly. >> >> >> _______________________________________________ >> 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 -- Daniel-Constantin Mierla http://www.asipto.com

2 5

cleaning up contacts...
by Krishna Kurapati 16 Oct '11

16 Oct '11

Hi, I am seeing several Contacts for same AOR in userlocation when I do "kamctl ul show". I suspect, it's a result of UA crashing after registering and then registering with different contact after restart. Since the UA crashed, it could not send Register with expire=0 to remove the previous contact. Since the userlocation API is not exported to configuration scripting, I am wondering how to clean any spurious contacts. Thanks Krish Kura

2 1

FIFO error in cfg file
by Peter Schrock 16 Oct '11

16 Oct '11

I tried opening with "kamctl moni" and I get this error message: ERROR: Error opening Kamailio's FIFO FIFO ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "FIFO")' in your config ERROR: and also have loaded the mi_fifo module. I looked at the cfg file and it looks like what it is asking for is already there. Any suggestions would be welcomed. Peter

2 1

Force socket on reply
by Jijo 14 Oct '11

14 Oct '11

Hi All, Our system has two IP address, one is used for kamailio and 2nd one is used for data. The problem is suppose a request came in to kamailio on a TCP connection with first IP address and connection is torned down before sending the response. Later when the response is send out kamialio is using the 2nd IP to create the connection. The reason is kamailio doesn't have force socket in in reply, so it uses INADDR_ANY for soruce addres .so kernel uses source address based on the detaintion IP. I tired to force the socket using pseudo variable $fs or force_send_socket(), but neither of them worked from reply_route. So i modfied the code relay_replay to set the SND_F_FORCE_SOCKET which will use the address where request is recieved as source address to make the TCP connection The code change is shown below. Please let me know is there anyother way we could acheive it? }else{ buf = build_res_buf_from_sip_res( relayed_msg, &res_len ); /* if we build a message from shmem, we need to remove via delete lumps which are now stirred in the shmem-ed structure */ if (branch!=relay) { free_via_clen_lump(&relayed_msg->add_rm); } /* update send_flags with possible additions from the reply route */ SND_FLAGS_OR(&uas_rb->dst.send_flags, &uas_rb->dst.send_flags, &relayed_msg->rpl_send_flags); /* Make the response to use the same IP where it receives the message */ * * * uas_rb->dst.send_flags.f |= SND_F_FORCE_SOCKET; * } Thanks JIjo

2 4

Cannot concect to iptel.org
by Enda Mannion 14 Oct '11

14 Oct '11

Hi, I am trying to connect to the iptel.org SIP server using the recommended jitsi SIP UA. Is the service still up and running? I am using my newly created username and password but I get no response back from the REGISTER I send. Enda

1 0

a=nortpproxy:yes
by Austin Einter 13 Oct '11

13 Oct '11

If the SIP UA is not behind the NAT, and Kamailio insersts a=nortpproxy:yes in SDP body, can it cause one sided audio. I am using PJSIP as my UA, kamailio 3.1.5 , rtp proxy 1.2.1 as intermediate proxy for signalling and media. In kamailio config, I have enabled WITH_NAT. On call setup, only one way audio is there. When I looked at SIP messages, the only additional stuff I can see a=nortpproxy and nat=yes in recpord route. Without kamailio as intermediate proxy, audio is good both way. So I feel, when I bring up kamailio as intermediate proxy I am doing some mistake configuration wise either at PJSIP UA or in Kamailio. Has anybody faced this issue previously. If so please share how can we fix this issue. Thanks in advance Austin

2 1

Tls protocol testing with iptel account
by Lello 13 Oct '11

13 Oct '11

Hi, I'm working with a softphone project named mjSip, now I've implemented the TLS protocol in the softphone and I would try it with 2 iptel account setted on two pc, on the first I have: # MjSip UA configuration file # ___________________________________________ # host_port=5061 transport_protocols= tls outbound_proxy=sip.iptel.org from_url=<sip:alexlel@iptel.org> username=alexlel realm=iptel.org passwd=2387lelloh debug_level=8 log_path=log max_logsize=2000 do_register=yes keepalive_time=8000 _______________________________ the second is such as this. However I have no response in SSL/TLS by the iptel server. Does it work in TLS? Thanks, Regards.

2 1

Employment you've been searching!
by apache＠iptel.org 13 Oct '11

13 Oct '11

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: I would like to take this time to welcome you to our hiring process and give you a brief synopsis of the position's benefits and requirements. If you are taking a career break, are on a maternity leave, recently retired or simply looking for some part-time job, this position is for you. [...] Content analysis details: (12.2 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [188.49.28.2 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: it-jobsearch.com] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: it-jobsearch.com] 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?188.49.28.2>] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [188.49.28.2 listed in bb.barracudacentral.org] 1.5 BAYES_60 BODY: Bayes spam probability is 60 to 80% [score: 0.6541] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

Can't create a new account with kamctl
by Peter Schrock 12 Oct '11

12 Oct '11

I have finally got to the section of the install section "10. Ready to Rock". Unfortunately, I did get the error message "ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN". I did the two steps that the instructions gave and cannot find ".kamctlrc" to "add SIP_DOMAIN=mysipserver.com" Please help. I am not quite understanding what should be done to fix this Peter

2 1

Kamailio Developer Seminar, Oct 24-24, San Francisco
by Daniel-Constantin Mierla 12 Oct '11

12 Oct '11

Hello, the place and date for Kamailio/SER Developer Seminar have been set: San Mateo (just south of San Francisco, CA, in the nearby of international airport), for one day and a half, Oct 24-25, 2011. I posted more details, including about content and registration, at: * http://www.kamailio.org/w/2011/10/kamailio-developer-seminar-san-francisco-… Participating fee is 250USD. Selection of attendees is done by the rule of first come first served. The number of seats is limited, don't wait too long if you want to attend. Looking forward to meeting many of you there. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

Can anyone help for a small amount of money.
by Henrik Aagaard Sørensen 12 Oct '11

12 Oct '11

Hi everyone. I'm still facing problems with my Kamailio Dispatcher <-> FreeSWITCH setup. Everything works when connecting directly to FreeSWITCH. When connecting to Kamailio, which should work as a dispatcher (load-balancer) in front of FreeSWITCH I get "Too many hops", "Timeouts" etc. My test setup with SIP clients is an iPhone, an Android, an iPad and a laptops with Ubuntu. Can someone help me out, for a small amount of money, until I can get it to work? Hopefully it's just the kamailio.cfg which needs to do things correctly. Currently I'm using this http://pastebin.com/mahKECAw with help from Morten in this mailing list.

1 0

Creating a database error
by Peter Schrock 12 Oct '11

12 Oct '11

This is what I am getting: INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not found ERROR: Creating core database and grant privileges failed! I looked at the files and cannot find the problem. Please help. Peter

3 4

Kamailio Dispatcher and FreeSwitch, Too many hops.
by Henrik Aagaard Sørensen 12 Oct '11

12 Oct '11

I have a setup with Kamailio as dispatcher in front of a FreeSwitch server. This is my kamailio.cfg: http://pastebin.com/8PR2GFBD I'm currently getting "Too many hops" when calling between SIP clients. I am able to call to FreeSwitch and listen to voicemail, hold music etc. After a long conversation with a FreeSwitch expert, and some tests, I was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch. I've also run tshark on both FreeSwitch and Kamailio and when calling between clients they just send the packets between each other. Can anyone help me out? I've tried to Google a lot for this problem and asked in several IRC channels, mailing lists and forums. Without any luck.

6 21

Re: [SR-Users] Problem with Route containing two elements
by davy van de moere 12 Oct '11

12 Oct '11

Owki, reading RFC-3261 again ;) Imho Kamailio does what it is supposed to do, it forwards the packet to the first element in the Route header. But as I received the BYE packet from <sip:their_sbc> , that one should have stripped itself out of the Route header, making everything just work, right? Now I can hackishly fix it with a rewritehostport in the middle. But that will eventually not scale up that nicely. Ergo, does anyone have a good idea on how to e.g. have Kamailio in this situation ignore the first uri in the route header? Or am I still not getting it :) ? thx ! BYE sip:05xxxxxx at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060;transport=udp SIP/2.0 Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP their_sbc;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789 at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> ;lr=on;ftag=as5716fec8> Max-Forwards: 69 From: "Some_Telecom" <sip:02xxxxxxxx at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>;tag=as5716fec8 To: <sip:05xxxxxx at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6 at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060;transport=udp X-Enswitch-Source: their_internals:5060

2 2

proxy BYE to itself
by caio 12 Oct '11

12 Oct '11

Hello, I have the following issue. A calls B, talk for a while, then B sends a BYE, but A never receives it. The BYE is transmited from the proxy listening interface:port to itself, but not to the asterisk port. U 2011/10/08 00:29:29.207335 Kamailio_IP:5060 -> Kamailio_IP:5060 BYE sip:Kamailio_IP;lr=on SIP/2.0. The scenario is: A (pstn subs) --> Asterisk:5080 --> Kamailio:5060 --> B (registered sip ua) Asterisk and Kamailio are in the same box, listening both at the same IP (eth iface) address, but one at port 5080, and the proxy at 5060 (udp). The configuration is based on kamailio.cfg file provided by the v3.1.5 (with a few modules added, like dialplan and cr..etc). If A is who send the BYE, everything goes well. I attach the full sip capture. Thanks for you time. Claudio

1 1

Problem with Route containing two elements
by davy van de moere 11 Oct '11

11 Oct '11

I'm trying to integrate to an integrics enswitch, almost everything works like a charm, except on BYE packets as Kamailio in my setup forwards these incorrectly. Digging somewhat I believe the culprit sits in the Route header which comes from the enswitch: Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Kamailio takes into account the first argument (the thing sip:their_sbc...) on which it cannot find the correct transaction. And forwards it back to the sender, instead of forwarding it correctly to my freeswitch. which gives a storm of traffic ;) If I compare that with an ACK packet in the same transaction , having this as a Route header : Route: <sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Kamailio sends it along correctly. IMHO if I could convince Kamailio to always take the last part of the Route header into account and ignore the first one it would correctly work. What would be a good approach? Rewriting the route header looks abit harsch. disclaimer : I'm not sure which component is disrespecting the RFC, if any, or if I just fully not get something :) Short overview + packet detail (with some obfuscation): My setup: their_sbc --- my_kamailio --- freeswitch --- my_kamailio --- enduser BYE sip:05xxxxxx@my_kamailio:5060;transport=udp SIP/2.0 Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP their_sbc;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Max-Forwards: 69 From: "Some_Telecom" <sip:02xxxxxxxx@their_sbc>;tag=as5716fec8 To: <sip:05xxxxxx@their_sbc:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6@their_sbc CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx@my_kamailio:5060;transport=udp X-Enswitch-Source: their_internals:5060 BYE sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio;lr=on;ftag=as5716fec8 SIP/2.0 Record-Route: <sip:05xxxxxx@my_kamailio;lr=on;ftag=as5716fec8> Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP my_kamailio;branch=z9hG4bK1ae8.b4f95af2.0 Via: SIP/2.0/UDP my_kamailio;rport=5060;branch=z9hG4bK1ae8.a4f95af2.0 Via: SIP/2.0/UDP their_sbc;rport=5060;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Max-Forwards: 67 From: "Some_Telecom" <sip:02xxxxxxxx@their_sbc>;tag=as5716fec8 To: <sip:05xxxxxx@their_sbc:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6@their_sbc CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx@my_kamailio:5060;transport=udp X-Enswitch-Source: their_internals:5060

2 1

[OT] SIP on the Web
by Iñaki Baz Castillo 11 Oct '11

11 Oct '11

Hi folks, I would like to show a project my colleage José Luis Millán and me are working on. It's about SIP protocol running in a web browser. When RTCweb (media capable web browsers) becomes a reality both technologies together will allow real SIP endpoints coded in JavaScript running in a web browser and interoperating with real SIP networks. Bye bye Flash and custom protocol gateways. http://sip-on-the-web.aliax.net/ Cheers. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 10

limiting concurrent calls with dialog module
by Uri Shacked 11 Oct '11

11 Oct '11

Hi, I am trying to create concurrent calls limitations. two questions: 1. should i use dialog module? or is there a better module? 2. i was trying to work with the dialog module, but didnt understand how and where to configure the commands and which to use.... any examples or ideas? BR, Uri

10 17

releasing v3.2.0
by Daniel-Constantin Mierla 11 Oct '11

11 Oct '11

Hello, it is time to set the time to release v3.2.0. From my point of view, testing went very well, considering also that core and main modules were not touched much during this devel cycle, the most of the efforts were on new modules and application service components such as presence server, Lua ... A convenient date for me will be Tuesday, October 18, 2011, allowing to work a bit on packaging and docs. If there are other preferences, jump in this discussion. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 8

(no subject)
by alan chiu 10 Oct '11

10 Oct '11

http://toehugger.com/CRE/info.php?html143

1 0

Audio quality issue
by Austin Einter 10 Oct '11

10 Oct '11

Hi I am using Kamailio 3.1.5 . I am using RTP proxy also. I have used default kamailio.cfg.sample fiile , and just added line #!define WITH_NAT. I have another Main proxy. I wanted all my signalling and media packets should just pass through machine where Kamailio and RTP proxy are running. With this I found, call is established, all signalling and media packets are passing through kamailio / rtp-proxy. So far so good. One way audio stream (from called party to calling party) quality is good. The other audio stream (from calling party to called party is very bad. Did anybody face this issue? Please help me to sort out this issue audio quality issue. Regards Austin

4 5

Carrierroute: Could not allocate shared memory
by Mino Haluz 10 Oct '11

10 Oct '11

Hi, is there any setting which could allow me to set maximum memory per module ? As I am testing the carrier route module, I've added for testing purposes 100 000 rules sofar. When I start the kamailio, it gives me : Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute [../../ut.h:702]: could not allocate shared memory from available pool Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute [cr_rule.c:155]: could not allocate shared memory from available pool Thanks.

3 4

Memory Allocation in Kamailio
by Austin Einter 08 Oct '11

08 Oct '11

Hi Just a quick question. How memory is allocated in Kamailio code. Few places I saw pkg_malloc function. Is it the only function used to allocate memory or there are similar functions present in kamailio. Thanks Austin

3 2

tm and msilo module
by Mark Anthony C. Delfin 08 Oct '11

08 Oct '11

Dear List, I am trying to enable msilo on Kamailio but I'm getting an error ct 8 11:30:30 localhost kamailio[13856]: ERROR: tm [tm_load.c:48]: tm:load_tm: Module not initialized yet, make sure that all modules that need tm module are loaded after tm in the configuration file Oct 8 11:30:30 localhost kamailio[13856]: INFO: sl [sl.c:163]: could not bind tm module - only stateless mode available Oct 8 11:30:30 localhost kamailio[13856]: ERROR: tm [tm_load.c:48]: tm:load_tm: Module not initialized yet, make sure that all modules that need tm module are loaded after tm in the configuration file Oct 8 11:30:30 localhost kamailio[13856]: ERROR: msilo [msilo.c:333]: can't load TM API Oct 8 11:30:30 localhost kamailio[13856]: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module msilo (/usr/local/lib64/kamailio/modules_k/msilo.so) Any ideas how can I fixed it. Thank you in Advance. Regards, Mark Anthony C. Delfin

2 1

Kamailio RTP Proxy Questions
by Austin Einter 07 Oct '11

07 Oct '11

Hi I downloaded 3.1.5 Kamalio source, did make and install. Tried to run, and tested few basic scenarios, everything working fine. I have a main proxy, want to use kamailio as intermediate proxy and my requirement is all RTP packets should pass through machine in which Kamalio proxy is running. I beleive RTP Proxy I can run in same machine where Kamailio proxy is running with proper configuration . For this I am referring, http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-… page. In one of the steps it says "So, if we are using the Rtpproxy server with default configuration, we have to open /etc/default/rtpproxy file and uncomment following line regarding of udp socket, that will be sued for interconnection:". I tried to search rtpproxy configuration file, could not locate. While making and installing Kamalio proxy, the commands I used "make prefix=/usr/local/ all and make prefix=/usr/local/ install". Can somebody point me why rtpproxy config file is missing. Also the link I am refering to is correct or wrong one. Best Regards, Austin

2 6

Multiple INVITEs and discarded call_control
by Mino Haluz 07 Oct '11

07 Oct '11

Hi, my kamailio server is receiving from some customers 3 identical INVITEs when call is initiated (separated by 200ms). Those 3 INVITEs are making a big problem with call_control: WARNING: call_control [call_control.c:1156]: dialog to trace controlled call was not created. discarding callcontrol. That is why, the prepaid limit is not working at all in this case. This way the user can hack the prepaid protection of the account. Otherwise the call_control is fuilly functional. Anybody experienced the similar problem? If so, how to resolve it? Thanks, Mino Haluz

2 4

no more pkg mem on ul_dump
by Klaus Darilion 06 Oct '11

06 Oct '11

Hi! Kamailio 1.3.4: "kamctl ul monitor" causes: ERROR: <core> [attr.c:80]: no more pkg mem (21) ERROR: mi_fifo [fifo_fnc.c:513]: command (ul_dump) processing failed There are ~ 2000 entries in location table. Default PKG size is used. I remember there were some similar reports recently, but couldn't find the relevant threads. Any hints for the cause of this problem and workarounds? Thanks Klaus

4 8

Compiling on a 2.4 kernel
by Pedro Antonio Vico Solano 06 Oct '11

06 Oct '11

2 3

Problems with domain on registrations.
by Henrik Aagaard Sørensen 06 Oct '11

06 Oct '11

I have the current setup: SIP client <-> SER Proxy <-> SIP Trunk. My reason for the SER Proxy is that the SIP trunk only allows connections from approved IP addresses (such as my server with SER Proxy). My user on the SIP Trunk has the following authentication username@domain. I am able to register to the SIP Trunk as I'm setting up my SIP klient with username to username, domain to domain and then add my SER Proxy IP address as proxy. This works and means that I'm able to place calls to the SIP Trunk. BUT people cannot place calls from the SIP Trunk to my SIP client, due to the fact that on the SIP Trunk my locations is registered as username@SER-Procy-IP. So when people are calling me, the SER Proxy is looking for username@SER-Procy-IP and NOT username@domain, as I'm registered with. And then SER Proxy reports: 404 Not Found. Can anyone help me how to set this up correctly? I do not have access to the SIP Trunk, but I have all access to the SIP client and SIP Proxy.

2 2

Removing Contact from location table when TCP connection from UA is dropped...
by Krishna Kurapati 06 Oct '11

06 Oct '11

Hi, When UA connects with the Proxy over TCP, the contact is mostly bound to the firewall's binding IP and port for that connection. When the connection is dropped, the contact in Kamailio is no longer valid even though there is time left to expire the contact. Is there some way to link TCP connection and the contact in such a way that when TCP connection is dropped, contact from the location can be removed? Thanks Krish Kura

2 1

New in Kamailio
by Leonardo Sena 05 Oct '11

05 Oct '11

Hello I´m a new contribuitor of the list and my english dont very good then i hope of help more in the future I am installing Kamailio and Siremis but i dont understand like siremis works and where i can create the routes i would like step by step to routes rules. Best Regards, Leonardo Sena

1 0

Balancing incoming traffic (DID)
by Anto 05 Oct '11

05 Oct '11

Hello It presents a problem because I do not know how to assimilate the implementation. Until now we had a design of a proxy to balance out between several gateway, but it will add another proxy for another service (other users) and use the same gateway output or perhaps different (although not that be better if divide or unify the gateway). The problem as I have with the routing entry, which is not very well how to implement it. Now we have the design like this: -GW ----\ / - CARRIER1 USER ------ KAMAILIO ----- GW ---- + \ - CARRIER2 -GW ----/ And becomes like this: -GW ----\ / - CARRIER1 USER ------ KAMAILIOSERVICE1 ----- GW ---- + \ - CARRIER2 -GW ----/ -GW ----\ / - CARRIER1 USER ------ KAMAILIOSERVICE2 ----- GW ---- + \ - CARRIER2 -GW ----/ Or maybe using the same gateway. The problem is that each carrier provides DID number (I use the same for both services). Kamailio had thought to put a gateway between the carriers and that they did check the service and send it to the corresponding gateway (in the case of the gateway separately), or put it in the proxy gateway and making checking for routing to proxy (in the case of using the same gateway). What would be better? Any recommendations on how to put this into practice?. The biggest question that I believe is the routing table, as if this new proxy routes the traffic into and leaves her in the field via ip, also pass through the output data (I have doubts whether SIP level, following standard, you can not pass through the exit) and want it to be direct (USER -> Proxy -> GW -> CARRIER), just past the entrance. Anyone could advise me where I could find more information to set up this scenario?. thanks and sorry for bothering you with this question ;-). best regards

2 2

limiting concurrent calls with dialog module
by Uri Shacked 05 Oct '11

05 Oct '11

Hi, Thanks for the help. The idea of using SQLOPS for many things crossed my mind.... But....I am using ACC with DB, Dialog with DB and so... isnt it too much? i mean for load and capacity reasons? BR, Uri

2 1

Re: [SR-Users] Dialog ref Error
by Uri Shacked 05 Oct '11

05 Oct '11

On Wed, Oct 5, 2011 at 2:34 PM, Uri Shacked <ushacked(a)gmail.com> wrote: > OK... > now i see the problem is some where in the cfg file... > i get te following error : > > 0(3774) : <core> [cfg.y:3406]: parse error in config file > //etc/kamailio/kamailio.cfg, from line 944, column 2 to line 945, column 0: > syntax error > 0(3774) : <core> [cfg.y:3406]: parse error in config file > //etc/kamailio/kamailio.cfg, from line 944, column 2 to line 945, column 0: > bad command > ERROR: bad config file (2 errors) > 0(3774) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start > 0(3774) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash > table > 0(3774) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing > semaphores > 0(3774) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb > lists > 0(3774) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done > 0(3774) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy > 0(3774) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory > lock > > > here is my complete cfg file. > any ideas? > > > ... > > ######## Routing Logic ######## > > # Main SIP request routing logic > > # - processing of any incoming SIP request starts with this route > > route { > > setflag(FLT_TRC); # Trace message > > # per request initial checks > > route(REQINIT); > > # NAT detection > > route(NAT); > > # handle requests within SIP dialogs > > route(WITHINDLG); > > ### only initial requests (no To tag) > > # CANCEL processing > > if (is_method("CANCEL")) > > { > > if (t_check_trans()) > > t_relay(); > > exit; > > } > > t_check_trans(); > > # authentication > > route(AUTH); > > # record routing for dialog forming requests (in case they are routed) > > # - remove preloaded route headers > > remove_hf("Route"); > > if (is_method("INVITE|SUBSCRIBE")) > > record_route(); > > # account only INVITEs > > if (is_method("INVITE")) > > { > > #!ifdef WITH_DIALPLAN > > dp_translate("1", "$rU/$rU"); > > xlog("translated to $rU \n"); > > #!endif > > #!ifdef WITH_CARRIERROUTE > > ####cr_user_carrier("$rU", "$td", "$avp(carrierid)"); > > ####xlog("translated to var $avp(carrierid) \n"); > > cr_route("OSP1", "kamailio2", "$rU", "$rU", "call_id"); > > #!endif > > setflag(FLT_ACC); # do accounting > > } > > # dispatch requests to foreign domains > > route(SIPOUT); > > ### requests for my local domains > > # handle presence related requests > > route(PRESENCE); > > # handle registrations > > route(REGISTRAR); > > if ($rU==$null) > > { > > # request with no Username in RURI > > sl_send_reply("484","Address Incomplete"); > > exit; > > } > > # dispatch destinations to PSTN > > route(PSTN); > > # user location service > > route(LOCATION); > > route(RELAY); > > } > > route[RELAY] { > > #!ifdef WITH_NAT > > if (check_route_param("nat=yes")) { > > setbflag(FLB_NATB); > > } > > if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { > > route(RTPPROXY); > > } > > #!endif > > /* example how to enable some additional event routes */ > > if (is_method("INVITE")) { > > #t_on_branch("BRANCH_ONE"); > > t_on_reply("REPLY_ONE"); > > t_on_failure("FAIL_ONE"); > > } > > if (!t_relay()) { > > sl_reply_error(); > > } > > exit; > > } > > # Per SIP request initial checks > > route[REQINIT] { > > #!ifdef WITH_ANTIFLOOD > > # flood dection from same IP and traffic ban for a while > > # be sure you exclude checking trusted peers, such as pstn gateways > > # - local host excluded (e.g., loop to self) > > if(src_ip!=myself) > > { > > if($sht(ipban=>$si)!=$null) > > { > > # ip is already blocked > > xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); > > exit; > > } > > if (!pike_check_req()) > > { > > xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); > > $sht(ipban=>$si) = 1; > > exit; > > } > > } > > #!endif > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483","Too Many Hops"); > > exit; > > } > > if(!sanity_check("1511", "7")) > > { > > xlog("Malformed SIP message from $si:$sp\n"); > > exit; > > } > > #!ifdef WITH_PIPELIMIT > > # perform pipe match for INVITE method > > if (is_method("INVITE")) { > > if (!pl_check("avg")) { > > pl_drop(); > > exit; > > } > > if (!pl_check("peek")) { > > pl_drop(); > > exit; > > } > > #endif > > #!ifdef WITH_DIALOG > > xlog("currently active dialogs are $DLG_count\n"); > > dlg_manage(); > > dlg_setflag(FLT_DLG); > > if (is_method("INVITE")) { > > $var(size) = 0; > > get_profile_size("test","$rU","$var(size)"); > > xlog("currently, the user $rU has $var(size) active calls\n"); > > if( $var(size) >= 5 ) > > { > > sl_send_reply("507", "Simultaneous calls limit reached"); > > xlog("Limit of $var(size) active calls has been reached \n"); > > exit; > > } > > set_dlg_profile("test","$rU"); > > } > > #!endif > > } > > # Handle requests within SIP dialogs > > route[WITHINDLG] { > > if (has_totag()) { > > # sequential request withing a dialog should > > # take the path determined by record-routing > > if (loose_route()) { > > if (is_method("BYE")) { > > setflag(FLT_ACC); # do accounting ... > > setflag(FLT_ACCFAILED); # ... even if the transaction fails > > } > > route(RELAY); > > } else { > > if (is_method("SUBSCRIBE") && uri == myself) { > > # in-dialog subscribe requests > > route(PRESENCE); > > exit; > > } > > if ( is_method("ACK") ) { > > if ( t_check_trans() ) { > > # no loose-route, but stateful ACK; > > # must be an ACK after a 487 > > # or e.g. 404 from upstream server > > t_relay(); > > exit; > > } else { > > # ACK without matching transaction ... ignore and discard > > exit; > > } > > } > > sl_send_reply("404","Not here"); > > } > > exit; > > } > > } > > # Handle SIP registrations > > route[REGISTRAR] { > > if (is_method("REGISTER")) > > { > > if(isflagset(FLT_NATS)) > > { > > setbflag(FLB_NATB); > > # uncomment next line to do SIP NAT pinging > > ## setbflag(FLB_NATSIPPING); > > } > > if (!save("location")) > > sl_reply_error(); > > exit; > > } > > } > > # USER location service > > route[LOCATION] { > > #!ifdef WITH_ALIASDB > > # search in DB-based aliases > > alias_db_lookup("dbaliases"); > > #!endif > > if (!lookup("location")) { > > switch ($rc) { > > case -1: > > case -3: > > t_newtran(); > > t_reply("404", "Not Found"); > > exit; > > case -2: > > sl_send_reply("405", "Method Not Allowed"); > > exit; > > } > > } > > # when routing via usrloc, log the missed calls also > > if (is_method("INVITE")) > > { > > setflag(FLT_ACCMISSED); > > } > > } > > # Presence server route > > route[PRESENCE] { > > if(!is_method("PUBLISH|SUBSCRIBE")) > > return; > > #!ifdef WITH_PRESENCE > > if (!t_newtran()) > > { > > sl_reply_error(); > > exit; > > }; > > if(is_method("PUBLISH")) > > { > > handle_publish(); > > t_release(); > > } > > else > > if( is_method("SUBSCRIBE")) > > { > > handle_subscribe(); > > t_release(); > > } > > exit; > > #!endif > > # if presence enabled, this part will not be executed > > if (is_method("PUBLISH") || $rU==$null) > > { > > sl_send_reply("404", "Not here"); > > exit; > > } > > return; > > } > > # Authentication route > > route[AUTH] { > > #!ifdef WITH_AUTH > > if (is_method("REGISTER")) > > { > > # authenticate the REGISTER requests (uncomment to enable auth) > > if (!www_authorize("$td", "subscriber")) > > { > > www_challenge("$td", "0"); > > exit; > > } > > if ($au!=$tU) > > { > > sl_send_reply("403","Forbidden auth ID"); > > exit; > > } > > } else { > > #!ifdef WITH_IPAUTH > > if(allow_source_address()) > > { > > # source IP allowed > > return; > > } > > #!endif > > # authenticate if from local subscriber > > if (from_uri==myself) > > { > > if (!proxy_authorize("$fd", "subscriber")) { > > proxy_challenge("$fd", "0"); > > exit; > > } > > if (is_method("PUBLISH")) > > { > > if ($au!=$tU) { > > sl_send_reply("403","Forbidden auth ID"); > > exit; > > } > > } else { > > if ($au!=$fU) { > > sl_send_reply("403","Forbidden auth ID"); > > exit; > > } > > } > > consume_credentials(); > > # caller authenticated > > } else { > > # caller is not local subscriber, then check if it calls > > # a local destination, otherwise deny, not an open relay here > > if (!uri==myself) > > { > > sl_send_reply("403","Not relaying"); > > exit; > > } > > } > > } > > #!endif > > return; > > } > > # Caller NAT detection route > > route[NAT] { > > #!ifdef WITH_NAT > > force_rport(); > > if (nat_uac_test("19")) { > > if (method=="REGISTER") { > > fix_nated_register(); > > } else { > > fix_nated_contact(); > > } > > setflag(FLT_NATS); > > } > > #!endif > > return; > > } > > # RTPProxy control > > route[RTPPROXY] { > > #!ifdef WITH_NAT > > if (is_method("BYE")) { > > unforce_rtp_proxy(); > > } else if (is_method("INVITE")){ > > force_rtp_proxy(); > > } > > if (!has_totag()) add_rr_param(";nat=yes"); > > #!endif > > return; > > } > > # Routing to foreign domains > > route[SIPOUT] { > > if (!uri==myself) > > { > > append_hf("P-hint: outbound\r\n"); > > route(RELAY); > > } > > } > > # PSTN GW routing > > route[PSTN] { > > #!ifdef WITH_PSTN > > # check if PSTN GW IP is defined > > if (strempty($sel(cfg_get.pstn.gw_ip))) { > > xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); > > return; > > } > > # route to PSTN dialed numbers starting with '+' or '00' > > # (international format) > > # - update the condition to match your dialing rules for PSTN routing > > if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) > > return; > > # only local users allowed to call > > if(from_uri!=myself) { > > sl_send_reply("403", "Not Allowed"); > > exit; > > } > > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); > > route(RELAY); > > exit; > > #!endif > > return; > > } > > # XMLRPC routing > > #!ifdef WITH_XMLRPC > > route[XMLRPC] > > { > > # allow XMLRPC from localhost > > if ((method=="POST" || method=="GET") > > && (src_ip==127.0.0.1)) { > > # close connection only for xmlrpclib user agents (there is a bug in > > # xmlrpclib: it waits for EOF before interpreting the response). > > if ($hdr(User-Agent) =~ "xmlrpclib") > > set_reply_close(); > > set_reply_no_connect(); > > dispatch_rpc(); > > exit; > > } > > send_reply("403", "Forbidden"); > > exit; > > } > > #!endif > > # Sample branch router > > branch_route[BRANCH_ONE] { > > xdbg("new branch at $ru\n"); > > } > > # Sample onreply route > > onreply_route[REPLY_ONE] { > > xdbg("incoming reply\n"); > > #!ifdef WITH_NAT > > if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) > > && status=~"(183)|(2[0-9][0-9])") { > > force_rtp_proxy(); > > } > > if (isbflagset("6")) { > > fix_nated_contact(); > > } > > #!endif > > } > > route[STATS] { > > sql_query("ca", > > "insert into statistics (time_stamp,shm_used_size,shm_real_used_size," > > "shm_max_used_size,shm_free_used_size,ul_users,ul_contacts) values ($Ts," > > "$stat(used_size),$stat(real_used_size),$stat(max_used_size)," > > "$stat(free_size),$stat(location-users),$stat(location-contacts))", > > "ra"); > > } > > # Sample failure route > > failure_route[FAIL_ONE] { > > #!ifdef WITH_NAT > > if (is_method("INVITE") > > && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { > > unforce_rtp_proxy(); > > } > > #!endif > > if (t_is_canceled()) { > > exit; > > } > > # uncomment the following lines if you want to block client > > # redirect based on 3xx replies. > > ##if (t_check_status("3[0-9][0-9]")) { > > ##t_reply("404","Not found"); > > ## exit; > > ##} > > # uncomment the following lines if you want to redirect the failed > > # calls to a different new destination > > ##if (t_check_status("486|408")) { > > ## sethostport("192.168.2.100:5060"); > > ## append_branch(); > > ## # do not set the missed call flag again > > ## t_relay(); > > ##} > > } > > On Wed, Oct 5, 2011 at 12:12 PM, Uri Shacked <ushacked(a)gmail.com> wrote: > >> >> >> ---------- Forwarded message ---------- >> From: Uri Shacked <uri.shacked(a)gmail.com> >> Date: Wed, Oct 5, 2011 at 11:48 AM >> Subject: dialog ref Error >> To: sr-users(a)lists.sip-router.org >> >> >> Hi, >> >> i keep on getting this error: >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_hash.c:247]: new dialog on hash 531 >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_handlers.c:264]: route_set , contact sip:036264529@10.2.0.55:5061, >> cseq 1 and bind_addr udp:10.2.0.5:5060 >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_hash.c:519]: ref dlg 0xb417f5a0 with 3 -> 3 >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_hash.c:600]: unref dlg 0xb417f5a0 with 1, crt ref count: 3 >> >> anyone knows what it is? >> >> here is the dialog part in my cfg: >> >> >> *# ------ dialog params --------* >> >> modparam("dialog", "db_url", DBURL) >> >> ##modparam("dialog", "rr_param", "did") >> >> modparam("dialog", "default_timeout", 7200) >> >> modparam("dialog", "dlg_match_mode", >> 0) >> >> modparam("dialog", "detect_spirals", 1) >> >> modparam("dialog", "db_mode", 1) >> >> ##modparam("dialog", "db_fetch_rows", 100) >> >> modparam("dialog", "dlg_flag", FLT_DLG) >> >> modparam("dialog", "profiles_with_value", "test") >> >> >> >> .......... >> >> dlg_manage(); >> >> ##dlg_setflag(FLT_DLG); >> >> if (is_method("INVITE")) { >> >> $var(size) = 0; >> >> get_profile_size("test","$rU","$var(size)"); >> >> xlog("currently, the user $rU has $var(size) active calls\n"); >> >> if( $var(size) >= 5 ){ >> >> sl_send_reply("507", "Simultaneous calls limit reached"); >> >> xlog("Limit of $var(size) active calls has been reached \n"); >> >> exit;} >> >> set_dlg_profile("test","$rU");} >> >> ........ >> >> >

2 1

Dialog ref Error
by Uri Shacked 05 Oct '11

05 Oct '11

---------- Forwarded message ---------- From: Uri Shacked <uri.shacked(a)gmail.com> Date: Wed, Oct 5, 2011 at 11:48 AM Subject: dialog ref Error To: sr-users(a)lists.sip-router.org Hi, i keep on getting this error: Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_hash.c:247]: new dialog on hash 531 Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_handlers.c:264]: route_set , contact sip:036264529@10.2.0.55:5061, cseq 1 and bind_addr udp:10.2.0.5:5060 Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_hash.c:519]: ref dlg 0xb417f5a0 with 3 -> 3 Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_hash.c:600]: unref dlg 0xb417f5a0 with 1, crt ref count: 3 anyone knows what it is? here is the dialog part in my cfg: *# ------ dialog params --------* modparam("dialog", "db_url", DBURL) ##modparam("dialog", "rr_param", "did") modparam("dialog", "default_timeout", 7200) modparam("dialog", "dlg_match_mode", 0) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 1) ##modparam("dialog", "db_fetch_rows", 100) modparam("dialog", "dlg_flag", FLT_DLG) modparam("dialog", "profiles_with_value", "test") .......... dlg_manage(); ##dlg_setflag(FLT_DLG); if (is_method("INVITE")) { $var(size) = 0; get_profile_size("test","$rU","$var(size)"); xlog("currently, the user $rU has $var(size) active calls\n"); if( $var(size) >= 5 ){ sl_send_reply("507", "Simultaneous calls limit reached"); xlog("Limit of $var(size) active calls has been reached \n"); exit;} set_dlg_profile("test","$rU");} ........

2 1

kamailio sip-capture error on compile
by Jeff Anderson 05 Oct '11

05 Oct '11

I am trying to get kamailio sip-capture up and running for use with homer. I can get the service to start but i receive the following error. [root@homer kamailio]# /usr/local/sbin/kamailio -c loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(10208) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 Listening on udp: 10.40.0.22:5060 Aliases: config file ok, exiting... [root@homer kamailio]# kamctl start INFO: Starting Kamailio : INFO: started (pid: 10226) [root@homer kamailio]# [root@homer kamailio]# tail -f /var/log/messages Oct 3 05:30:17 homer /usr/local/sbin/kamailio[1554]: NOTICE: <core> [main.c:704]: Thank you for flying kamailio Oct 3 05:30:32 homer kamailio: WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 Oct 3 05:31:59 homer kernel: device eth0 entered promiscuous mode Oct 3 05:32:47 homer kernel: device eth0 left promiscuous mode Oct 3 05:52:10 homer /usr/local/sbin/kamailio[1753]: NOTICE: <core> [main.c:704]: Thank you for flying kamailio Oct 3 05:52:12 homer kamailio: WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 Oct 3 05:55:35 homer kernel: device eth0 entered promiscuous mode Oct 3 05:57:46 homer kernel: device eth0 left promiscuous mode Oct 3 07:05:35 homer /usr/local/sbin/kamailio[1894]: NOTICE: <core> [main.c:704]: Thank you for flying kamailio Oct 3 07:18:43 homer kamailio: WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 ^C [root@homer kamailio]# Does anyone have any ideas on how to resolve this error? Thanks, Jeff Anderson

3 3

Installing Versign Intermediate CA Certificate In Kamailio 3.1.5
by arif tuhin 05 Oct '11

05 Oct '11

I've got trial certificate from verisign. They signed the server certificate with an intermediate ca and gave instructions for installing it into apache/IIS etc. Both issued certificate and the intermediate CA should be installed in the server. if only server certificate is installed , it will not be trusted by default. Now problem is in apache mod_ssl there is a field called ServerChainFile where the intermediate ca name should be in place, but in the tls module documentation, i find no place for intermediate ca. How should i install the intermediate ca the? Thanks in advance.

1 0

db_text
by Bruno Bresciani 04 Oct '11

04 Oct '11

Hello ALL, Someone knows why at line 149 in .../kamailio-3.1.2/modules_k/ db_text/dbt_tb.c the shm_malloc function sometimes allocates memory to dtp->prev but sometimes doesn't? I'm asking this question because when the subscriber table is being loading to shared memory in db_text module, dtp->prev is allocated but on version, dispatcher, location table it doesn't... Analysing the source code I can't see any reason to this behavior. The allocation of dtp->prev to subscriber table is generating a core at db_text module. Best Regards

2 5

Siremis CDR problems
by Klaus Darilion 04 Oct '11

04 Oct '11

Hi! When populating cdrs table from timer route I get some errors: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Duplicate entry '088827BCD0738B6F(a)111.11.111.11-00E0F510042319395872000014D3-6ADB' for key 2 ERROR: <core> [db_query.c:130]: error while submitting query ERROR: sqlops [sql_api.c:217]: cannot do the query Checking the acc table for those key I found the entries in the attachment. Maybe it be that the stored procedure can not handle PRACK or reINVITEs correctly? Thanks Klaus

2 3

new wiki system
by Daniel-Constantin Mierla 04 Oct '11

04 Oct '11

Hello, I installed a new wiki site, targeting to collect documentation for upcoming v3.2.0 and further releases. You can find it at: * http://www.kamailio.org/wiki/ Hopefully, it will make the process of updating/migrating the relevant documentation from old wiki (http://www.kamailio.org/dokuwiki/) straightforward and offer a clean reference for upcoming stable version. Old wiki is in place since 2005, collected a lot of pages, but many are for very old versions of kamailio, making difficult, at least for new comers, to sort out documentation for latest stable. All of you are invited to help with this process of adding documentation for 3.2.0 on the new wiki. The old wiki will stay in place, it will not be removed. Both are based on Dokuwiki, using different themes. The new one is at the beginning, so no much content in there yet, but by the time of releasing 3.2.0, should contain sufficient documentation for get started with that version. One very demanded thing was the alphabetic index for parameters, functions and mi commands from modules. This page contains new generated indexes for version 3.2.0. More details in a separate email. If you have questions about the new wiki, don't know where to place some tutorial, feel free to ask on mailing lists. Ultimately, pages can be relocated, but try to use namespaces for related topics. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 4

dispatcher, LCR, carrierroute or mtree?
by Skyler 04 Oct '11

04 Oct '11

Hi all, I need to setup an inbound DID router for 8 proxies and ~300 DID's. Very simple, a call comes in from pstn > kamailio looks up DID for exact match in mysql > returns location > If db result returns noservice redirect to 1.2.3.4 else remain in call-flow and send to destination. Having a failure route capability would be nice. From what I've read on each of the 4 modules, any could do what I need though each have their own complexities. I'm still new to kamailio and hoping to utilize the group here for some advice. I'd like to keep this as simple as possible then adding/deleting DID's from the db as needed for routing. In your experience which would be the best module to use in order to achieve my goal? TIA, Skyler

4 5

Trouble Setting up Kamailio
by Peter Schrock 04 Oct '11

04 Oct '11

I am using the site: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git to load up Kamailio on my ppc using Leopard X. I managed to figure out the mysql issue and got it to install, but now there are some variances to the instruction from the stated install site and I am kind of lost. If I could get some help with the following - "7. Create MySQL database" This whole section seems confusing. It doesn't say what I am suppose to look for. I am sure what it says is all you need to do, but I am not sure if it is completed correctly or not. - "8. Edit configuration file" This section could use more specific information as to how to set up the configuration file. - "9. The init.d script" This section says I am suppose to put "*kamailio.init*" in the directory "*/etc/init.d*", then I am suppose chmod after it is copied. First of, I don't have that directory, am I suppose to create it or is it suppose to be created already? Also, it says to "*adduser*" with some options. Every time I try, I get the error message that it doesn't recognize the "*adduser*" command. Please help Peter

2 1

New to Kamailio - make problem
by Peter Schrock 03 Oct '11

03 Oct '11

I can't seem to get kamailio to "*make*". I am following the directions on the wiki on how to install. I have downloaded, compiled, and installed mysql server and every time I run to "*make all*" with kamailio, I get this error message. At first, I was thinking that *km_db_mysql.c* couldn't find the file. The header seems to be looking for *mysql.h* (at *#include <mysql/mysql.h>*). I found the file from installing mysql server and I thought I'd help * km_db_mysql.c* to find the file, but it still gives me this error. I am using a power pc running OS X.5.1. CC (gcc) [M db_mysql.so] km_db_mysql.o km_db_mysql.c:49:19: error: mysql/mysql.h: No such file or directory km_db_mysql.c: In function ‘kam_mysql_mod_init’: km_db_mysql.c:92: warning: implicit declaration of function ‘mysql_get_client_info’ km_db_mysql.c:92: warning: format ‘%s’ expects type ‘char *’, but argument 7 has type ‘int’ km_db_mysql.c:92: warning: format ‘%s’ expects type ‘char *’, but argument 6 has type ‘int’ make[1]: *** [km_db_mysql.o] Error 1 make: *** [modules] Error 1 Peter Schrock

3 3

Combining Dispatcher and NAT_Helper modules.
by Henrik Aagaard Sørensen 03 Oct '11

03 Oct '11

I'm currently having a Kamailio running as load balancer, at it works most of the time. But sometimes fails due to NAT issues. This is my config: http://pastebin.com/8F8X45xh Can someone help me combine the NAT_Helper module into this config? It's not the variables to set or RTPProxy, but in the Routing Logic.

1 0

Incoming calls question
by Javier Vidal -- Quasar 03 Oct '11

03 Oct '11

i have a question, i have to make a system to recieve several calls and the same time, and depending of destination DDI i have to rewrite the URI's IP. I am not going to have Registered users, only IP validation. The question is: What is the better way to consult thousands DDIs and get the IP from mysql? I had thought, use sqlops.so module, but i think that i could have problems with it. Another posibility is use the carrierroute and use the full pattern for the DDI. Thank for all sugestions. Javier V.

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alphabetic indexes for module parameters, functions, ...
by Daniel-Constantin Mierla 03 Oct '11

03 Oct '11

Hello, for the last versions the alphabetic indexes were not generated - there were different people doing that in the past, using various self-written tools. For upcoming version, 3.2.0, I spend some time to structure the docbook files for modules in order to be easier to get from them the names for functions, parameters and mi commands. Therefore, the new wiki system has fresh generated indexes from modules' documentations. You can find them at: * http://www.kamailio.org/wiki/alphaindexes:3.2.x:modparameters * http://www.kamailio.org/wiki/alphaindexes:3.2.x:modfunctions * http://www.kamailio.org/wiki/alphaindexes:3.2.x:micommands Being generated from docbook xml files, the indexes contain name, path to the module in source tree and module name, the last one with link to html readme file of the module published to kamailio.org site. There is no link at the moment pointing inside the html readmes to the place where the item is documented, but each readme has table of content, so it is easy to spot it. We have lots lots of modules right now, therefore it was not easy to check if some parameters, functions or mi commands were not indexed. If you find some missing, please write back here so we can fix the dockbook or the indexing tool, depending where the problem is. Hope the indexes will be helpful for many of you. Ending the email with another invitation to all of you to port (update to upcoming v3.2.0) your favorite tutorial from old wiki system (http://www.kamailio.org/dokuwiki/) and add to the new wiki (http://www.kamailio.org/wiki/). Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Cflag is not getting updated.
by Krishna Kurapati 03 Oct '11

03 Oct '11

Hi, I am hitting bit of a wall here. - I am setting bflags for NAT, and SIPPING before save("location") however, when I do "kamctl ul show", the Cflag is 0. I also checked see if the bflags are set or not in the register routine - they are being set. What am I missing? snippets of the config files. #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_method", "INFO") modparam("nathelper", "sipping_from", "sip:pinger@50.62.1.7") # params needed for NAT traversal in other modules modparam("nathelper|registrar|usrloc", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); setbflag(FLB_NATB); } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { xlog("L_ALERT", "REGISTER: From $fU, $fd To $tU, $td"); if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } # 7/28/11 - make sure only one contact location is active # at any given time # unregister("location", "$rU"); if (isbflagset(FLB_NATB)) { xlog("L_ALERT", "REGISTRAR: For $fU, $fd FLB_NATB set"); } if (isbflagset(FLB_NATSIPPING)) { xlog("L_ALERT", "REGISTRAR: For $fU, $fd FLB_NATSIPPING set"); } if (!save("location")) { xlog("L_ALERT", "REGISTRAR: From $fU, $fd Location could not be saved"); sl_reply_error(); } exit; } } The debug log snippets ================== Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: ALERT: <script>: REGISTER: From p_krishna_test_com, 50.62.1.7 To p_krishna_test_com, 50.62.1.7 Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: ALERT: <script>: REGISTRAR: For p_krishna_test_com, 50.62.1.7 FLB_NATB set Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: ALERT: <script>: REGISTRAR: For p_krishna_test_com, 50.62.1.7 FLB_NATSIPPING set Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: DEBUG: sl [sl.c:278]: reply in stateless mode (sl) Output of kamctl command ==================== $ sudo kamctl ul show Domain:: location table=512 records=1 max_slot=1 AOR:: p_krishna_test_com Contact:: sip:p_krishna_test_com@75.62.17.45:50073;transport=tcp;ob Q= Expires:: 287 Callid:: ad2d3a7bf39a4345861db1661576a7f3 Cseq:: 54856 User-agent:: QliqDesktop 1.0.0 (pjproject 1.10.0/win32) State:: CS_NEW Flags:: 0 Cflag:: 0 <================= Cflag is not being set. Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Thanks Krish Kura

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Kamailio as dispatcher fails due to from domain on extension.
by Henrik Aagaard Sørensen 02 Oct '11

02 Oct '11

I'm using Kamailio as dispatcher to a FreeSwitch setup. Currently it "almost" work. I have 2 extensions connected to my FreeSwitch via Kamailio. However, I can only send calls one way because Kamailio seems to change the domain on the extension. One extension is registered with ID(a)sip.my-domain.com. This I can send calls to. The other extension is registered with ID@local-ip. This I cannot send calls to. Isn't it possible to have Kamailio register the extension on my FreeSwitch with @sip.my-domain.com? If I register both extension directly to my FreeSwitch everything works.

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Re: [SR-Users] Setting avp from load_credentials
by Alex Balashov 01 Oct '11

01 Oct '11

Alternately, if you dislike any of these paradigms or find the order of operations inconvenient, just roll your own. That's what sqlops is for. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Oct 1, 2011, at 10:43 AM, "graham(a)g-rock.net" <graham(a)g-rock.net> wrote: > I know I am Auth'ing users, but apparently need to confirm the location of www_authorize. Thanks for the pointers Alex. > > Sent from my HTC on the Now Network from Sprint! > > ----- Reply message ----- > From: "Alex Balashov" <abalashov(a)evaristesys.com> > Date: Sat, Oct 1, 2011 9:07 am > Subject: [SR-Users] Setting avp from load_credentials > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users(a)lists.sip-router.org> > > Sure it is, but credentials only get loaded if you are doing proxy/www_authorize() beforehand. Are you? > > -- > This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. > > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On Oct 1, 2011, at 9:24 AM, Graham Wooden <graham(a)g-rock.net> wrote: > > > Hi there, > > > > Just doing a quick test and this doesn’t appear to be working. Kamailio > > 1.5.5-notls. Snippet of code: > > > > modparam("auth_db", "load_credentials", "username") > > . > > . > > . > > xlog("username is $avp(s:username)\n"); > > > > > > The result is '<null>'. Is this not available in the route{} ? > > > > Thanks, > > > > -graham > > > > > > > > > > > > _______________________________________________ > > 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 > >

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Re: [SR-Users] Setting avp from load_credentials
by graham＠g-rock.net 01 Oct '11

01 Oct '11

I know I am Auth'ing users, but apparently need to confirm the location of www_authorize. Thanks for the pointers Alex. Sent from my HTC on the Now Network from Sprint! ----- Reply message ----- From: "Alex Balashov" <abalashov(a)evaristesys.com> Date: Sat, Oct 1, 2011 9:07 am Subject: [SR-Users] Setting avp from load_credentials To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users(a)lists.sip-router.org> Sure it is, but credentials only get loaded if you are doing proxy/www_authorize() beforehand. Are you? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Oct 1, 2011, at 9:24 AM, Graham Wooden <graham(a)g-rock.net> wrote: > Hi there, > > Just doing a quick test and this doesn’t appear to be working. Kamailio > 1.5.5-notls. Snippet of code: > > modparam("auth_db", "load_credentials", "username") > . > . > . > xlog("username is $avp(s:username)\n"); > > > The result is '<null>'. Is this not available in the route{} ? > > Thanks, > > -graham > > > > > > _______________________________________________ > 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

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Setting avp from load_credentials
by Graham Wooden 01 Oct '11

01 Oct '11

Hi there, Just doing a quick test and this doesn¹t appear to be working. Kamailio 1.5.5-notls. Snippet of code: modparam("auth_db", "load_credentials", "username") . . . xlog("username is $avp(s:username)\n"); The result is '<null>'. Is this not available in the route{} ? Thanks, -graham

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Use of $rU
by Graham Wooden 01 Oct '11

01 Oct '11

Hi there, What is pv for the equivalent of $rU for outbound? My inbound use of $rU has the correct username. All my subscribers are multi-number trunks, so on the outbound I need to associate the subscriber. This is kamailio 1.5. Thanks, -graham

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Handling quota/call limits - kamailio 1.5
by Graham Wooden 01 Oct '11

01 Oct '11

Hi there, Finally getting around to introducing some call limits and starting with the inbound calls. However, I ran into an issue about how to get the "in use" count to be updated in my quota table. What is the best way to get s:in_use to update so it can be calculated? Here is my snippet of code: if (is_method("INVITE")) { avp_db_query("select quota, in_use from quota where username='$rU'","$avp(s:quota);$avp(s:in_use)"); if ($avp(s:in_use) >= $avp(s:quota)){ xlog ("DEBUG: Trunk $rU is over quota"); sl_send_reply("503","User over quota"); } . . } This opens up two more questions: How to decrease the call count when a BYE is received - maybe do a subtraction and the do a avp_db_query to update the table? And of course there is the BYE that won't be present (ie. When a call just rings and hangs up before it gets answered). Maybe ACC's missed_calls can somewhat be used to help determine? Thanks for any pointers and guidance. I am running Kamailio 1.5. Thanks, -graham

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SNMPStats and multiple proxies
by Antanas Masevicius 30 Sep '11

30 Sep '11

Hello, been trying to run two sip-router proxies in parallel and to feed data from them to snmpd via agentx. But it looks like snmpd deamon directly routes snmp request just to one (probably first connected) proxy. What ways this can be solved? Currenly i am thinking like running two concurrent snmpd daemons binded to specific IP addresses. Is there a better way to do it? best regards, -- Antanas Masevicius

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Re: [SR-Users] Making iPhone SIP client work with Kamailio
by Krishna Kurapati 30 Sep '11

30 Sep '11

> > Hi, > > The iPhone client, when it goes into background, can only respond to > incoming messages. > That makes the Registration expire when there is no communication from the > server before > the expire time. I could make the Registration not expire at all. Not sure > it's the only option. > > Is there a way to send a Ping message from server periodically to the > iPhone SIP client. This > will wakeup the SIP client and resend REGISTER message. > > Thanks > Krish Kura >

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Cleaning up presence info when user is deleted
by Krishna Kurapati 30 Sep '11

30 Sep '11

Hi, When a user is removed using "kamctl rm", is there way way to clean up presence information in the database (xcap, presentity, watcher and active watcher tables) and in the cache? Is there a event that gets fired when the user is removed from subscriber table which I could handle in config file and and cleanup? Thanks Krish Kura

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Core db_text module
by Bruno Bresciani 30 Sep '11

30 Sep '11

Hi All, Kamailio 3.1.2 generate a core in db_text module when subscriber table is updated and after this action some SIP message require authentication process... Someone Can Help me to understand why this core is happening? Below is part of kamailio's trace and gdb core too. Kamailio's trace: *Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2327]: DEBUG: auth_db [authorize.c:239]: realm value [192.168.166.199]* Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2327]: DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1 *Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2327]: DEBUG: db_text [dbt_file.c:78]: [subscriber] was updated* Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2324]: ALERT: <core> [main.c:741]: child process 2327 exited by a signal 11 Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2324]: ALERT: <core> [main.c:744]: core was generated Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2324]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2335]: INFO: <core> [main.c:807]: INFO: signal 15 received c Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2335]: DEBUG: <core> [main.c:818]: Memory status (pkg): gdb core trace: Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 11, Segmentation fault. *#0 0x00f21603 in dbt_db_del_table (_dc=0xb6071e60, _s=0xbf837040, sync=0) at dbt_lib.c:238* 238 dbt_lib.c: Arquivo ou diretório não encontrado. in dbt_lib.c Best Regards

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Re: [SR-Users] simple question but can't find the answer
by Daniel-Constantin Mierla 30 Sep '11

30 Sep '11

Hello, On 9/30/11 12:11 AM, JJ Zeng wrote: > > Thanks Daniel. I followed your advice and install kamailio 3.1. Now > I get a weird problem: it works 1 out 5 times for the relaying. > try to set like tcp_connection_lifetime=3604 global parameter, or to a value higher than the longer registration time. > > The t_relay is where the the problem is, because I check the response > code and route(LOCATION) always succeeds, which means the entries > exist in "location". > > If I don't put set_forward_no_connect, then t_relay will take a long > time until my client time out. If I put it, then the client will get > 477 right away. > > BTW, the original config didn't work, either. That's why I started to > strip down some irrelevant stuff. > What errors you got with the default config? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Placing ds_select_dst in kamailio.cfg, where?
by Daniel-Constantin Mierla 29 Sep '11

29 Sep '11

Hello, please keep the mainling list cc-ed. Calling ds_select_dst() first in the main route block is not really a good place. First you have to handle sanity checks, within dialog requests a.s.o. Of course it is a matter of your purpose, but as general hint, selecting the destination should be done just (or as close as possible) before relaying. If you take default config and you have only load balancing, then you can replace location lookup part with load balancing. Other parts may be stripped as well, just giving some direction here. Cheers, Daniel On 9/28/11 11:11 PM, Henrik Aagaard Sørensen wrote: > That did the job. > > I just placed the ds_select_dst-code at the very first thing after route{ > > Thank you for your help! > > 2011/9/28 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > the devel version of dispatcher module README has a different > config example, maybe it helps you more: > > http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2522847 > > I am considering to update it again soon to be a bit more complete > and follow the structure of the default config file shipped with > kamailio. > > Cheers, > Daniel > > > On 9/28/11 6:30 PM, Henrik Aagaard Sørensen wrote: >> Hi Carsten. >> >> The configuration works. It's actually where (which line) in the >> configuration I should put: >> ds_select_dst("1", "0"); >> >> :) I'm not strong in Kamailio and just want it to load balance >> between Freswitch servers. So I have a basic installation of >> Kamailio, without MySQL, NAT, Auth etc. >> >> On Wed, Sep 28, 2011 at 5:09 PM, Carsten Bock >> <carsten(a)ng-voice.com <mailto:carsten@ng-voice.com>> wrote: >> >> Hi, >> >> When you've got problems with placing "ds_select_dst()" in your >> config, you should do the following: >> - Open an Editor of your choice >> - go to the line, where you want to place it >> - type: ds_select_dst("1", "0"); >> - save & exit >> >> Just kidding! You should do the following with your config: >> kamailio -f <path-to-your-config> -c >> >> What error is displayed? If none, then check your syslog for >> errors. >> Could you post your config? It is better to put a full path >> in the >> config to the dispatcher-file... otherwise it might be >> dependent of, >> where you are running the init.d-skript/kamailio.... >> >> I am using dispatcher in Kamailio 1.5, 3.1 and 3.2, >> everythings works >> fine with me. >> >> Kind regards, >> Carsten >> >> 2011/9/28 Henrik Aagaard Sørensen >> <henrikaagaardsorensen(a)gmail.com >> <mailto:henrikaagaardsorensen@gmail.com>>: >> > I've installed 2 Freeswitch on different servers and would >> like to have >> > Kamailio load balance between them. >> > >> > I've installed Kamailio following: >> > >> http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git >> > >> > I've created dispatcher.list and added: >> > >> > loadmodule "dispatcher.so" >> > >> > modparam("dispatcher", "list_file", "../etc/dispatcher.list") >> > >> > in the kamailio.cfg. >> > >> > But looking at: >> > >> http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html >> > I'm having difficulties to placing: >> > >> > ds_select_dst("1", "0"); >> > >> > in the kamailio.cfg that was created originally. >> > >> > Can anyone help me? >> > >> > _______________________________________________ >> > 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 >> > >> > >> >> >> >> -- >> Carsten Bock >> CEO (Geschäftsführer) >> >> ng-voice GmbH i. Gr. >> Schomburgstr. 80 >> D-22767 Hamburg / Germany >> >> http://www.ng-voice.com >> mailto:carsten@ng-voice.com <mailto:carsten@ng-voice.com> >> >> Mobile +49 179 2021244 <tel:%2B49%20179%202021244> >> Office +49 40 34927219 <tel:%2B49%2040%2034927219> >> Fax +49 40 34927220 <tel:%2B49%2040%2034927220> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Logmessage when Dispatcher-Module sets a dispatcher to probing
by Timo Klecker 29 Sep '11

29 Sep '11

Hey again, Btw: I am using Kamailio 3.1.5 and found the event_route in 3.2.x Dispatcher documentation. Is there a possibility in 3.1.x? Greetings Timo Von: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Timo Klecker Gesendet: Donnerstag, 29. September 2011 11:10 An: 'kamailio users' Betreff: [SR-Users] Logmessage when Dispatcher-Module sets a dispatcher to probing Hey, I'd like to have a log-entry in the kamailio logfile when #ds_probind_threshhold OPTIONS pings fail to reach a dispatcher (408 localy generated after #fr_timer ms). Is it possible to do this without changing code? Greetings Timo

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simple question but can't find the answer
by JJ Zeng 29 Sep '11

29 Sep '11

Hi, I am really new to ser(just installed it today). My situation is here: Both A and B make tcp connection to ser proxy server. If A sends "Invite" to B, ser would look up the tcp connection of B, and relay the sip request. If B is not connected, then 480 is returned to A. When B sends back the response, ser would relay it to A. It looks really simple, but I just can't figure out how to do it. Thanks. JJ

2 3

Logmessage when Dispatcher-Module sets a dispatcher to probing
by Timo Klecker 29 Sep '11

29 Sep '11

Hey, I'd like to have a log-entry in the kamailio logfile when #ds_probind_threshhold OPTIONS pings fail to reach a dispatcher (408 localy generated after #fr_timer ms). Is it possible to do this without changing code? Greetings Timo

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OpenIMS
by Carlos Ruiz Diaz 29 Sep '11

29 Sep '11

Hello list. I'm trying to test/implement http://www.openimscore.org/ to setup an IMS scenario for development purposes. The project seems quite inactive to me and I wonder if there is any other similar project or this one just died and is no longer maintained. The documentation didn't specify if there is any kind of compatibility with Kamailio and based on what was written, only SER core architecture is supported. Any help will be appreciated. Regards. Carlos.

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documentation update for 3.2.0 and wiki system
by Daniel-Constantin Mierla 28 Sep '11

28 Sep '11

Hello, in preparation for release 3.2.0, I was checking our wiki (on kamailio.org), which is quite rich in information, but some tutorials are for older versions. Maybe it is also a good time to do some restructuring along with the update of the useful tutorials to be valid for 3.2.0. For restructuring, the main goal is to provide some ways to let people find the documentation they need (and it is available there) easier. Some ideas, introducing categorization/tags for specific topics (e.g., radius, ldap, presence, asterisk, ...), introducing some index pages to link tutorials per version, etc. More suggestions are welcome here, let us know what will make your life easier in browsing the docs. On another hand, currently wiki is based on dokuwiki, which from my point of view did good job so far, but maybe we are missing another excellent wiki system that we should use, because provides more tools for this kind of purpose. One thing I was missing, was export to pdf, which I may be possible right now with dokuwiki plugins -- or even better, we can get something to export to docbook xml format with dokuwiki or other wiki system. In the same kind of subject, what about using git to store (or at least have a replica of) the wiki pages? IIRC, I saw somewhere that even for dokuwiki is possible to configure to use a git repository to store the wiki content instead of local CVS. This will allow people to mirror very easy the content. I could spot from time to time web spiders downloading the wiki. Comments or other suggestions to make the access to documentation easier? Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Placing ds_select_dst in kamailio.cfg, where?
by Henrik Aagaard Sørensen 28 Sep '11

28 Sep '11

I've installed 2 Freeswitch on different servers and would like to have Kamailio load balance between them. I've installed Kamailio following: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git I've created dispatcher.list and added: loadmodule "dispatcher.so" modparam("dispatcher", "list_file", "../etc/dispatcher.list") in the kamailio.cfg. But looking at: http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html I'm having difficulties to placing: ds_select_dst("1", "0"); in the kamailio.cfg that was created originally. Can anyone help me?

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Sip-router gentoo ebuild v3.1.5
by caio 28 Sep '11

28 Sep '11

Hello, Just want to announce the availability of an ebuild for version 3.1.5 (kamailio and ser flavours) on Gentoo Portage Overlays. Here the git link: * http://git.overlays.gentoo.org/gitweb/?p=user/caio.git;a=tree;f=net-misc/si… The unofficial overlays portage website: * http://gpo.zugaina.org/net-misc/sip-router Important, this is not an official ebuild, and neither it's not at official gentoo portage tree. You can use *layman* to add this overlay. Also at pkg/ser/gentoo path of sip-router project you can find the latest version. If you need any guidance to probe this ebuild under your gentoo linux do not hesitate to contact me or provide some feedback here at the lists. Best regards, Claudio

2 2

Accounting only the 2nd branch of missed serial forked call
by Ozren Lapcevic 27 Sep '11

27 Sep '11

Hi, I'm having some problems accounting missed serial forked calls to mysql database. I have following setup. Each user can have up to two contacts: telephone number (routed to asterisk) and SIP URI. Users can specify which contact has higher priority - which one should ring first. There is also SEMS voicemail which is forked as 3rd serial call leg if there is no answer at first two contacts. For example, I have two users: oz(a)abc.hr and pero(a)abc.hr. pero(a)abc.hr also has set telephone number as alternative number if he is not reachable at sip:pero@abc.hr. Moreover, pero(a)abc.hr has voicemail turned on. When oz(a)abc.hr calls pero(a)abc.hr, first pero(a)abc.hr's SIP client rings, then if there is no answer and after the timeout telephone number rings and finally, if there is no answer at telephone and after the timeout INVITE is forked to SEMS. There are two interesting scenarios accounting-wise which can happened: 1. oz(a)abc.hr calls pero(a)abc.hr, there are no answers and call is forked to voicemail. 2. oz(a)abc.hr calls pero(a)abc.hr, there is no answer at SIP client, but pero answers call at telephone. When scenario 1 happens, I want to have only one log (row) in missed_calls table. When scenario 2 happens, I don't want to have a log in missed_calls table. To accomplish this,* I want to log only the 2nd branch of the forked call. However, there is either a bug in acc module or I'm doing something wrong, and I can't get Kamailio to log only the 2nd branch*. I think that I am setting the FLT_ACCMISSED flag correctly - after the 2nd branch is handled and prior to calling the RELAY route. Logs show that FLT_ACCMISSED flag is set prior to calling t_relay(), and there are no errors in debug log. I am using $ru = "something" to rewrite URI prior to forking request. I can easily set up logging of every call (two missed calls for serially forked call to two locations) by setting FLT_ACCMISSED flag for each INVITE. I can set up logging of every call's 1st branch, by reseting FLT_ACCMISSED flag when handling 2nd branch of the call. Interestingly, logging of only the 2nd branch of the serial forked call works when there is no forking to voicemail! Any ideas how to solve this problem? Bellow are important parts of my config file. I'm running kamailio 3.1.4. Cheers Ozren # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_user=$tU;dst_domain=$td;src_ip=$si") #!endif ... # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); if (src_ip != ****) { # NAT detection route(NAT); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); if ( is_method("INVITE") ) { route(DBALIASES); #check for user defined forking priorities and timers route(FORK); } # user location service route(LOCATION); route(RELAY); } #check for user defined forking priorities and timers route[FORK]{ sql_query("con", "select * from usr_pref_custom where uuid='$tu'", "pref"); $avp(uuid)=$dbr(pref=>[0,0]); $avp(email)=$dbr(pref=>[0,1]); $avp(prio1)=$dbr(pref=>[0,2]); $avp(prio2)=$dbr(pref=>[0,3]); $avp(timer1)=$dbr(pref=>[0,5]); $avp(timer2)=$dbr(pref=>[0,6]); if (strlen($avp(prio1))>5) { # user has multiple contacts, do serial forking setflag(FLT_USRPREF); # set counter if (!$avp(prio)) { $avp(prio) = 1; } # overwrite request URI with highest priority contact if ($avp(prio1) =~ "^sip:00") { $ru = $avp(prio1) + "@host"; xlog("L_INFO","PRIO 1 is tel number, RURI set: $ru"); } else { $ru = $avp(prio1); xlog("L_INFO","PRIO 1 is SIP URI, RURI set: $ru"); } } } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); #if users have priorities set, use FAIL_FORK failure route if ( isflagset(FLT_USRPREF) ) { t_on_failure("FAIL_FORK"); } } if (isflagset(FLT_ACCMISSED)) xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS SET"); else xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS NOT SET"); if (!t_relay()) { sl_reply_error(); } exit; } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO","WITHINDLG, loose_route()"); if (is_method("BYE")) { xlog("L_INFO","WITHINDLG, BYE, DO ACCOUNTING"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # USER location service route[LOCATION] { #skip if $ru is telephone number if ($ru =~ "^sip:00") { xlog("L_INFO","SKIP lookup..."); } else { if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } } # when routing via usrloc, log the missed calls also, but only if user doesn't have prios set if ( is_method("INVITE") && !(isflagset(FLT_USRPREF))) { setflag(FLT_ACCMISSED); } } # Failure route for forked calls failure_route[FAIL_FORK] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if ($avp(prio) >= 1) { $avp(prio) = $avp(prio) + 1; # handle 2nd branch if ( ($avp(prio) == 2) && ( isflagset(FLT_USRPREF) )) { t_on_failure("FAIL_FORK"); if ($avp(prio2) =~ "^sip:00") { xlog("L_INFO","FAIL FORK, PRIO 2 is tel number"); $ru = $avp(prio2) + "@host"; } else { xlog("L_INFO","FAIL FORK, PRIO 2 is SIP URI"); $ru = $avp(prio2); route(LOCATION); } setflag(FLT_ACCMISSED); } else { $avp(prio) = 0; $ru = $(avp(uuid)); rewritehostport("host:port"); xlog("L_INFO","FAIL FORK, VOICEMAIL email:$avp(email), ru:$ru, br: $br"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: Email-Address=$avp(email)\r\n"); } route(RELAY); } if (t_is_canceled()) { exit; } }

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Kamailio with IMS extensions build
by loic.trottin＠orange.com 27 Sep '11

27 Sep '11

Hi all, I try to compile Kamailio with IMS extensions from OpenIMSCore available in carstenbock/ims branch in the GIT repository. I have some problems with the modules "Client_Ro" and "pcscf" that do not compile: CC (gcc) [M Client_Ro.so] ims_ro.o ims_ro.c: In function 'send_ccr_interim': ims_ro.c:594: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) ims_ro.c:594: error: (Each undeclared identifier is reported only once ims_ro.c:594: error: for each function it appears in.) ims_ro.c: In function 'send_ccr_stop': ims_ro.c:749: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) ims_ro.c: In function 'Ro_Send_CCR': ims_ro.c:831: error: 'struct dlg_binds' has no member named 'get_dlg' ims_ro.c:835: error: 'struct dlg_binds' has no member named 'unref_dlg' ims_ro.c:886: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) CC (gcc) [M pcscf.so] pcc_avp.o pcc_avp.c:1459: error: 'AVP_IMS_Subscription_Id_Type_E164' undeclared (first use in this function) pcc_avp.c:1459: error: (Each undeclared identifier is reported only once pcc_avp.c:1459: error: for each function it appears in.) pcc_avp.c:1461: error: 'AVP_IMS_Subscription_Id_Type_SIP_URI' undeclared (first use in this function) Does anybody have encountered these problems? Is there a stable version tagged of the branch carstenbock/ims available in the GIT repository ? Best regards, Loïc. ******************************************************************************** IMPORTANT.Les informations contenues dans ce message electronique y compris les fichiers attaches sont strictement confidentielles et peuvent etre protegees par la loi. Ce message electronique est destine exclusivement au(x) destinataire(s) mentionne(s) ci-dessus. Si vous avez recu ce message par erreur ou s il ne vous est pas destine, veuillez immediatement le signaler a l expediteur et effacer ce message et tous les fichiers eventuellement attaches. Toute lecture, exploitation ou transmission des informations contenues dans ce message est interdite. Tout message electronique est susceptible d alteration. A ce titre, le Groupe France Telecom decline toute responsabilite notamment s il a ete altere, deforme ou falsifie. De meme, il appartient au destinataire de s assurer de l absence de tout virus. IMPORTANT.This e-mail message and any attachments are strictly confidential and may be protected by law. This message is intended only for the named recipient(s) above. If you have received this message in error, or are not the named recipient(s), please immediately notify the sender and delete this e-mail message. Any unauthorized view, usage or disclosure ofthis message is prohibited. Since e-mail messages may not be reliable, France Telecom Group shall not be liable for any message if modified, changed or falsified. Additionally the recipient should ensure they are actually virus free. ********************************************************************************

3 3

need little help to setting up pike module
by dirtytoni＠gmx.de 27 Sep '11

27 Sep '11

Hi to all on this list =) jUSt a tiny user question aboUt pike module(at least kamailio doc). are these values for sampling_time_unit seconds or minutes ? it's not in the doc. same 4 remove_latency. thanks for helping me out sorry to bother you best regards Toni -- Empfehlen Sie GMX DSL Ihren Freunden und Bekannten und wir belohnen Sie mit bis zu 50,- Euro! https://freundschaftswerbung.gmx.de

2 1

Account 403 to RADIUS
by Mino Haluz 26 Sep '11

26 Sep '11

Hi, I have this code: if ( is_user_in("From", "blocked") && is_method("INVITE")) { xlog("L_INFO", "XLOG: [number_and_ruri_checks] NOTICE: Account ($fu) to ($ru) is blocked"); sl_send_reply("403", " Account blocked "); exit; } I would like to account also this answer to Radius, now I have it only in application xlog. Thanks, Mino

3 5

Migration from v3.1.x to 3.2.0
by Daniel-Constantin Mierla 26 Sep '11

26 Sep '11

Hello, a kind reminder for everyone to add to the wiki page hints for migration from v3.1.x to v3.2.0. Especially to developers that know what kind of major changes they did to existing modules in 3.1.x. The wiki is open for anyone: * http://sip-router.org/wiki/install/3.1.x-to-3.2.x I just updated the section related to database structure changes. If you spot errors/missing parts, go ahead and fix it. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] I am getting this error in the log When REGISTER message is received from jitsi
by Daniel-Constantin Mierla 23 Sep '11

23 Sep '11

On 9/22/11 9:04 PM, Krishna Kurapati wrote: > Thanks Daniel, > > Here it is... > > [...] It looked ok for my eye. Can you run kamailio with higher debug level (debug=4 in config file) and send the log messages related to parsing the URI for AoR? Cheers, Daniel > > > On Thu, Sep 22, 2011 at 1:44 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > can you grab such REGISTER with ngrep (or wireshark) and past it here? > > Cheers, > Daniel > > > On 9/22/11 8:07 PM, Krishna Kurapati wrote: >> Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: >> registrar [common.c:75]: failed to parse Address of Record >> Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: >> registrar [save.c:822]: failed to extract Address Of Record >> >> Is there a problem when AoR contains "_" >> >> for example is AoR krish_kura(a)sip.org <mailto:krish_kura@sip.org> >> is not allowed? >> >> Thanks >> Krish Kura >> >> >> >> _______________________________________________ >> 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 > Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 4

Re: [SR-Users] CANCEL is not processed in onsend_route
by Daniel-Constantin Mierla 23 Sep '11

23 Sep '11

Hello, On 9/15/11 4:02 PM, Timo Klecker wrote: > Hi Daniel, > > I managed to get a log with debug 4 from one call. I had to take out all > IP-Addresses and Usernames, hope you can use it. the log confirmed what I expected: Sep 15 15:32:33 vux896 /sbin/kamailio[31594]: DEBUG: siptrace [siptrace.c:954]: no uas msg, local transaction Sep 15 15:32:33 vux896 /sbin/kamailio[31594]: DEBUG: tm [t_fwd.c:1242]: DEBUG: e2e_cancel: e2e cancel proceeding Because a new CANCEL is generated from scratch, incoming message is lost and no longer available to check the sip trace flag. Now, the question is how to decide whether the cancel is to be siptraced. I can lookup for incoming CANCEL and check if it has the siptrace flag set, but there are CANCELs in case of local timeout or parallel forking, so I think it is better to actually lookup the INVITE transaction and if the siptrace flag is set for it, record the cancel. If there is no INVITE transaction anymore, the cancel is forwarded stateless and can be captured in onsend route, although it should be normally discarded if it is known that the proxy works in statefull mode only. Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 2

carrierroute module
by Mino Haluz 23 Sep '11

23 Sep '11

Hi, I have a problem with configuring the carrierroute module, I do not clearly understand how it should work from the documentation. What I would like to do is that I have 2 gateways (A.A.A.A and B.B.B.B), lots of scan_prefixes with different rewrite prefixes. The traffic should be sent to the 1 gateway, and if it fails (5xx,4xx codes) it should go to the another. I have only one SIP domain (domain.com). The question is, how the tables (domain_name, carrierroute, carrierfailroute) should be filled? Can I achieve this scenario with the carrierroute module ? Thanks. Mino

1 0

XCAP over SIP
by Krishna Kurapati 23 Sep '11

23 Sep '11

Hi, The document http://kb.asipto.com/kamailio:presence:k31-made-simple describes that XCAP can be sent over SIP instead of over HTTP. I tried to do however, the server is unable to update presence rules. I was using PUBLISH method. I was using content-type = text... Has anyone used SIP to send over XCAP? Appreciate if someone could share a link or configuration changes required. Thanks Krish

3 3

I am getting this error in the log When REGISTER message is received from jitsi
by Krishna Kurapati 22 Sep '11

22 Sep '11

Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: registrar [common.c:75]: failed to parse Address of Record Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: registrar [save.c:822]: failed to extract Address Of Record Is there a problem when AoR contains "_" for example is AoR krish_kura(a)sip.org is not allowed? Thanks Krish Kura

2 1

perl module compilation on centos
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, while building the v3.1.5 RPMs, the perl module failed to compile on centos 5 and 6, with following error: error: `PL_unitcheckav' undeclared (first use in this function ) so its packaging was disabled. It works fine on opensuse and fedora. Someone on twitter tried with the package perl-ExtUtils-Embed (like in fedora) and worked. However, OpenSUSE Build System fails to find that package. Is anyone here with latest centos 5 or 6 that can try compilation of perl module in v3.1.5 and let me know the result? Few months ago, it didn't fail for v3.1.4. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 5

Re: [SR-Users] Kamailio and FreeSWITCH realtime integration, tutorial?
by Henrik Aagaard Sørensen 22 Sep '11

22 Sep '11

But FreeSwitch don't use MySQL like Asterisk? So adapting isn't that easy or...??? On 21/09/2011, at 16.18, David <david(a)styleflare.com> wrote: You can adapt pretty easily. On 9/21/11 3:46 PM, Henrik Aagaard Sørensen wrote: Does anyone know if there somewhere exists a tutorial about Kamailio and FreeSWITCH realtime integration? I have Googled a lot and found: http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index On the same site there is a tutorial for Kamailio and Asterisk realtime integration, but not for FreeSWITCH. _______________________________________________ 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 _______________________________________________ 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

2 2

Re: [SR-Users] Problem with REGISTER retransmissions
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, I tried to reproduce with a very simple config to handle register requests. So I sent a REGISTER with expires 60 (very much the same as the one in the trace you sent), waited a while, then sent two REGISTERs having the CSeq incremented. All went fine. I tested with master and 3.1 branches (so the last is actually 3.1.5). Can you upgrade to 3.1.5 first and try again? If still fails, then update config to print the return code for save() function, something like: $var(r) = save("location"); xlog("code returned by save is $var(r)\n"); if($var(r) < 0) send_reply("500", "Failure"); Let me know the value of the returned code for the register that does not have a contact in reply. Cheers, Daniel On 9/20/11 2:10 PM, Alejandro Mingo wrote: > > Hi Daniel-Constantin, > > I send you the PCAP and LOG files corresponding to one of these > situations. As you can see in the LOG file, one thread (5444) handles > the first REGISTER, other thread (5447) handles the second REGISTER > (retransmission) and a third thread (5451) indicates that the binding > has expired, six seconds after the arrival of both REGISTERs. > > Thanks again. > > Alejandro. > > > > El 20/09/2011 10:54, Daniel-Constantin Mierla escribió: >> Hello, >> >> On 9/20/11 10:13 AM, Alejandro Mingo wrote: >>> Hi, >>> >>> Since we installed last version of Kamailio (3.1.4) >> >> the last version is now 3.1.5, but this is not really the most >> important aspect. However, it is recommended to upgrade, there were >> some fixed to registration handling when register requests have >> multiple contacts. >>> we have been experiencing a big problem with REGISTER >>> retransmissions. When the server receives a retransmitted REGISTER >>> it removes the binding and the UAC remains unregistered until next >>> refreshing period. I include a PCAP capture that shows this behaviour. >>> >>> I am not 100% sure that this problem never happened with older >>> versions. >>> >>> Has anybody experienced the same problem? >> I will try to reproduce and investigate myself. Anyhow, meanwhile >> maybe you can provide more hints faster. Do you have error messages >> in the syslog? Can you run with debug=4 in config file and send the >> output in syslog for such a case? -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 3

Kamailio 3.1.5 RPMS
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, just to let you know that the RPMs of Kamailio 3.1.5 were generated for CentOS, Fedora, OpenSUSE and RedHat. This time CentOS RPMs are packaged for both versions 5 and 6. Links are available on our wiki at: * http://www.kamailio.org/dokuwiki/doku.php/packages:rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Kamailio and FreeSWITCH realtime integration, tutorial?
by Henrik Aagaard Sørensen 22 Sep '11

22 Sep '11

Does anyone know if there somewhere exists a tutorial about Kamailio and FreeSWITCH realtime integration? I have Googled a lot and found: http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index On the same site there is a tutorial for Kamailio and Asterisk realtime integration, but not for FreeSWITCH.

4 4

Re: [SR-Users] Dialog module showing calls that have already been terminated
by Timo Reimann 21 Sep '11

21 Sep '11

Hey Phillip, On 20.09.2011 13:48, Phillman25 Kyriacou wrote: > Thanks for your email. > > Yes dlg_manage(); has to now be called on INVITE and BYE/CANCEL messages. > Where would i have to call loose_route()? Only on INVITE? On *all* in-dialog requests, i.e., all requests which contain a To tag. This may include Re-INVITEs too (but not initial INVITEs). > My configuration did not change between 3.1.2 and 3.1.5. > > Call flow example: > ============== > > Cisco PGW ===> Kamailio 3.1.5 ===> VOIP PROVIDER or ASTERISK PABX > > > The below is my configuration. Let's take a look at it: > #!KAMAILIO > # [...] > if(is_method("BYE|CANCEL")) > > { > > dlg_manage(); > > > > # per request initial checks > route(REQINIT); > > # NAT detection > route(NAT); > > # handle requests within SIP dialogs > route(WITHINDLG); [...] > # Handle requests within SIP dialogs > route[WITHINDLG] { > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { [...] So route[WITHINDLG] contains the logic to track in-dialog requests. However, you seem to call that route only from BYE and CANCEL requests, missing out ACKs, which is likely the reason why things go wrong. So make sure you run all in-dialog requests through that route, and you should be fine (hopefully). Cheers, --Timo

2 8

Kamailio Development Training, San Francisco Area, October 2011
by Daniel-Constantin Mierla 21 Sep '11

21 Sep '11

Hello, since I am attending GSoC'11 Mentors Summit and come to Bay Area again, it is time to bring the developer training in USA, therefore I am planning to organize a two-day seminar focused on developing Kamailio extensions in C. Among topics to be approached: - internal architecture - SIP parser - memory manager - locking manager - database API - config file language interpreter - RPC interface - pseudo-variables and transformations framework - module interface – write your own extensions in C as modules - inter-module APIs - tm, presence, a.s.o. - documentation docbook format This kind of training follows the pattern from the past, trying to keep a low attendee fee, targeting just to cover the expenses with the organization. I am discussing with several people for a good location at a convenient price, however, I am open for new alternatives, including companies/institutes that want to host the event at no cost in exchange of sending few attendees free of charge. Contact me if you can help here. Training is planned for October 24-25, right now the estimated fee per participant is 350USD (it will not exceed 400USD). The policy to accept people is first come, first served. You can register at this time to reserve your seat(s), emailing to me or to registration(a)kamailio.org. Number of seats is usually 10 to 20, last one organized in February in Barcelona filled up quickly. Once there will be enough interested people and a location selected, I will have more precise details regarding the participation to this seminar and I will send updates. Looking forward to meeting many of you and get new US developers on board of the project in the near future. Cheers, Daniel -- http://www.kamailio.org/w/daniel-constantin-mierla/

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how to remove media description from sdp?
by Juha Heinanen 21 Sep '11

21 Sep '11

trying to hack around the possible fix_nated_sdp("1") issue that i outlined in previous message, i thought to remove video media description from sdp. in sdpops module there is sdp_with_media(type), but for some reason no sdp_remove_media(type) function. is there some other means to get rid of sdp video media description? -- juha

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