- sr-users - kamailio.org

[Kamailio-Users] routing question
by Andrew Gallo 06 Nov '09

06 Nov '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings: Hope you don't mind a question from someone just getting his feet wet :) First off, if I want Kamailio to act as a SIP router among, say, a group of Asterisk PBXs (meaning, Kamailio will have no user registrations, but just pass SIP messages to and from the Asterisks), would I do this in the least cost routing table or in carrier routes table, or via another method? (assume for the moment that all the PBXs are under my control, these aren't connections outside my organization, no connections to the PSTN). I figured it would be with lcr, so I tried compiling 3.0.0RC1 with that module removed from the exclude list. I received the following error on 'make install' make[1]: Entering directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' make[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' config.mak included make[1]: Entering directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m32 - -minline-all-stringops -falign-loops -ftree-vectorize - -fno-strict-overflow -mtune=athlon64 -Wall -DNAME='"kamailio"' - -DVERSION='"3.0.0-rc1"' -DARCH='"i386"' -DOS='linux_' - -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux - -DSER_VER=3000000 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC - -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP - -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER - -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS - -DSTATISTICS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 - -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN - -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL - -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL - -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT - -I/usr/local/include -I/usr/local/include -I/opt/include - -I/usr/sfw/include -DOPENSER_MOD_INTERFACE -DMOD_NAME='"lcr"' -c hash.c - -o hash.o In file included from hash.c:26: lcr_mod.h:34:18: error: pcre.h: No such file or directory In file included from hash.c:26: lcr_mod.h:51: error: expected specifier-qualifier-list before âpcreâ hash.c:34: error: expected declaration specifiers or â...â before âpcreâ hash.c: In function âlcr_hash_table_insertâ: hash.c:56: error: âstruct lcr_infoâ has no member named âfrom_uri_reâ hash.c:56: error: âfrom_uri_reâ undeclared (first use in this function) hash.c:56: error: (Each undeclared identifier is reported only once hash.c:56: error: for each function it appears in.) hash.c:58: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:59: error: âstruct lcr_infoâ has no member named âfirst_gwâ hash.c:60: error: âstruct lcr_infoâ has no member named âpriorityâ hash.c:66: error: âstruct lcr_infoâ has no member named ânextâ hash.c: In function âlcr_hash_table_contents_freeâ: hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:106: error: âstruct lcr_infoâ has no member named âgrp_idâ hash.c:107: error: âstruct lcr_infoâ has no member named âfrom_uri_reâ hash.c:108: error: âstruct lcr_infoâ has no member named âfrom_uri_reâ hash.c:110: error: âstruct lcr_infoâ has no member named ânextâ make[1]: *** [hash.o] Error 1 make[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/lcr' make: *** [install-modules] Error 1 If I leave lcr in the exclude list, I can make/make install, but if I try to add an entry into the lcr table, I get this error: # kamctl lcr addgw labCM 172.16.59.6 5060 1 2 1 ERROR 1054 (42S22) at line 1: Unknown column 'ping' in 'field list' ERROR: lcr - SQL Error System info: Kamailio 3.0.0 RC1 from GIT debian 5.0.3 (2.6.26-2-686) gcc -v Using built-in specs. Target: i486-linux-gnu Configured with: ../src/configure -v --with-pkgversion='Debian 4.3.2-1.1' --with-bugurl=file:///usr/share/doc/gcc-4.3/README.Bugs - --enable-languages=c,c++,fortran,objc,obj-c++ --prefix=/usr - --enable-shared --with-system-zlib --libexecdir=/usr/lib - --without-included-gettext --enable-threads=posix --enable-nls - --with-gxx-include-dir=/usr/include/c++/4.3 --program-suffix=-4.3 - --enable-clocale=gnu --enable-libstdcxx-debug --enable-objc-gc - --enable-mpfr --enable-targets=all --enable-cld - --enable-checking=release --build=i486-linux-gnu --host=i486-linux-gnu - --target=i486-linux-gnu Thread model: posix gcc version 4.3.2 (Debian 4.3.2-1.1) Thank you! -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAkrx/CMACgkQQr/gMVyFYySWQgCeJoW2VLWRI85bHdC9ThwnEwLc pWQAoJZVtBEzdRI8+3xBoLnlWJxNXbj/ =Wwjk -----END PGP SIGNATURE-----

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[Kamailio-Users] Dispatcher module question
by DJB 05 Nov '09

05 Nov '09

Is there a way that I can pass variable into ds_select_dst(set, alg) on the set parameter so I can query from database table for setid; for instance, ds_select_dst("avp(customer_setid)", "4"). I've tried, but it did not work. Please advise. Thank you.

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difference between memlog and memdbg
by Klaus Darilion 05 Nov '09

05 Nov '09

Hi! What's the difference betwenn memlog and memdbg? thanks klaus

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[Kamailio-Users] uac_replace_from check
by jh＠tutpro.com 05 Nov '09

05 Nov '09

if i have understood correctly, if uac_replace_from() is called in "auto" mode, encoded from uri is added into an extra param of r-r header when the request is forwarded towards uas. does the module check that reply to the request from uas really includes the added parameter in its r-r header or is correct operation of in-dialog requests at the mercy of the uas? -- juha

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Re: [Kamailio-Users] LCR problem with "id" column and match priorities
by jh＠tutpro.com 05 Nov '09

05 Nov '09

it turned out that the problems ricardo reported on gw selection in lcr module were due to the fact that he has the same gw (same in terms of ip address) in more than one gw group and in each group the gw has different trip/tag attributes. load_gws() function tries the same gateway only once, but it is not currently well defined from which group gw's attributes are taken when request is sent to the gateway. currently the attributes are taken from the gw group, which happens to be the first in in-memory gw table. one could think that this is a feature, i.e., if the same gw is in many gw groups, it must have the same attributes in every group. the other alternative is that this is a bug, i.e., same gw's attributes can differ from group to group and the ones that are used must be the ones of the chosen group. i have a fix written for sr_3.0 lcr module, but i'm hesitating to introduce the fix also to version 1.5, because the fix has not been tested well yet. any opinions on what should be done? -- juha

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Re: [Kamailio-Users] syntax error
by Daniel-Constantin Mierla 05 Nov '09

05 Nov '09

don't forget -- always cc the mailing list. On 05.11.2009 5:39 Uhr, GEMINORIUM crux wrote: > Daniel, > i am transferring tls protocol to udp protocol, > sorry, i don't mention it. > > client <---tls----> kamilio <----udp---> asterisk > > thanks for your reply. > > > On Wed, Nov 4, 2009 at 7:38 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 04.11.2009 14:34 Uhr, GEMINORIUM crux wrote: > > hi, > i am new to kamailio, > > getting syntax error: > $var(test) = "udp:"+$rd+":"+$rp; > t_relay_to_udp($var(test)); > > t_relay_to_udp($rd,$rp); > > working with string. > > is there any reason ? how to resolve it ? > i have to use database so i need to use variable. > > any suggestion is appreciated. > > you have to do: > > $du = "sip:"+$rd+":"+$rp; > t_relay(); > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > * http://www.asipto.com/ > > -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] vim syntax file for 3.0
by Daniel-Constantin Mierla 04 Nov '09

04 Nov '09

Hello, I updated the vim syntax file for Kamailio 3.0 and SR3.0/devel. I attached as well a script to auto-detect the type of .cfg file based on matching compatibility string (i.e., #!XYZ), loadmodule or main route line. To install, copy the files in your home dir - see the hints inside each file to know which one to place where: .vim/syntax/ser.vim .vim/ftdetect/ser.vim If you don't install the file type detect script, then you can use ":setf ser" once you opened the config file. Enjoy, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ " Copy this file to $HOME/.vim/ftdetect/ser.vim func! s:cfgType() let max = line("$") > 200 ? 200 : line("$") for n in range(1, max) if getline(n) =~ '^\s*#!$KAMAILIO\|OPENSER\|SER\|ALL\|MAXCOMPAT$' set filetype=ser return elseif getline(n) =~ '^\s*modparam\s*(\s*"[^"]\+"' set filetype=ser return elseif getline(n) =~ '^\s*route\s*{\s*' set filetype=ser return endif endfor setf cfg endfunc au BufNewFile,BufRead *.cfg call s:cfgType() " -*- vim -*- " FILE: ser.vim " LAST MODIFICATION: 2009-05-28 18:30 " (C) Copyright 2008 Stanisław Pitucha <viraptor(a)gmail.com> " (C) Copyright 2009 Daniel-Constantin Mierla <miconda(a)gmail.com> " Version: 1.01 " USAGE: " " Save this file to $VIMFILES/syntax/ser.vim. Either add a detection " script to filetypes.vim, or set filetype manually to "ser" when " editing SIP Router configuration file with 'setf ser'. " " List of keyword and core functions taken from latest dev version of " SIP Router. Module functions not included. " " Tested only on vim 7.1 " " Example: "setf ser" " " REQUIREMENTS: " vim (>= 7) if exists("b:current_syntax") finish endif syn match serConfigParamLine '^[^=]\+=.*$' contains=serCoreParameter,serString,serConfigConstant,serSpecial,serNumber,serCppComment,serHashComment syn region serConfigModparam start='^\s*modparam\s*(' end=')' contains=serString,serNumber syn match serConfigModule '^\s*loadmodule\s*"[^"]\+"' contains=serString syn keyword serTodo TODO FIXME XXX contained syn match serOperator '!\|&&\|||\|=[~=]\?\|>\|<\|+\|-\|/\|\*\||\|&\|^\|\~\|defined\|eq\|ieq\|ne\|ine\|mod' display contained syn region serCppComment start='/\*' end='\*/' contains=serTodo syn match serHashDefine '#!define\s\|#!ifdef\s\|#!ifndef\s\|#!endif\|#!else\|#!KAMAILIO\|#!OPENSER\|#!SER\|#!MAXCOMPAT\|#!ALL' " syn match serHashDefine '^\s*#!.+$' syn match serHashComment '#[^!].*$\|#$' contains=serTodo syn match serStringEscape '\\.' contained syn match serNumber '[0-9]\+' contained syn region serString matchgroup=Normal start='"' skip='\\"' end='"' contained contains=serVariable,serStringEscape syn match serVariable "$[a-zA-Z_][a-zA-Z0-9_]*$([^)]\+)$\?" contained syn match serIdentifier '[a-zA-Z_][a-zA-Z0-9_]*' contained syn keyword serStatement route if else switch case default break exit return drop while include_file contained syn keyword serSpecial yes no on off true false enabled disabled contained syn keyword serCoreKeyword af dst_ip dst_port from_uri method msg:len proto status snd_af snd_ip snd_port snd_proto src_ip src_port to_af to_ip to_port to_proto to_uri uri uri:host uri:port contained syn keyword serCoreValue udp UDP tcp TCP tls TLS sctp SCTP inet INET inet6 INET6 sslv23 SSLv23 SSLV23 sslv2 SSLv2 SSLV2 sslv3 SSLv3 SSLV3 tlsv1 TLSv1 TLSV1 max_len myself contained syn keyword serCoreFunction forward forward_tcp forward_udp forward_tls forward_sctp send send_tcp log error exec force_rport add_rport force_tcp_alias add_tcp_alias udp_mtu udp_mtu_try_proto setflag resetflag isflagset flags bool setavpflag resetavpflag isavpflagset avpflags rewritehost sethost seth rewritehostport sethostport sethp rewritehostporttrans sethostporttrans sethpt rewriteuser setuser setu rewriteuserpass setuserpass setup rewriteport setport setp rewriteuri seturi revert_uri prefix strip strip_tail userphone append_branch set_advertised_address set_advertised_port force_send_socket contained syn keyword serCoreParameter debug fork log_stderror log_facility listen alias auto_aliases dns rev_dns dns_try_ipv6 dns_try_naptr dns_srv_lb dns_srv_loadbalancing dns_udp_pref dns_udp_preference dns_tcp_pref dns_tcp_preference dns_tls_pref dns_tls_preference dns_sctp_pref dns_sctp_preference dns_retr_time dns_retr_no dns_servers_no dns_use_search_list dns_search_full_match dns_cache_init use_dns_cache use_dns_failover dns_cache_flags dns_cache_negative_ttl dns_cache_min_ttl dns_cache_max_ttl dns_cache_mem dns_cache_gc_interval dns_cache_del_nonexp dns_cache_delete_nonexpired dst_blacklist_init use_dst_blacklist dst_blacklist_mem dst_blacklist_expire dst_blacklist_ttl dst_blacklist_gc_interval port statistics maxbuffer children check_via phone2tel syn_branch memlog mem_log memdbg mem_dbg sip_warning server_signature reply_to_via user uid group gid chroot workdir wdir mhomed disable_tcp tcp_children tcp_accept_aliases tcp_send_timeout tcp_connect_timeout tcp_connection_lifetime tcp_poll_method tcp_max_connections tcp_no_connect tcp_source_ipv4 tcp_source_ipv6 tcp_fd_cache tcp_buf_write tcp_async tcp_conn_wq_max tcp_wq_max tcp_rd_buf_size tcp_wq_blk_size tcp_defer_accept tcp_delayed_ack tcp_syncnt tcp_linger2 tcp_keepalive tcp_keepidle tcp_keepintvl tcp_keepcnt tcp_crlf_ping disable_tls tls_disable enable_tls tls_enable tlslog tls_log tls_port_no tls_method tls_verify tls_require_certificate tls_certificate tls_private_key tls_ca_list tls_handshake_timeout tls_send_timeout disable_sctp enable_sctp sctp_children sctp_socket_rcvbuf sctp_socket_receive_buffer sctp_socket_sndbuf sctp_socket_send_buffer sctp_autoclose sctp_send_ttl sctp_send_retries advertised_address advertised_port disable_core_dump open_files_limit shm_force_alloc mlock_pages real_time rt_prio rt_policy rt_timer1_prio rt_fast_timer_prio rt_ftimer_prio rt_timer1_policy rt_ftimer_policy rt_timer2_prio rt_stimer_prio rt_timer2_policy rt_stimer_policy mcast_loopback mcast_ttl tos pmtu_discovery exit_timeout ser_kill_timeout max_while_loops stun_refresh_interval stun_allow_stun stun_allow_fp server_id description descr desc loadpath mpath contained syn region serBlock start='{' end='}' contained contains=serBlock,@serCodeElements syn match serRouteBlock '$failure_\|onreply_\|branch_\|event_\|onsend_$\?route$\s*\[[^\]]\+\]$\?' contained contains=serNumber,serString,serIdentifier syn region serRrouteBlockFold matchgroup=serRouteBlock start="$failure_\|onreply_\|branch_\|event_\|onsend_$\?route$\s*\[[^\]]\+\]$\?\s*\n\?{" matchgroup=NONE end="}" contains=serBlock,@serCodeElements syn cluster serCodeElements contains=serHashDefine,serCppComment,serHashComment,serNumber,serString,serVariable,serOperator,serStatement,serKeyword,serCoreKeyword,serCoreValue,serCoreFunction,serIdentifier hi def link serCppComment Comment hi def link serHashComment Comment hi def link serHashDefine Special hi def link serTodo Todo hi def link serConfigModparam Function hi def link serConfigModule Keyword hi def link serKeyword Keyword hi def link serCoreKeyword Special hi def link serCoreValue Special hi def link serCoreFunction Function hi def link serRouteBlock Type hi def link serRrouteBlockFold Type hi def link serIdentifier Identifier hi def link serSpecial Special hi def link serCoreParameter Keyword hi def link serOperator Operator hi def link serStatement Conditional hi def link serNumber Number hi def link serVariable Identifier hi def link serString String hi def link serStringEscape Special let b:current_syntax = "ser"

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[Kamailio-Users] Social Networking Event
by Daniel-Constantin Mierla 04 Nov '09

04 Nov '09

Hello, several folks working with Kamailio, SIP Router, SER, OpenIMSCore, SEMS and Asterisk are in Berlin next week, so we think of having a dinner (or beer) meeting Thursday, 19:00, Nov 12, 2009. If happens that you are around and want to join, please send me an email to make sure you get a seat. As usual for this kind of event, free participation with everyone paying for himself/herself. Just to have an idea about how it could be, see some photos from previous similar events: http://www.asipto.com/gallery/v/social_meeting_barcelona2008/ http://www.asipto.com/gallery/v/FOSDEM2009/ From private discussions so far, we should be already about 10 people, confirmed including myself, Olle E. Johansson (Asterisk) and Elena-Ramona Modroiu. There is a lot to celebrate, from integration work within SIP Router Project to upcoming Kamailio 3.0 (now RC1). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] [Fwd: Re: uac_replace_from again]
by marius zbihlei 04 Nov '09

04 Nov '09

Forgot to forward to list

1 0

[Kamailio-Users] syntax error
by GEMINORIUM crux 04 Nov '09

04 Nov '09

hi, i am new to kamailio, getting syntax error: $var(test) = "udp:"+$rd+":"+$rp; t_relay_to_udp($var(test)); t_relay_to_udp($rd,$rp); working with string. is there any reason ? how to resolve it ? i have to use database so i need to use variable. any suggestion is appreciated. thanks

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[Kamailio-Users] Kamailio 1.5.0 and NAT
by Путято Денис 04 Nov '09

04 Nov '09

Hello everybody I have such problem with kamailio and client which behind NAT This is REGISTER message which came from client. This message came from public IP (for example, 1.1.1.1) and source port 44197. (This information I got from sip_trace) REGISTER sip:sip.ptl.ru:5060;transport=UDP SIP/2.0 From: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=a82eda3-13c4-47798948-1a1296-5259a46 To: <sip:xxxxxxx@sip.ptl.ru:5060> Call-ID: a82eda3-13c4-47798948-1a1296-9641c88(a)sip.ptl.ru CSeq: 446229198 REGISTER Via: SIP/2.0/UDP 10.130.237.163:5060;branch=z9hG4bK-47799e27-6b9841-84448db Max-Forwards: 70 Supported: timer Contact: <sip:xxxxxxx@10.130.237.163:5060;transport=UDP> Expires: 60 Content-Length: 0 This is Unauthorized message was sent by kamailio to 1.1.1.1 and: destination port not 44197 (as I expected) but 5060!!! SIP/2.0 401 Unauthorized From: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=a82eda3-13c4-47798948-1a1296-5259a46 To: <sip:xxxxxxx@sip.ptl.ru:5060>;tag=fdc454a3cdd8f9e5718382b74f33aeed.3b81 Call-ID: a82eda3-13c4-47798948-1a1296-9641c88(a)sip.ptl.ru CSeq: 446229198 REGISTER Via: SIP/2.0/UDP 10.130.237.163:5060;branch=z9hG4bK-47799e27-6b9841-84448db;received=1.1.1.1 WWW-Authenticate: Digest realm="domain name", nonce="4af0129b000000047c1020258f9f0efc4de1b76663dd2ab4" Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 And as you understand 401 reply, which was sent to 1.1.1.1:5060 can't reach client's equipment. A piece of kamailio.cfg relating to REGISTER processing: route(0) { :. if (!is_method("INVITE")) route(8); :. if (is_method("REGISTER")) { route(11); exit; } :.. } route(8) { if (nat_uac_test("5")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } route(11) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if (isflagset(5)) setbflag(6); if (!save("location")) { sl_reply_error(); exit; } } Hearty thanks for any help.

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[Kamailio-Users] 404 sparks next gw and 503
by Anders 03 Nov '09

03 Nov '09

Hi, This list becomes a mix of high-level tech and low-level beginner questions... - and I'm in the latter group. Making two lists might be over-kill, though. Here's my issue: My call termination provider is rightfully returning a 404 for a call to a number that doesn't exist. This currently seems to spark Kamailio to look for the next gw, but in this case there is no next gw set up in the lcr/gw - so it returns a 503 to the caller. If I'm confident that my provider is right that there is no such number, and 404 is the correct answer, wouldn't it be better to return a 404 to the caller rather than a 503? With 503 stats also get screwed up, because my client can just see a bunch of 503s, which is not a very detailed picture of the real situation. If I should return a 404, how would I go about this? What would I do in cfg to ensure that a 404 return does not spark it to look for a next gw, but sends 404 back to the caller? Alternatively, I could try to figure out how to get the return from my provider saved to the acc table on the INVITE record, and I'd have something more detailed to run stats on. Does that make sense? Thanks!

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[Kamailio-Users] OpenSER/Kamailio behind NAT
by systeo 02 Nov '09

02 Nov '09

Hi, I successfully installed Kamailio on the Amazon computing cloud (EWS), and it runs fine. However, I can't get to it (register) from any UACs from the public internet. In amazon, all the instances (i.e. "virtual machines) are in a "private" IP, kindda like the private IP you get in a LAN. However, Amazoin gives you a "public IP" that can be mapped to any machine instance in your account, i.e. the machine that kamailio is running. However, when i use this public IP in the configuration of the UACs, it cannot get to it. How can I get kamailio to listen on this public IP? Secondly, and related to the first, when I run kamailio in my LAN (behind a router), say on 192.168.1.113, and all my UACs are in the same LAN, say 192.168.1.112, 192.168.1.114, etc, I can register ok and make calls just fine. However, when I try to come in from the public internet, no UACs can register to my Kamailio server (running on 192.168.1.113), even though I am forwarding port 5060 from my router to 192.168.1.113 (where kamailio is running). I think this problem is related to the first one, in that I cannot get Openser (kamailio) to listen on a public IP, when it is behing a "LAN". Can anyone help? Lastly, the same issue with rtpproxy also behind the LAN, but it needs a pulic IP, right? Please help...anyonw. Hassen BEN SLIMA Analyst developer TEL : 00216 74 863 507 FAX : 00216 74 863 506 E-MAIL : <mailto:tarak@systeo.biz> hassen(a)systeo.biz Web : <http://www.systeo.biz> www.systeo.biz Adresse : Route de Tunis km 10, Technopôle de Sfax - 3021 - Sakiet EZZIT - Sfax - Tunisie <http://www.systeo.biz/> Sans titre-4

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[Kamailio-Users] force_rtp_proxy() usage with IP that is not natively homed
by Alex Balashov 02 Nov '09

02 Nov '09

Greetings, Please excuse the apparent silliness of this request, but I am curious if there is a combination of flags passed to force_rtp_proxy() and/or rtpproxy invocation options that can allow force_rtp_proxy() to substitute an SDP endpoint IP that is not a physical device/interface on the host on which rtpproxy is running? Let me explain further: 1. I have a host with a single network interface that has private IP 192.168.1.254. 2. I am receiving an inbound call from outside the network, coming in on a public IP address 4.4.4.4 that is a 1-to-1 DNAT/SNAT to 192.168.1.254. 3. The call is forwarded to a PBX on the same LAN as the native interface, e.g. 192.168.1.250. 4. I would like to set up SDP in such a way that 192.168.1.250 sendsmedia to the 192.168.1.254 interface while the outside host initiating the call sends its media to 4.4.4.4. Under normal circumstances in which both the public and private interface were physically present on the machine, this would call for the use of "bridging" mode (rtpproxy invoked with -l 4.4.4.4/192.168.1.254 and use of "ie" and "ei" flags with force_rtp_proxy() - we have previously established that rtpproxy_offer/answer do not actually seem to work as advertised). But is it possible to "bridge" to an IP that is not routed to the box? I expected that force_rtp_proxy()'s second parameter - ip_address - would do this. However, this does not appear to let me override the SDP endpoint with any IP address I want; it seems that I am still limited to choosing from those IPs physically available on the rtpproxy host. I would guess that this is because ip_address is passed as a parameter through rtpproxy's control API socket and is validated by rtpproxy itself, and if it is not a local existing address, is overridden with one that is. Is there a clever combination of flags that can be used to accomplish this, or is it a lost cause? Thanks! -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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[Serusers] error: SER/FIFO not accessible: 2
by Brian Artigas 02 Nov '09

02 Nov '09

Hello, I'm trying to add a user to the database and getting the following error back: error: SER/FIFO not accessible: 2 Here is an excerpt from my ser.cfg file: debug=3 fork=no log_stderror=yes listen=192.168.1.30 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo_mode=0777 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) Anyone have any ideas? Thanks, Brian

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[Kamailio-Users] Broken Debian packaging in Kamailio 3.0.0-RC
by Kim Philby 31 Oct '09

31 Oct '09

Hi! I've tried to make Debian packages for latest Kamailio, but some issues arised. First, packages are named ser-stuff, shouldn't they be called kamailio-stuff even though Kamailio now uses SIP-Router core? Also, Debian package making seems to be broken: # install the *.sql files for ser_mysql.sh script cp -f scripts/mysql/*.sql /home/kmphlb/src/kamailio-3.0.0/kamailio/debian/ser/usr/lib/ser/ cp: target «/home/kmphlb/src/kamailio-3.0.0/kamailio/debian/ser/usr/lib/ser/» is not a directory make[1]: *** [install] Error 1 Directories are named 'ser'. Shouldn't it be just 'kamailio'? Regards, -- Kim Philby

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[Kamailio-Users] Kamailio 3.0.0 RC1
by Daniel-Constantin Mierla 30 Oct '09

30 Oct '09

Hello, I created a tarball with sources of Kamailio 3.0.0 RC1 in order to easy the usage and testing: http://www.kamailio.org/pub/kamailio/3.0.0-rc1/src/kamailio-3.0.0-rc1_src.t… There is also now a tutorial to guide the installation of kamailio 3.0 branch from GIT repository: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git Two wiki pages were created to collect what is new in Kamailio 3.0 and how to migrate to this version: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x http://www.kamailio.org/dokuwiki/doku.php/install:1.5.x-to-3.0.0 Both documents need a lot of care since many new features are missing or are not detailed, and definitely update guidelines do not cover all changes. Help is very much appreciated there. As we keep hunting and fixing issues, improve documentation, I hope to get to full release Kamailio 3.0.0 very soon. Enjoy RC1 meanwhile and report any issue you find to sr-dev(a)lists.sip-router.org. Thanks, Daniel

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[Kamailio-Users] Route header problem
by Alex Rendour 30 Oct '09

30 Oct '09

Hello, We have a problem with route on header. I receive an ack from X.X.X.X (asterisk) on Y.Y.Y.Y (kamailio 1.5.3). Y.Y.Y.Y relay ACK to Z.Z.Z.Z Y.Y.Y.Y this route from X.X.X.X: <sip:Y.Y.Y.Y;lr=on;ftag=as7e58db45>,<sip:11dc0d7918b56374@Z.Z.Z.Z;lr> And forward as the same to Z.Z.Z.Z. The route header must be : <sip:11dc0d7918b56374@Z.Z.Z.Z;lr> I think my kamailio must delete the first part of the route header, like this explaination: http://www.tech-invite.com/Ti-sip-dialog.html#inv-ack Could you help me? Regards, -- alex

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[Kamailio-Users] kam 1.5 syslog on solaris 10
by Robert McGilvray 29 Oct '09

29 Oct '09

I'm having some issues getting kam to log correctly to syslog after it daemonizes. I get all the initialization entries in the proper logfile but nothing after that - I have xlog statements throughout the config which only show up in syslog if fork=no. I've tested my syslog config with the system logger which works correctly executed as the kamailio user. If I run syslog in debug I don't see it receiving anything from kam after initialization. Any ideas? log_stderror=no log_facility=LOG_LOCAL1 debug=7 fork=yes children=4 /etc/syslog.conf (with tabs) local1.debug /var/log/globeop/kamailio.log This is the correct format of syslog in solaris. It only uses wildcard (*) in facility not levels. Specifying debug will include everything at higher levels. From the man page: For a given facility and level, syslogd matches all messages for that level and all higher levels. For example, an entry that specifies a level of crit also logs messages at the alert and emerg levels. This config is working correctly on sol 10 with openser 1.3. ykt1sipprd2# tail -f /var/log/globeop/kamailio.log Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 951507 local1.debug] DBG:core:grep_aliases: comparing host [2:loghost:5060] with us [1:loghost:5060] Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 951507 local1.debug] DBG:core:grep_aliases: comparing host [2:loghost:5060] with us [1:ykt1sipprd2:5060] Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 793175 local1.debug] DBG:core:grep_aliases: no match for: [2:loghost:5060] Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 634971 local1.debug] DBG:core:socket2str: <tcp:172.20.151.148:5060> Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 797031 local1.debug] DBG:core:fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 596959 local1.debug] DBG:core:fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16560 Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 362601 local1.debug] DBG:core:fm_malloc_init: params (fc400000, 33554432), start=fc400000 Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 551434 local1.debug] DBG:core:shm_mem_init_mallocs: success Oct 27 09:50:46 ykt1sipprd2 kamailio[3813]: [ID 123545 local1.info] INFO:core:init_tcp: using /dev/poll as the TCP io watch method (auto detected) Oct 27 09:50:46 ykt1sipprd2 kamailio[3815]: [ID 721503 local1.warning] WARNING:core:daemonize: pid file contains old pid, replacing pid -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] How to take MAC address
by Chandrakant Solanki 29 Oct '09

29 Oct '09

Hi I want to know the MAC address of registered client on kamailio... suppose i have 3 Client registered with 192.168.1.2, 192.168.1.3 and 192.168.1.4 from these 3 Client, i am registered successfully with 101, 102, 103(a)test.com But for additional authentication ... i also need MAC address for this Client. Is there any module, method, function or variable to take MAC address or is there any alternative for the above case. Waiting for favorable reply... -- Regards, Chandrakant Solanki

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[Kamailio-Users] Pass calls from/to another realm to gateway
by Alexander 29 Oct '09

29 Oct '09

I've encountered one problem I can not solve :( The situation is following: we've got Kamailio working together with our own SIP platform. Our platform is about various call processing business logic and billing. All calls from SIP users should pass through our platform. For now we successfully can make and receive calls inside our domain - NAT is handled fine in most cases, instant messages are handled and so on. It looks like that: - Kamailio receives a call from caller - Call is redirected to our platform - Platform redirects call back to Kamailio and it looks for callee. I've tried to implement that logic. It seems to work in some cases, but I encounter one problem. Sometimes Kamailio can not forward a call to the platform via rewritehostport(). It tries to forward SIP request, but nothing happens - only retransmission handler is called. I mean, after call to t_relay(), I see this: Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... Oct 27 12:12:35 [16751] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0xb618de58, INVITE sip:200213@62.117.120.101:5061 SIP/2.0... SIP requests seem to be correct, and configuration file works ok from time to time. The problem I've described happens when Kamailio receives a call from another domain (for example, from sipbroker). Inside our domain everything works fine and INVITE passes to our platform correctly. I've attached configuration file and debug output. Files were written for and by Opensips, but with Kamailio I encounter the same problem. Is this a problem with configuration file or something else?

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[Kamailio-Users] roadmap to kamailio 3.0.0
by Daniel-Constantin Mierla 29 Oct '09

29 Oct '09

Hello, I am thinking of planning to release Kamailio 3.0.0 next Wednesday or day after (October 28 or 29). With some delays, testing is going fine in my side, still some bits to be done, here is what I collected: - taking in consideration $du updates in branch route - import fixes from sr_3.0 branch - complete core statistics and handling of drop reply I believe is feasible, in the worse case we can package it at respective date and allow a bit more time for testing, but we unfreeze code and can go for new development. I encourage people to start writing migration documents. Let's see other opinions as well. Cheers, Daniel

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Re: [Kamailio-Users] How to take MAC address
by Zahid Mehmood 28 Oct '09

28 Oct '09

I know that the Polycom phones can be configured to add the MAC address in one of the headers (probably the user agent hdr). -- Zahid ----- Original Message ----- From: users-bounces(a)lists.kamailio.org <users-bounces(a)lists.kamailio.org> To: users(a)lists.kamailio.org <users(a)lists.kamailio.org> Sent: Wed Oct 28 02:58:05 2009 Subject: Re: [Kamailio-Users] How to take MAC address On Wednesday 28 October 2009 06:35:28 Chandrakant Solanki wrote: > Hi > > I want to know the MAC address of registered client on kamailio... > > suppose i have 3 Client registered with 192.168.1.2, 192.168.1.3 and > 192.168.1.4 > > from these 3 Client, i am registered successfully with 101, 102, > 103(a)test.com > > But for additional authentication ... i also need MAC address for this > Client. > > Is there any module, method, function or variable to take MAC address or is > there any alternative for the above case. > > > Waiting for favorable reply... You will not have luck outsite "your proxy network" ... I mean ... that if your proxy is on 192.168.0.10/24 and your clients on any other network, you will always get's the same MAC, the mac of the 192.168.0.10/24 gateway. Your only hope is that SOME SIP devices add their MAC as part of some subscribe or register petition. I don't see any security increase on this, because MAC's could be easily spoofed. -- Raúl Alexis Betancor Santana Dimensión Virtual _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

2 1

Re: [Kamailio-Users] Asterisk to Kamailio Call forwarding error
by Daniel-Constantin Mierla 28 Oct '09

28 Oct '09

On 28.10.2009 11:23 Uhr, Chandrakant Solanki wrote: > > > On Wed, Oct 28, 2009 at 3:50 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 28.10.2009 11:14 Uhr, Chandrakant Solanki wrote: > > > > On Wed, Oct 28, 2009 at 3:18 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com> > <mailto:miconda@gmail.com <mailto:miconda@gmail.com>>> wrote: > > > > On 28.10.2009 10:37 Uhr, Chandrakant Solanki wrote: > > Hi > > Call Sequence is ... > > Client A/B register on Kamailio and both are registered > successfully.. > Firewall is stopped. > > Now... Client A call to Client B.. > > Client A => Kamailio => Asterisk => Kamailio => Client B > > Above is call sequence, when call goes from Asterisk to > Kamailio ... then found below error.... > > ERROR:tm:t_forward_nonack: no branch for forwarding > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: > ERROR:tm:w_t_relay: t_forward_nonack failed > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** > OUT route[8] > > what processing are you doing in kamailio for this case? Do > you do > same failure routing to another destination if B does not > answer? > Such error occurs when r-uri is changed in failure route but no > append_branch() was executed. > > Cheers, > Daniel > > > > hi > > where could i put this thing.... > > Here is my config file for kamailio... > > http://pastebin.com/m34e07510 > > Please help me out ... > > > your config is big to troubleshoot it quick and easy. Definitely > you have call forwarding, what version of kamailio are you using? > the header of file says openser 1.2. > > I suggest you use ngrep/wireshark to watch the sip traffic and put > xlog() messages in the config to correlate and get hints about > what happesn. > > Cheers, > Daniel > > PS. Keep cc-ing to mailing list, there are people that can help > better or faster inc ase they use the sip wizard config. > > -- > Daniel-Constantin Mierla > * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin > * http://www.asipto.com/index.php/sip-router-masterclass/ > > > hi > > i m using kamailio-1.5.2-tls... > if you don't keep mailing list in the discussion thread, your messages will be ignored... Daniel

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Re: [Kamailio-Users] Asterisk to Kamailio Call forwarding error
by Daniel-Constantin Mierla 28 Oct '09

28 Oct '09

On 28.10.2009 11:14 Uhr, Chandrakant Solanki wrote: > > > On Wed, Oct 28, 2009 at 3:18 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 28.10.2009 10:37 Uhr, Chandrakant Solanki wrote: > > Hi > > Call Sequence is ... > > Client A/B register on Kamailio and both are registered > successfully.. > Firewall is stopped. > > Now... Client A call to Client B.. > > Client A => Kamailio => Asterisk => Kamailio => Client B > > Above is call sequence, when call goes from Asterisk to > Kamailio ... then found below error.... > > ERROR:tm:t_forward_nonack: no branch for forwarding > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: > ERROR:tm:w_t_relay: t_forward_nonack failed > Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** OUT route[8] > > what processing are you doing in kamailio for this case? Do you do > same failure routing to another destination if B does not answer? > Such error occurs when r-uri is changed in failure route but no > append_branch() was executed. > > Cheers, > Daniel > > > > hi > > where could i put this thing.... > > Here is my config file for kamailio... > > http://pastebin.com/m34e07510 > > Please help me out ... > your config is big to troubleshoot it quick and easy. Definitely you have call forwarding, what version of kamailio are you using? the header of file says openser 1.2. I suggest you use ngrep/wireshark to watch the sip traffic and put xlog() messages in the config to correlate and get hints about what happesn. Cheers, Daniel PS. Keep cc-ing to mailing list, there are people that can help better or faster inc ase they use the sip wizard config. -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] kamctl online
by Chandrakant Solanki 28 Oct '09

28 Oct '09

hello I would like to print user@domainname while i execute kamctl online command... Any Idea -- Regards, Chandrakant Solanki

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[Kamailio-Users] Asterisk to Kamailio Call forwarding error
by Chandrakant Solanki 28 Oct '09

28 Oct '09

Hi Call Sequence is ... Client A/B register on Kamailio and both are registered successfully.. Firewall is stopped. Now... Client A call to Client B.. Client A => Kamailio => Asterisk => Kamailio => Client B Above is call sequence, when call goes from Asterisk to Kamailio ... then found below error.... ERROR:tm:t_forward_nonack: no branch for forwarding Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ERROR:tm:w_t_relay: t_forward_nonack failed Oct 28 02:14:07 openser1 /sbin/kamailio[2915]: ***** OUT route[8] -- Regards, Chandrakant Solanki

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Re: [Kamailio-Users] hung dialogs + no BYE
by Daniel-Constantin Mierla 27 Oct '09

27 Oct '09

On 27.10.2009 21:22 Uhr, Anders wrote: > That was exactly the problem Daniel - no BYE was ever sent from the > UA, so that's what we need to fix! > please keep cc-ing the mailing list. You can fix the problem by identifying the sip devices that do not send BYE. You are doing record routing, right? In 1.5, dialog can send BYE at call timeout -- does not help much you, but can close eventual open channels in gateways. The missing BYE happens when call is between two SIP phones? Or between sip phone and pstn gateway/media server? Cheers, Daniel > Thanks!! > > On Tue, Oct 27, 2009 at 4:16 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> On 26.10.2009 16:41 Uhr, Anders wrote: >> >>> Hi, >>> >>> I have two issues, and I think they are connected. The number of >>> Active Dialogs keeps growing - as if some of them are hung. Not all of >>> them, but some of them. At the same time, I have seen that from a >>> specific customer, there is no BYE in the 'acc' table in the >>> accounting. So, my conclusion - no BYE means it's not finished means >>> it's hung... - right? >>> >>> Any ideas where to look? >>> >>> >>> >> if you don't get a BYE in acc table then maybe was not sent and that keeps >> the dialog active. You can set a max time per call -- timeout -- for each >> dialog: >> >> http://kamailio.org/docs/modules/1.5.x/dialog.html >> >> However, is good to identify why BYE is not coming, maybe is a fraud attempt >> or a broken sip device. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >> * http://www.asipto.com/index.php/sip-router-masterclass/ >> >> >> -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] newbie question
by Luis Jimenez 27 Oct '09

27 Oct '09

Hello, list. Exactly what happens with a dialog (at accounting level) when i reload kamailio with dialogs in process? Thanks in advance. Luis Jimenez

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[Kamailio-Users] hung dialogs + no BYE
by Anders 27 Oct '09

27 Oct '09

Hi, I have two issues, and I think they are connected. The number of Active Dialogs keeps growing - as if some of them are hung. Not all of them, but some of them. At the same time, I have seen that from a specific customer, there is no BYE in the 'acc' table in the accounting. So, my conclusion - no BYE means it's not finished means it's hung... - right? Any ideas where to look? Thanks!

2 1

[Kamailio-Users] LCR problem with "id" column and match priorities
by jh＠tutpro.com 24 Oct '09

24 Oct '09

ricardo, in situation where you see higher priority valued gateway group selected first, do you know if the lower priority valued group would be used if none of the gateways in the higher priority valued gateway group can serve the request? if it is used, then both exist in the hash table. the list in the hash table is sorted by prefix length and priority. -- juha

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[Kamailio-Users] acc: accounting flag for CANCEL
by Alex Balashov 24 Oct '09

24 Oct '09

If I have acc set to log CANCEL'd calls, do I need to set the accounting flag on handling CANCEL request, since it is a new transaction relative to the initial INVITE, or will it get logged automatically? Thanks, -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

2 1

[Kamailio-Users] basic SIP forwarding with Asterisk
by Jeff Brower 23 Oct '09

23 Oct '09

All- Can we use Asterisk combined with Kamailio as follows: __________ ___________ | | | | SIP ___| |___ SIP ___| Kamailio |___ SIP | | | rtpproxy | | Asterisk | | | | | | | | | RTP ___| |___ RTP ___| DSP card |___ RTP (G711) |__________| (G711) |___________| (G729, G723, GSM-AMR, EVRC) We've already implemented an rtpproxy interface to the DSP card, which has its own GbE port. Our question is whether we can we perform some type of basic SIP forwarding or "SIP pass-thru", but still invoke rtpproxy for call setup and tear-down and/or when media attributes for the call change? We're getting a lot of requests from customers who -- for whatever reasons -- need to use or continue to use Asterisk, but need to also add transcoding, ec, encryption, and other compute-intensive requirements that Asterisk doesn't support (or at least doesn't support at higher capacity or without going unstable). Thanks. -Jeff

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[Kamailio-Users] Anybody using domainpolicy module?
by Klaus Darilion 23 Oct '09

23 Oct '09

Hi! The domainpolicy draft is obsolete and the module needs adaptions to work with sip-router. Thus, we plan to remove it, as I guess nobody uses it at all. So, if you use this module, please speek up NOW! regards klaus

1 0

[Kamailio-Users] Search for text with a space in it
by CSB 22 Oct '09

22 Oct '09

OpenSER 1.3 When I try to match a phrase with a space in it I don't get a match. I'm looking for "User-Agent: My UA" None of the following seem to work: if (search("User-Agent:\sMy\sUA")) { if (search("User-Agent:[ ]My[ ]UA")) { if (search("User-Agent: My UA")) { Could anyone point out my obvious mistake? Thanks

2 1

[Kamailio-Users] Kamailio ignores Via in replies?
by Alex Hermann 22 Oct '09

22 Oct '09

Hello, while debugging a problem today I encountered a rather strange issue. The client's UA/router was messing up the Via header, with as result that all the IP's in the Via's were replaced by the public IP of the DSL modem. Now the weird part: because of the mangled Via's, the expected result is that the reply would be routed back to the IP of the client. Instead, the replies were still following the "correct" return path through a series of OpenSER/Kamailio proxies. How is this possible? According to rfc3261, a proxy must forward the reply to the next Via in the received _response_. It seems Kamailio is incorrectly keeping the Via from the request and sends the reply to the stored address of the Via from the _request_. Btw, no 'received' parameter is involved. Can anyone with inside knowledge of Kamailio share his/her view on this issue? And possibly, how to fix it. I suspect I can use check_via=1 to drop incoming replies with incorrect via, although the behaviour of check_via is undocumented (what does it do when check fails?). And why is the default value 0 (as has every example config on the web)? There's also reply_to_via, but according to docs that's only for locally generated replies. Also, its use is beyond me, as the correct value is dependent on the src_ip of the request and thus can't possibly be a global option. Alex.

2 1

[Kamailio-Users] SIP message
by Rajesh 22 Oct '09

22 Oct '09

Dear All, I am using a separate Location server.I would like to know which SIP message is used to call the location server.To be more clear, as INVITE messages are used to make calls, I would like to know the similar SIP message which is used to store the user details in the location table..Is that REGISTER message ?...Correct me if iam wrong.. -- Regards, Rajesh

2 1

[Kamailio-Users] Kamailio v1.5.3 released
by Daniel-Constantin Mierla 21 Oct '09

21 Oct '09

A new release in 1.5 series is out. Kamailio 1.5.3 is based on the latest version of branch 1.5, including many fixes in code and documentation, therefore those running 1.5.0, 1.5.1 or 1.5.2 are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.5.3/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/1.5.3/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.5.3/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Binaries and packages will be uploaded at: http://www.kamailio.org/pub/kamailio/1.5.3/ Modules' documentation: http://www.kamailio.org/docs/modules/1.5.x/ What is new in 1.5.x release series is summarized in the announcement of v1.5.0: http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes Enjoy and stay tunned -- new major release Kamailio 3.0.0 is going to be out very soon, Daniel

1 0

[Kamailio-Users] LCR problem with "id" column and match priorities.
by Ricardo Martinez 21 Oct '09

21 Oct '09

Hello List. I'm having a weird behavior with the LCR module. I have a particular prefix routed as follows (according to the LCR tables) : mysql> select * from lcr where prefix=0251; +-----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +-----+--------+----------+--------+----------+ | 89 | 1351 | | 51 | 1 | | 127 | 1351 | | 43 | 2 | +-----+--------+----------+--------+----------+ The gw's group are as follows : mysql> select * from gw where grp_id=51; +-----+----------------+--------+---------------+----------+------+----- -------+-----------+-------+------+--------+------+-------+ | id | gw_name | grp_id | ip_addr | hostname | port | uri_scheme | transport | strip | tag | weight | ping | flags | +-----+----------------+--------+---------------+----------+------+----- -------+-----------+-------+------+--------+------+-------+ | 100 | GW1.TERMN1 | 51 | XX.XXX.XXX.16 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | | 101 | GW2.TERMN1 | 51 | XX.XXX.XXX.13 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | | 102 | GW3.TERMN1 | 51 | XX.XXX.XXX.32 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | | 103 | GW4.TERMN1 | 51 | XX.XXX.XXX.35 | | 5060 | 1 | 1 | 4 | 1351 | 25 | 1 | 0 | +-----+----------------+--------+---------------+----------+------+----- -------+-----------+-------+------+--------+------+-------+ 4 rows in set (0.00 sec) mysql> mysql> select * from gw where grp_id=43; +----+------------------+--------+---------------+----------+------+---- --------+-----------+-------+------+--------+------+-------+ | id | gw_name | grp_id | ip_addr | hostname | port | uri_scheme | transport | strip | tag | weight | ping | flags | +----+------------------+--------+---------------+----------+------+---- --------+-----------+-------+------+--------+------+-------+ | 76 | GW1.TERMN222 | 43 | XX.XXX.XXX.16 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | | 77 | GW2.TERMN222 | 43 | XX.XXX.XXX.13 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | | 78 | GW3.TERMN222 | 43 | XX.XXX.XXX.32 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | | 79 | GW4.TERMN222 | 43 | XX.XXX.XXX.35 | | 5060 | 1 | 1 | 4 | 1343 | 25 | 1 | 0 | +----+------------------+--------+---------------+----------+------+---- --------+-----------+-------+------+--------+------+-------+ 4 rows in set (0.00 sec) So, when a call with prefix 1351 arrived at the proxy, the LCR module is changing the RURI for 1343 (matching the 2nd priority for that prefix). This keep happening even if change the priority value 2 for priority value 5. I even changed the priority 1 for priority 0 as follows : +-----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +-----+--------+----------+--------+----------+ | 89 | 1351 | | 51 | 0 | | 127 | 1351 | | 43 | 1 | +-----+--------+----------+--------+----------+ But the LCR module keep using the "second" priority. After doing some tests I've found two ways to use the "first" priority for the prefix. 1.- Removing the second priority 2.- Re-ordering the "id" column and using correlative values as indicated in the next table : mysql> select * from lcr where prefix=0251 order by id; +-----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +-----+--------+----------+--------+----------+ | 127 | 1351 | | 51 | 1 | | 128 | 1351 | | 43 | 2 | +-----+--------+----------+--------+----------+ So, when the prefix are with "no correlatives" values in the "id" column seems to be a problem to match the first priority. But when the values are correlative there is no problem. Is this the correct behavior? Maybe I'm missing something? What happens for example if I need to add a third priority to a certain prefix after doing several changes in the LCR table? I need to re-order every time the "id" values to make them correlatives? I really hope that someone could give me a hand here. Ricardo Martinez.-

3 11

[Serusers] question regarding sip protocol
by Sabatella, Michael 21 Oct '09

21 Oct '09

Hi, I am new to the sip protocol. I have been doing packet traces to understand how the protocol works. I am using the ser sip proxy version 0.9.6. I notice that once my phone registers with a proxy I get a 200 ok. Every 20 seconds after that the client sends a packet to the proxy. It is an Request: OPTIONS packet. My proxy responds with a Status:404 Not found. I tried to do some research on this but could not find anything useful. Does this response of a 404 matter? How can you fix this? I have read that the client does this as an application level ping to make sure the proxy is up. If I have 100s of people registering that is a lot of extra traffic. Is there a way to stop this options query or have the proxy respond properly? Thanks Michael Network Services Engineer ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

2 1

[Kamailio-Users] preparing Kamailio 1.5.3
by Daniel-Constantin Mierla 21 Oct '09

21 Oct '09

Hello, Kamailio 1.5.3 will be released today. Please commit anything you have for branch 1.5 on Sourceforge.net svn in the next two hours. Afterwards we will start preparing the release. Cheers, Daniel

1 0

[Kamailio-Users] max channel capacity
by Jeff Brower 21 Oct '09

21 Oct '09

All- On a reasonably fast server, say quad-core, what is the approx maximum number of G711 IP calls when using Kamailio and rtpproxy? What if rtpproxy runs on a second server? No echo can, no packet concealment, etc... just G711. -Jeff

3 5

Re: [Kamailio-Users] use case for settlement free peering
by Daniel-Constantin Mierla 20 Oct '09

20 Oct '09

welcome -- cc-ing mailing list is recommended all the time -- help people to get the conclusion of discussions. On 20.10.2009 20:04 Uhr, A G wrote: > Great. Thanks for the feedback! > > On Mon, Oct 19, 2009 at 3:47 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> Hello, >> >> On 19.10.2009 20:18 Uhr, A G wrote: >> >>> Greetings: >>> >>> I'm looking for advice on a project/proof of concept I'm working on. >>> I would like to create a settlement-free peering fabric for voice >>> traffic between and among some peer institutions in my area. Because >>> this is more of a side-project for cost cutting measure, I'm primarily >>> looking at open source software, though commercial product >>> recommendations would be helpful as well. >>> >>> The organizations I would like to connect have their own PBXs with >>> large blocks of numbers (whole NPA-NXXs), with no number portability >>> in or out. I imagine both at the individual PBXs and peering fabric, >>> the number routing would be static. To put another way, we would >>> manually configure which connections the block of telephone numbers is >>> reachable at. >>> >>> Here is the required ASCII art diagram :) >>> >>> >>> +-------+ >>> | PBX | >>> +-------+ >>> | >>> +-------+ >>> | SBC | >>> +-------+ >>> | >>> | >>> +---+ +---+ .--------. +---+ +---+ >>> | P | | S | / \ | S | | P | >>> | B |--| B |------ ( ???? )----------| B |--| B | >>> | X | | C | \ / | C | | X | >>> +---+ +---+ `---------' +---+ +---+ >>> | >>> | >>> +-------+ >>> | SBC | >>> +-------+ >>> | >>> +-------+ >>> | PBX | >>> +-------+ >>> >>> >>> >>> >>> For scalability reasons, a full mesh of connections between and among >>> the SBCs is not an attractive option. >>> >>> Here's what I think I need: >>> Basic SIP routing >>> TCP, TLS, and UDP support >>> >>> What would be nice to have: >>> IPv6 >>> CDR >>> >>> What is probably not needed: >>> User agent client registration, presence, IM, voice mail >>> >>> I see there are several different open source voice projects. >>> Do you think this is an appropriate use for Kamailio? >>> >>> >> Kamailio can be used in such scenario. It has a lot of features to help you >> routing calls -- see modules such as lcr, carrierroute, dialplan. >> >> If you do heavy tls/tcp, then upcoming Kamailio 3.0 will have lot of >> improvements in this areas. IPv6 is supported as well for core routing. You >> can generate CDRs using acc module and some stored procedures -- see for >> example: >> http://siremis.asipto.com/install-accounting/ >> >> Cheers, >> Daniel >> >> >>> I'm seeking comments on what you would use for this situation. >>> Are there any existing projects along these lines? >>> Is there one project that is better than another for this application? >>> >>> >>> >> -- >> Daniel-Constantin Mierla >> * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >> * http://www.asipto.com/index.php/sip-router-masterclass/ >> >> >> -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

2 1

Re: [Kamailio-Users] Problem with LCR "ping"
by Alexandr Dubovikov 20 Oct '09

20 Oct '09

Hi Ricardo, you use ping_interval 10 seconds. For 126 gateways it's extrem short . Can you set this param up to 60 seconds and check again ? Wbr, Alexandr P.S. will be nice to see the result from "ngrep -W byline -t -l port 5060", if it possible :-) > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Ricardo Martinez > Sent: Thursday, October 15, 2009 4:48 PM > To: users(a)lists.kamailio.org > Subject: [Kamailio-Users] Problem with LCR "ping" > > Hello list. > > I have a question/problem regarding the LCR module. I'm using Kamailio version 1.5.2. The LCR tables have : > > > > gw table : 126 records > > lrc tables : 120 records. > > > > I'm using the "ping" (ping = 1) feature in all my gateways, part of the cfg file is as follows : > > > > modparam("lcr","db_url","mysql://openserro:openserro@localhost/openser") > > modparam("tm", "fr_inv_timer_avp", "$avp(i:704)") > > modparam("lcr", "gw_uri_avp", "$avp(i:709)") > > modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") > > modparam("lcr", "ruri_user_avp", "$avp(i:500)") > > modparam("lcr", "flags_avp", "$avp(i:712)") > > modparam("lcr", "fetch_rows", 3000) > > modparam("lcr", "ping_interval", 10) > > modparam("lcr", "ping_from", "sip:pinger@mydomain.net") > > modparam("lcr", "ping_method", "OPTIONS") > > > > From time to time I can see 3 o 4 gateways marked as down (with the command kamctl lcr dump). I'm not manually monitoring the gateways every minute so is possible that in a particular moment the gateways could be unreachable, but when I run the command "kamctl lcr dump" the gateways are on-line, that's for sure. > > So I'm not understanding why LCR keep marking the gateways as down, even if they are answering to the OPTIONS request. Could this be a bug in the LCR module? Or maybe I'm missing something?. The silly part of all this is the gateways marked as down are always the same. Could this be related to a memory issue?? > > > > Can someone guide me here?. > > > > Thanks in advance, > > Regards, > > Ricardo Martinez.- > > > > -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info

5 8

[Serusers] SER Staring Problem
by sourav.das1＠wipro.com 20 Oct '09

20 Oct '09

Hi, I am trying to install the SER through the following steps. Download and extract the package : * Download the source package containing mysql support for SER from: http://ftp.iptel.org/pub/ser/latest/ <http://ftp.iptel.org/pub/ser/latest/> * Save the downloaded package at /usr/src folder * Extract the package * Change directory to the extracted package Make and install Make the program and the modules and then install the program in the folder /usr/local. # make && make modules # make prefix=/usr/local install After installing the SER on the Linux system I am trying to start SER through the following command. /usr/local/sbin/serctl start While executing the above command I am getting the following error. Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed Please let me know how to resolve this or something is missing. Regards, Sourav Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 1

[Kamailio-Users] kamailio 1.5.3 scheduled for next wednesday
by Henning Westerholt 20 Oct '09

20 Oct '09

Hi all, as it has been three month since the last minor release of the kamailio 1.5 branch i'd suggest that we do a 1.5.3 release soon. I want to propose next wednesday the 14.10.2009 as date for the release. So if you have any pending patches in the trunk or in your internal repositories that you want to like to see in this release, please commit them until 12:00 UTC this day. Thank you, Henning -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

4 8

[Kamailio-Users] use case for settlement free peering
by A G 19 Oct '09

19 Oct '09

Greetings: I'm looking for advice on a project/proof of concept I'm working on. I would like to create a settlement-free peering fabric for voice traffic between and among some peer institutions in my area. Because this is more of a side-project for cost cutting measure, I'm primarily looking at open source software, though commercial product recommendations would be helpful as well. The organizations I would like to connect have their own PBXs with large blocks of numbers (whole NPA-NXXs), with no number portability in or out. I imagine both at the individual PBXs and peering fabric, the number routing would be static. To put another way, we would manually configure which connections the block of telephone numbers is reachable at. Here is the required ASCII art diagram :) +-------+ | PBX | +-------+ | +-------+ | SBC | +-------+ | | +---+ +---+ .--------. +---+ +---+ | P | | S | / \ | S | | P | | B |--| B |------ ( ???? )----------| B |--| B | | X | | C | \ / | C | | X | +---+ +---+ `---------' +---+ +---+ | | +-------+ | SBC | +-------+ | +-------+ | PBX | +-------+ For scalability reasons, a full mesh of connections between and among the SBCs is not an attractive option. Here's what I think I need: Basic SIP routing TCP, TLS, and UDP support What would be nice to have: IPv6 CDR What is probably not needed: User agent client registration, presence, IM, voice mail I see there are several different open source voice projects. Do you think this is an appropriate use for Kamailio? I'm seeking comments on what you would use for this situation. Are there any existing projects along these lines? Is there one project that is better than another for this application? Thank you

2 1

Presence gateway between SIP and XMPP
by afshin afzali 18 Oct '09

18 Oct '09

Hi Guys, You stated : "*This module is a gateway for presence between SIP and XMPP. * * It translates one format into another and uses xmpp, pua and presence modules to manage the transmition of presence state information*" My first question is that does it enable exchange of presence data between SIP and XMPP in two ways ? If yes how I could use the pua_xmpp_notify and pua_xmpp_req_winfo APIs to do so ? Regards, -- Afshin

1 0

[Kamailio-Users] Odd behavior svn rev 2:5901M
by brandon＠cryy.com 16 Oct '09

16 Oct '09

Hello, I am noticing very rarely however it is happening where kamailio will route to the completely wrong sip uri. In this scenario I see it routing to one of our voicemail applications, thus placing voicemails into the wrong mailbox identifier. What is odd in this particular scenario I was able to capture is that it attempts to deliver for the same call-id to the right location and wrong location within a very very very small time frame causing our voicemail application to pick the wrong branch. What is even odder is this is a recovery from a 487 request cancelled so it is coming from failure route, appending a branch, so there should only be a single branch to voicemail. I clearly see in the logs after setting the ruri that it is correct, I have no idea where it is getting this ghost mailbox uri identifier, it is almost as some how the call got mixed up with a completely different call. Just throwing this out here to see if anyone else has experienced this. I will attach supporting evidence to this email for this scenario shortly. The last thing to mention for now is that it is definitely random and not sure-proof reproducible. Thank you for your time and input ahead of time. Look forward to hearing back from you all! Have a great weekend. Sent from my Verizon Wireless BlackBerry

1 0

[Kamailio-Users] Problem with LCR "ping"
by Ricardo Martinez 16 Oct '09

16 Oct '09

Hello list. I have a question/problem regarding the LCR module. I'm using Kamailio version 1.5.2. The LCR tables have : gw table : 126 records lrc tables : 120 records. I'm using the "ping" (ping = 1) feature in all my gateways, part of the cfg file is as follows : modparam("lcr","db_url","mysql://openserro:openserro@localhost/openser") modparam("tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "fetch_rows", 3000) modparam("lcr", "ping_interval", 10) modparam("lcr", "ping_from", "sip:pinger@mydomain.net") modparam("lcr", "ping_method", "OPTIONS") >From time to time I can see 3 o 4 gateways marked as down (with the command kamctl lcr dump). I'm not manually monitoring the gateways every minute so is possible that in a particular moment the gateways could be unreachable, but when I run the command "kamctl lcr dump" the gateways are on-line, that's for sure. So I'm not understanding why LCR keep marking the gateways as down, even if they are answering to the OPTIONS request. Could this be a bug in the LCR module? Or maybe I'm missing something?. The silly part of all this is the gateways marked as down are always the same. Could this be related to a memory issue?? Can someone guide me here?. Thanks in advance, Regards, Ricardo Martinez.-

4 5

[Kamailio-Users] dispatcher failover between two groups
by Robert McGilvray 16 Oct '09

16 Oct '09

Kam 1.5 I'm using dialplan module to return a setid for the matching regex which is passed onto dispatcher to select the proper gateways. This works fine including failover to the next gateway in the group in failure_route. What I can't quite wrap my head around is how I can setup failover between different dispatcher groups. The scenario is as follows: 2 external PSTN gateways (sip trunks) in group 1 6 internal gateways with PSTN access via PRIs. Two gateways per group for a total of three separate setids in dispatcher. I'm doing some LCR across the 6 internal gateways which are all in different cities via regex matching in dialplan. What I'd like to do is try one of the internal groups first if it matches the local NPA/NXX, and if ALL gateways in that group fail move on to the external gateway group, and try all those gateways (and vice versa in my external gateways fail for non matching numbers). I thought of two possible solutions but I'm not sure if either will work how I expect. 1) Use the priority field in dispatcher to setup a large gateway group that has all the gateways I want to use. - I tested this out in kam 1.5 and the priority field seems to be ignored when hasing on call-id. - What is the intended usage of priority? Will setting two gateways to 0 and the rest to 10 cause the load balancing functions to only select the ones with 0? It seems to be ignoring priority and using all 8 gateways 2) Once in a failure_route can I move the processing into a request_route and call ds_select_dst again with a different setid and append it as a branch? Is there a cleaner solution? Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

1 0

[Kamailio-Users] Codecs
by Anders 16 Oct '09

16 Oct '09

Hi, A very basis 101-question, which I couldn't find the answer to elsewhere: how/where do I see/set codecs allowed and disallowed in Kamailio? Thanks!

2 1

Re: [Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks
by Örn Arnarson 15 Oct '09

15 Oct '09

Hi, I have exactly the same scenario; no NAT involved. Just trying to bridge between these two interfaces. Now, to clarify better what I have done; As I understand it, force_rtp_proxy / rtpproxy_offer needs to be used to inform RTP proxy that it will have to relay the packets. So, that pretty much makes it necessary to use that call. As I understand it, you can't call both fix_nated_sdp and force_rtp_proxy, as the SDP will be mangled (two IP addresses, one after the other). However, I can't get the SDP rewritten properly without using fix_nated_sdp. No matter how I call force_rtp_proxy, I can't specify my own IP address. It works as expected when using fix_nated_sdp though. Here's how I call it: force_rtp_proxy("coe","10.252.1.8"); I've tried multiple variants of flags, c, co, o, coe, coi, i, no flags, etc etc, but no matter what I do, the public interface IP address does not change to the one supplied in the function, and remains the public IP address. So, to summarize; I can change all the headers to be correct with the fix_nated_sdp function. When using that function, RTPProxy does not relay the RTP streams. When I use force_rtp_proxy / rtpproxy_offer, RTPProxy does indeed relay the streams, but does not alter the SDP IP address information to the required specifications. Is there something that I'm missing in the usage of the force_rtp_proxy function, or is it not working as documented? Best regards, Örn 2009/10/14 Alex Balashov <abalashov(a)evaristesys.com>: > Örn Arnarson wrote: > >> Oh, I forgot one thing; >> >> I'm using fix_nated_sdp, as previously mentioned. I have a hunch that >> because I'm not using force_rtp_proxy/rtpproxy_offer it's not being >> relayed (a hunch, because I don't fully understand how rtpproxy >> works). Problem is though, I can't get the IP addresses correct by >> using rtpproxy_offer. > > On behalf of Joe (same company) I will add that there is no NAT scenario > here. Both the private and public network are directly routable from the > rtpproxy box; there is no IP address oversubscription, and no reason to > ignore ports or addresses in SDP and replace them with received parameters. > This is not a far-end NAT traversal solution. > > What is basically being asked here is how to bridge between two interfaces > (internal, external) without NAT using rtpproxy, and if anyone can provide a > canonical implementation or example. > > We've got it running in bridge mode, e.g. > > rtpproxy -F -s udp:10.x.x.x:5050 -l 10.x.x.x/89.y.y.y > > ... and basically, are just wondering how to get this to work on the > Kamailio side using rtpproxy_offer()/rtpproxy_answer(). > > The RTP proxy machine is a dedicated host; the SIP proxies are virtual. > > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 >

1 0

[Kamailio-Users] SIP-T messages
by Luis Jimenez 15 Oct '09

15 Oct '09

Hi, can some one please guide me on how can i parse the received SIP-T messages from the SDP in Kamailio. Thanks in advance Luis Jimenez

2 1

[Kamailio-Users] How to remove route header field
by Mythili D 14 Oct '09

14 Oct '09

hii..Good morning. When we are making a call from one UA to other UA, it passes through proxies. ACK message also passes through the proxies. But i need the ACK message not to go through proxies. It must directly go from UA to UA. How can i make in cfg. There must not be route header filed in SIP meessage flow. Please help me. Thanks & Regards, Mythili.K

2 1

[Kamailio-Users] AVP problem
by Ricardo Martinez 13 Oct '09

13 Oct '09

Hello list. I'm getting this error trying to start the kamailio process /usr/local/sbin/kamailio[29243]: INFO:core:init_mod: initializing module mediaproxy /usr/local/sbin/kamailio[29243]: ERROR:core:pv_parse_spec: error searching pvar "avp" /usr/local/sbin/kamailio[29243]: ERROR:core:pv_parse_spec: wrong char [s/115] in [$avp(s:signaling_ip)] at [5 (5)] /usr/local/sbin/kamailio[29243]: CRITICAL:mediaproxy:mod_init: invalid AVP specification for signaling_ip_avp: `$avp(s:signaling_ip)' /usr/local/sbin/kamailio[29243]: ERROR:core:init_mod: failed to initialize module mediaproxy This is the line in the kamailio.cfg file : modparam("mediaproxy", "signaling_ip_avp", "$avp(s:signaling_ip)") I'm using kamailio 1.5.2 What I'm missing here?. Thanks, Ricardo Martinez

2 2

[Kamailio-Users] SIMPLE vs XMPP: The Resolution
by Iñaki Baz Castillo 13 Oct '09

13 Oct '09

Architectures of XMPP and SIMPLE (OMA specifications) for a full rich presence environment: - XMPP: http://oversip.net/public/XMPP.jpeg - SIMPLE (OMA): http://oversip.net/public/SIP_SIMPLE_OMA.jpeg -- Iñaki Baz Castillo <ibc(a)aliax.net>

8 19

[Kamailio-Users] SIMPLE vs XMPP: The Resolution
by Iñaki Baz Castillo 13 Oct '09

13 Oct '09

Attached the architecture of XMPP and SIMPLE (OMA specifications) for a presence environment. Self explanatory. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 7

[Kamailio-Users] Connecting two or more proxies
by Mythili D 13 Oct '09

13 Oct '09

Hi Good morning. I'm going to connect two proxies. I'm having two servers with openser installed. One user (UAC) will register with one server in one system which is not having openser and the other user (UAS) will register with second server in other syatem which is not having openser. I want to connect the two proxies through cfg file. Hoiw to give the connection. My IP's are UAC registered with 1000(a)192.168.64.77 at other system (192.168.64.85) 192.168.64.77 is one proxy. 192.168.64.94 is secind proxy. UAS registerd with 10(a)192.168.64.94 at other system (192.168.64.128) Now the person at 85 registered with 77 is going to call the person at 128 registered with 94. I have configuered my cfg's in this way: In 77 cfg file, i configuered *if(is_method("INVITE") && !src_ip=="192.168.64.77") * *{ * * rewritehost("192.168.64.77"); * *} * ** In 99 cfg file, i configuered *if(is_method("INVITE") && !src_ip=="192.168.64.94") * *{ * *rewritehost("192.168.64.94"); * *}* Is there any other way to connect. Please help me as soon as possible. Thanks & Regards, Mythili.K

2 1

[Kamailio-Users] keep tos value from incoming SIP message
by kévin l'heveder 13 Oct '09

13 Oct '09

Hi all, I found out how to set the tos value in a outgoing SIP message with the tos parameter from core-cookbook. But my issue is to forward an incoming SIP message keeping the tos value unchange. Is there already a way to do that, something already implemented? Or is there a way to get the tos value from incoming message and dynamically change the tos parameter? Regards,

2 1

[Kamailio-Users] using a proxy and a registrar(only) server to make a cal
by Rajesh 13 Oct '09

13 Oct '09

Dear All, I would like to establish a set up where i have two servers, one acting as a proxy and the other acting as a registrar server(only). Client A ---->Proxy--->Registrar. I am able to register using the above mentioned set up, but am unable to make call.. i get "timed out error" when i try calling. am ataching my cfg below.Any help is greatly appreciated. (My proxy server wil not do registration and my registrar server wil only do registration) ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "imc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr 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) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* 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", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("acc", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #-------tm params----- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) #-------imc params----- modparam("imc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("imc", "hash_size", 8) modparam("imc", "imc_cmd_start_char", "#") #modparam("imc", "imc_cmd_start_char", "#") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication xlog("__The next step is Registration__\n"); rewritehost("192.168.64.77"); xlog("__Registration Successful__\n"); #nr route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain #nr if( is_method("PUBLISH|SUBSCRIBE")) #nr route(2); #nr if (is_method("REGISTER")) #nr { #nr if (!save("location")) #nr sl_reply_error(); #nr exit; #nr } ########## if(uri=~"sip:q.*@") # if(uri=~"sip:q@192.168.64.75") { # IMC - message xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { 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 setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("192.168.64.75", "subscriber")) { www_challenge("192.168.64.75", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { #imc#if (!proxy_authorize("", "subscriber")) { #imc#proxy_challenge("", "0"); #imc#exit; #imc#} #imc#if (is_method("PUBLISH")) #imc#{ #imc# if ($au!=$tU) { #imc# sl_send_reply("403","Forbidden auth ID"); #imc# exit; #imc# } #imc#} else { #imc# if ($au!=$fU) { #imc# sl_send_reply("403","Forbidden auth ID"); #imc# exit; #imc# } #imc#} #imc# consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE") #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { 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(); ##} } -- Regards, Rajesh

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Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Henning Westerholt 13 Oct '09

13 Oct '09

On Montag, 12. Oktober 2009, mashudi wrote: > Hi Henning Westerholt , > did you ever handle SIP-T using Kamailio? > could you share it? Hi Mashudi, sorry, i did not handled SIP-T so far with openser/kamailio. Regards, Henning > > On Friday 13 February 2009, Ovidiu Sas wrote: > >> The multipart body content parsing is embedded in the sdp parser. > >> The sdp parser is available in the source tree under the ./parser/sdp > >> directory. > > > > There is also some documentation about the parser on our wiki: > > > > http://www.kamailio.org/dokuwiki/doku.php/development:sdp-parsing > > > > Cheers, > > > > Henning > > > > _______________________________________________ > > Kamailio (OpenSER) - Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > ***************************************** > > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > > Yogya > > ***************************************** > > ================================= > Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, > DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? > Ikuti Dahsyatnya FLEXI KOMUNITAS. > Ketik CREATE<spasi>[NAMA GRUP], sms ke 345. > Contoh: CREATE SMU2, sms ke 345. > Informasi selanjutnya : > - hubungi 147 > - http://www.telkomflexi.com > - ketik INFO, sms ke 345 > > > ================================= > This Email has been scanned : Anti Virus > =================================

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[Kamailio-Users] Sip-T
by ljjimenez＠gmail.com 12 Oct '09

12 Oct '09

Hi list, can somebody guide on how to read, if it is possible, application/isup payload sent using SIP-T to do routing decisions based of the contents. Thanks in advance Luis Jimenez Luis Jimenez

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[Serusers] Via: branch tags for CANCELs sometimes don't match the one in the INVITE
by Frank Durda IV 12 Oct '09

12 Oct '09

I have been tracking a problem for some time regarding CANCELs that the switch beyond SER was randomly ignoring. There appear to be a number of issues, but one that was identified is caused by SER unexpectedly emitting a branch tag on the Via: header of the CANCEL with a value that isn't identical to the value that appeared in the INVITE message. When the CANCEL arrives at the destination switch with a different branch tag than what was shown in the INVITE, the destination switch correctly rejects the CANCEL with a 481, saying it never heard of this call. So the call doesn't really end. They say they are following RFC 3261, and that does state that the Via: header MUST exactly match. This variance of the branch tag that SER emits doesn't happen every time and is infrequent and random, occurring, perhaps 1% of the time. 99% of the time, the same branch value appears in both INVITE and CANCEL messages. This occasional mismatch is being seen in 2.0.0-rc1 on FreeBSD, Server: SER (2.0.0-rc1 (x86_64/freebsd)) syn_branch=0 is set because branches of zero in ACKs were a really bad thing too. Based on a number of these that have been reviewed, the branch tag generated by SER has three parts, eg branch=STRING1.STRING2.0 I note that STRING1 remains the same but when the malfunction occurs, STRING2 is completely different, as in: FROM INVITE branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c11079.0 FROM CANCEL branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd34935.0 Below is a sample call showing one of these events. Any suggestions on what the problem might be or if this is a known bug (hopefully with a patch) are appreciated. - - - - - - - - - - - - - - - - - - Here is a sample call. IP addresses and phone numbers have been altered, but call ID and branch tags are untouched: CALL ARRIVES AT SER U 2009/10/05 23:00:45.616070 216.0.0.1:5060 -> 208.9.10.11:5060 INVITE sip:9615551212@208.9.10.11:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C P-Asserted-Identity: <sip:8881112222@216.0.0.1:5060> To: sip:9615551212@208.9.10.11:5060 Contact: <sip:8881112222@216.0.0.1:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE Max-Forwards: 70 Content-Length: 281 Allow: INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 v=0 o=SIPUAC_1.92.1.8 2077471748 1 IN IP4 55.66.77.88 s=call c=IN IP4 55.66.77.88 t=0 0 m=audio 21356 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 SER SENDS TRYING BACK TO CALLING SWITCH U 2009/10/05 23:00:45.626174 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 100 trying -- your call is important to us From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: sip:9615551212@208.9.10.11:5060 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: SER (2.0.0-rc1 (x86_64/freebsd)) Content-Length: 0 SER SENDS CALL ON TO PSTN SWITCH U 2009/10/05 23:00:45.626192 10.100.20.42:5060 -> 10.133.44.55:5060 INVITE sip:9615551212@10.133.44.55:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C P-Asserted-Identity: <sip:8881112222@216.0.0.1:5060> To: sip:9615551212@208.9.10.11:5060 Contact: <sip:8881112222@216.0.0.1:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE Max-Forwards: 16 Content-Length: 301 Allow: INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 v=0 o=SIPUAC_1.92.1.8 2077471748 1 IN IP4 55.66.77.88 s=call c=IN IP4 55.66.77.88 t=0 0 m=audio 55878 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 a=nortpproxy:yes CALLING SWITCH TELLS SER IT IS TRYING U 2009/10/05 23:00:45.629000 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 100 Trying Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Content-Length: 0 CALLING PARTY CHANGES MIND - GOES ON-HOOK U 2009/10/05 23:00:50.579059 216.0.0.1:5060 -> 208.9.10.11:5060 CANCEL sip:9615551212@208.9.10.11:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 SER SENDS CANCEL TO PSTN SWITCH, BUT BRANCH TAGS DON'T MATCH: U 2009/10/05 23:00:50.590039 10.100.20.42:5060 -> 10.133.44.55:5060 CANCEL sip:9615551212@10.133.44.55:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL Max-Forwards: 16 Content-Length: 0 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd3493 5.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 NOTE BRANCH MISMATCH: FROM INVITE branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c11079.0 FROM CANCEL branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd34935.0 SER SENDS 200 CANCELING BACK TO CALLING SWITCH U 2009/10/05 23:00:50.590054 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 200 canceling From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=68cf7f6b22161b0d4dc719dbc13e05af-90a6 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: SER (2.0.0-rc1 (x86_64/freebsd)) Content-Length: 0 PSTN SWITCH REJECTS CANCEL, NOT RECOGNIZING THE CALL PSTN NETWORK CONTINUES TO PROCESS CALL DESPITE CALLER NOW BEING ON-HOOK U 2009/10/05 23:00:50.591489 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 481 Call/Transaction Does Not Exist Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+0+b6afb3ac Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd3493 5.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Content-Length: 0 PSTN SWITCH SENDS BACK ACM/183 Session Progress TO SER U 2009/10/05 23:00:52.619487 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 183 Session Progress Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@10.133.44.55:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no SER SENDS 183 SESSION PROGRESS BACK TO CALLING SWITCH U 2009/10/05 23:00:52.629924 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 183 Session Progress Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@208.9.10.11:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no SER RECEIVES 200 OK FROM PSTN SWITCH U 2009/10/05 23:00:58.129670 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@10.133.44.55:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no SER PASSES 200 OK ON TO CALLING SWITCH U 2009/10/05 23:00:58.137724 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@208.9.10.11:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no BECAUSE OF NO RESPONSE TO 200 OK, PSTN SWITCH SENDS IT AGAIN U 2009/10/05 23:00:58.629267 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@10.133.44.55:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no U 2009/10/05 23:00:58.639701 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@208.9.10.11:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no CALLING SWITCH FINALLY SENDS AN ACK TO SER TO GET RETRIES TO STOP. U 2009/10/05 23:00:58.690296 216.0.0.1:5060 -> 208.9.10.11:5060 ACK sip:9615551212@208.9.10.11:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 ACK Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_2 SER FORWARDS ACK TO PSTN SWITCH. PSTN NETWORK THINKS CALL IS ALIVE, VOIP NETWORK KNOWS CALL IS DEAD. U 2009/10/05 23:00:58.690355 10.100.20.42:5060 -> 10.133.44.55:5060 ACK sip:9615551212@10.133.44.55:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 ACK Max-Forwards: 16 Content-Length: 0 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.1fc8626bef702f1428ad6fd4fd2aa7a f.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_2 .... So the called party has answered the call and can't hear anything. Depending on whether a human or automation was called, this call could remain active on the PSTN for some time.

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Henning Westerholt 12 Oct '09

12 Oct '09

On Montag, 12. Oktober 2009, Vibhor Singhal wrote: > Please find attached the text file that contains the error that I encounter > while installing Kamailio.I could not get any solution for this.Please > guide me how to resolve this problem. Hi Vibhor, just paste the error message in the mail, its easier then this office document.. You need to install the correct mysql development libraries to be able to compile the db_mysql module. For debian you'll find the package names in the documents in the wiki i posted to you i think. And i'd suggest that you use the latest stable version of kamailio "1.5.2" instead of 1.5.0. This way you'll not run into bugs that were already fixed. Regards, Henning

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Henning Westerholt 12 Oct '09

12 Oct '09

On Montag, 12. Oktober 2009, Vibhor Singhal wrote: > Thanks for the information. Can you please refer any forum or document that > can guide me on how to make Kamailio as Asterisk's frontend. Hi Vibhor, please CC always to the user list. Just use the search machine of your choice. ;) Here are some example posts with more information: http://www.voip-info.org/wiki/view/Asterisk+at+large http://www.kamailio.org/pipermail/users/2008-February/016181.html http://www.nabble.com/Load-Balancing-Asterisk-using-Kamailio-td23250215.html > And in Kamailio's SIP_trace module, will I get SIP message in the raw form > as mentioned below: Hm, never used this module, but i think it will store the messages in the database. But you could modfiy the module, its open source after all.. Regards, Henning

2 1

[Kamailio-Users] Retransmitted SUBSCRIBEs
by catalina oancea 12 Oct '09

12 Oct '09

Hello, If a phone re-sends a presence SUBSCRIBE requests even after 5 or 500 milliseconds, kamailio replied with '400 Bad request' to the second SUBSCRIBE The scenario is: Phone Kamailio SUBSCRIBE ----------> 401 <---------- SUBSCRIBE ----------> 202 <---------- Pause [ 5ms] (or 500 ms, the same thing happens) SUBSCRIBE ----------> (previous SUBSCRIBE retransmitted) 400 Bad request <---------- Is this the expected behavior? Is there any way to get kamailio to recognize the retransmission? This causes problems with Snom phones because they receive 400 reply when they retransmit the SUBSCRIBE, and presence stops working. Thanks Catalina

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[Kamailio-Users] off-topic :: empty user part of Contact header
by kokoska rokoska 12 Oct '09

12 Oct '09

Hello, I have a little bit off-topic question, but - may be - someone could give me an advice :-) One of our users, when replying to INVITE, sends in 200 OK Contact header without user part: SIP/2.0 200 OK CSeq: 102 INVITE Contact: <sip:1.2.3.4:1234> ... instead of: SIP/2.0 200 OK CSeq: 102 INVITE Contact: <sip:user@1.2.3.4:1234> ... and it fools our SIP<->ISUP GW and it generates completely strange R-URI in ACK :-( My questions are: 1. Is it valid Contact (without user part)? 2. What SIP<->ISUP GW should do (mirror Contact without user part to R-URI in ACK, reject/drop 200 OK etc.)? May be I can ask them to change behaviour... 3. What should do our proxy (drop 200 OK, using htable try to repair R-URI in ACK or Contact in 200 OK etc.)? 4. ??? Any suggestion is very appreciated :-) Best regards, kokoska.rokoska

2 2

Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Henning Westerholt 09 Oct '09

09 Oct '09

On Freitag, 9. Oktober 2009, Vibhor Singhal wrote: > We are developing CTI applications where we need SIP data from PBX such as > SourceIP,Username,Destination IP,TAG,VIA,Contact,CALL-ID,C-Sequence etc for > each of the SIP packets such as REGISTER,BYE,INVITE,ACK,SUBSCRIBE,NOTIFY > etc. > > e.g Wrieshark displays SIP packets for activities going on SIP port in the > below mentioned format: > [..] > > For this we are trying Kamailio as Asterisk frontend.Plz tell us about the > installation process and how to make kamailio as asterisk's fromtend Hallo Vibhor, for installation of kamailio take a look to the wiki, for example at: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn In the "installation" section there are also some more informations: http://www.kamailio.org:80/dokuwiki/doku.php How to use kamailio as a frontend to asterisk has been discussed a lot on this list in the past, the easiest way is probably use the "dispatcher" module. Take a look to the docs for this module for more informations. Best regards, Henning

1 0

[Kamailio-Users] Presence issue - pua dialoginfo
by Francisco Javier Lizarán Vilches 09 Oct '09

09 Oct '09

Hi all: I have a problem using Kamailio 1.5.2 as proxy&presence server. User A is subscribed to B's event dialog, and Kamailio handles this subscription. When a call arrives to pstn number C, the proxy resolves it via ENUM and sends the INVITE to B. The issue is that the PUBLISH request uri is C, because pua_dialoginfo module looks into To uri instead of Request uri, so A is not informed about B state. Has anybody any idea to work around this? Thanks: Francisco Javier Lizarán Vilches ---------------------------------------------- BARIK (Grupo Ormazabal) Departamento Técnico Tel.: +34 917 479 900 www.barik.es <http://www.barik.es/> ----------------------------------------------

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Iñaki Baz Castillo 09 Oct '09

09 Oct '09

2009/10/9 Vibhor Singhal <vibhor.singhal(a)coraltele.com>: > We are a telcom company in Noida developing CTI applications. > We are using Asterisk-1.4.and have configured manager.conf file and > connection with the AMI is successfull using username and password. We want > to monitor/fetch SIP packets from TCP port. Please let us know how to take > SIP data from tcp port. This is a list about Kamailio, not about Asterisk.

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Raúl Alexis Betancor Santana 09 Oct '09

09 Oct '09

On Friday 09 October 2009 06:56:51 Vibhor Singhal wrote: > Hi, > > > We are a telcom company in Noida developing CTI applications. > We are using Asterisk-1.4.and have configured manager.conf file and > connection with the AMI is successfull using username and password. We want > to monitor/fetch SIP packets from TCP port. Please let us know how to take > SIP data from tcp port. First of all .. this is not the correct list to ask this kind of questions . Then Asterisk does not support SIP over TCP (not yet). > Our scenario: > > We are developing CTI applications where we need SIP data from PBX such as > SourceIP,Username,Destination IP,TAG,VIA,Contact,CALL-ID,C-Sequence etc for > each of the SIP packets such as REGISTER,BYE,INVITE,ACK,SUBSCRIBE,NOTIFY > etc. Asterisk will not give you that information over the AMI, you will need to develop a module for asterisk or modify chan_sip to archive that goal. > e.g Wrieshark displays SIP packets for activities going on SIP port in the > below mentioned format: > After receiving these SIP Packets from Asterisk,we can parse that data > according to our requirements. > Is there any text/data file or database where asterisk stores all the > information in above format? No > If we need to fetch this data at realtime from the port than how can we do > that? You have toons of ways for skying a cat ... you could ... : - Develop your custom sip-cature program, using libpcap for example - Work with ngrep for getting about what you need - Develop a wireshark filter. - ... > For this we are trying Kamailio as Asterisk frontend.Plz tell us about the > installation process and how to make kamailio as asterisk's fromtend > > > Request for an early reply!!! > Regards If you need such kind of work and you don't have time to learn yourself, better to look up for a consultant and pay for that work. -- Raúl Alexis Betancor Santana Dimensión Virtual

2 1

[Kamailio-Users] How to fetch SIP Data from Asterisk?
by Vibhor Singhal 09 Oct '09

09 Oct '09

Hi, We are a telcom company in Noida developing CTI applications. We are using Asterisk-1.4.and have configured manager.conf file and connection with the AMI is successfull using username and password. We want to monitor/fetch SIP packets from TCP port. Please let us know how to take SIP data from tcp port. Our scenario: We are developing CTI applications where we need SIP data from PBX such as SourceIP,Username,Destination IP,TAG,VIA,Contact,CALL-ID,C-Sequence etc for each of the SIP packets such as REGISTER,BYE,INVITE,ACK,SUBSCRIBE,NOTIFY etc. e.g Wrieshark displays SIP packets for activities going on SIP port in the below mentioned format: o SIP Packet as displayed on wireshark******************* SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.35:5060;branch=z9hG4bK10e203f8;rport=5060 Contact: <sip:1000000@192.168.3.15:45118> To: <sip:1000@192.168.3.15:45118>;tag=1f0f342c From: "Unknown"<sip:Unknown@192.168.3.35>;tag=as4a585648 Call-ID: 38670a55c46d1235YzY2YWQxZTM0ODk4NTRhZDI5MjY3YjMyYjE4OTlhZjc. CSeq: 102 NOTIFY User-Agent: X-Lite release 1003l stamp 30942 Content-Length: 0 After receiving these SIP Packets from Asterisk,we can parse that data according to our requirements. Is there any text/data file or database where asterisk stores all the information in above format? If we need to fetch this data at realtime from the port than how can we do that? For this we are trying Kamailio as Asterisk frontend.Plz tell us about the installation process and how to make kamailio as asterisk's fromtend Request for an early reply!!! Regards Regards, Vibhor Singhal Software Engineer Mob : +919891380819 " If you fail to plan,you plan to fail" Coral Telecom Limited E2, Sector 63, Noida - 201301, UP, India. F: +91-120-4035888 E: corporate(a)coraltele.com www.coraltele.com <http://www.coraltele.com/index.asp>

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[Kamailio-Users] Thousands of CANCEL messages - problem with my configuration?
by CSB 08 Oct '09

08 Oct '09

We had a very strange situation last night with our proxy (running OpenSer 1.3). We received an INVITE from a UA. This was then cancelled by the UA. This then resulted in OpenSER generating 40000+ cancel messages with our server talking to itself i.e. the toip and fromip are the same in the sip_trace table. Incidentally, this proxy processes thousands of calls a day so I'm not sure what gave rise to this problem but I guess a particular set of events has highlighted a bug in my config file. The other very curious thing is that somehow our other SIP proxy gets in on the act. Cancel message 40000-odd looks like this: CANCEL sip:031234567@domain.com SIP/2.0 Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.5d037b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.2d037b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.67037b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.20037b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.8ff27b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.e16dcb3.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.516dcb3.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.3c5dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.05e27b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.b4e27b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.365dcb3.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.52e27b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.be4dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.33c27b85.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.2d3dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.84a27b85.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.da2dcb3.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.002dcb3.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.561dcb3.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.111dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.4b627b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.ca627b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.5a627b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.1a627b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.ed0dcb3.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.d9627b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.c9627b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.b9627b85.0 Via: SIP/2.0/UDP 210.48.001.001:5060;branch=z9hG4bK519fe576;rport=5060 From: "Customer" <sip:12345678@domain.com>;tag=as02736614 To: <sip:031234567@domain.com> Contact: <sip:6496306317@210.48.001.001> Call-ID: 771d537932ab79225888a341190691c2(a)domain.com CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 41 Content-Length: 0 I have pasted what I think are the relevant parts from the config file. Could anyone advise where the problem might be? route { # ----------------------------------------------------------------- ### SipTrace. # ----------------------------------------------------------------- if ! is_method("OPTIONS|SUBSCRIBE|NOTIFY") { sip_trace(); setflag(28); } # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #If it's an INVITE & client is NATed, #xlog("welcome"); if (method=="INVITE" && client_nat_test("3")) { #Record-route and specify the record-route header explicitly record_route_preset("147.202.001.001:5060;nat=yes"); # insert IP address xlog("invite and nated so record_route_preset"); #If not Nated and not REGISTER then normal record-route } else if (method!="REGISTER") { record_route(); xlog("not register and not nated so record_route"); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); #call end_media_session in case we're proxying media end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { #Ensure we are dealing with a re-INVITE. Only connected calls have tag= #entry in the To header. So if it's loose routed and doesn't have totag #and it's an invite or reply then something's wrong xlog("loose_route"); if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); xlog("403 in lr"); return; }; if (method=="INVITE") { if (!allow_trusted()) { xlog("!allow_ trusted for r-uri <$ru>"); if (!proxy_authorize("","subscriber")) { proxy_challenge("domain.com","1"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; #check client is nated or that we've already identified it's nated xlog("Re-invite. rs $rs si $si rm $rm ru $ru tu $tu fu $fu fd $fd rr $rr"); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setbflag(6); xlog("Recipient is nated so setbflag 6 and use mediaproxy: r-uri <$ru>"); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # Processes out-of-dialogue messages i.e. # 1. new dialogue # 2. message not destined for us (but we are relaying or proxying) xlog("Call Type Processing Section"); if (!is_uri_host_local()) { xlog("!is_uri_host_local"); if (is_from_local() || allow_trusted() ) { xlog("is_from_local() || allow_trusted()"); route(5); route(1); } else { sl_send_reply("403", "Forbidden"); xlog("403 in call type processing. rs $rs si $si rm $rm ru $ru tu $tu fu $fu fd $fd rr $rr"); }; return; }; sip_trace(); if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; } else if (method=="SUBSCRIBE") { route(7); return; }; route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- xlog("hit route(1). rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); #Call reply_route(1) to intercept response messages heading to the client t_on_reply("1"); #Try to relay the message to its destination if (!t_relay()) { #If it can't and it's an INVITE or ACK end the media proxying log(1,"route[1] !t_relay"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); } else { log(1,"route[1] t_relay"); }; } onreply_route[1] #Handles message that are returned to the sender i.e. response to caller's original request { xlog("hit onreply_route(1). rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); if (isbflagset(6)) { xlog("flag(6) is set for $fu . We're currently in onreply_route(1)"); }; if (isbflagset(7)) { xlog("flag(7) is set for $fu . We're currently in onreply_route(1)"); }; if (status=~"(180)|(183)|2[0-9][0-9]") { xlog("status matches for $fu . We're currently in onreply_route(1)"); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #Check the SDP payload length. Assume if we have something we can call mediaproxy if (!search("^Content-Length:[ ]*0")) { log(1,"using media proxy in onreply_route(1)"); use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Any advice appreciated. Cameron

3 3

[Kamailio-Users] BLF and MWI problem on Grandstreams
by David 08 Oct '09

08 Oct '09

All, I upgraded to Kamailio 1.5 and had a lot of problems getting BLF and MWI working. So I will list here some of my notes in case some body else ever has the same issue. Firstly, in my tests I am using a GXP2000, GXP2020 and a SPA962. I also tested with a SNOM360. my issue was the MWI seemed to fail after an hour, and I could not find why. So in searching my config, I found that when the phones were resubscribing to MWI and BLF, it was being caught in this condition : if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(4); } else { sl_send_reply("404","Not here"); exit; } } My subscribe code handling was after this block, I noticed the issue from the 404 the grandstream was getting when trying to subscribe. I therefore added handling for the subscribe packets in this section, and my MWI and BLF seems to be working. else if ( is_method("PUBLISH") || is_method("SUBSCRIBE") ) { route(5); exit ; } Where route(5) does the appropriate handling of these two types of SIP packets. It should be noted, when I speak of MWI I am referring to presence message-summary which is sent via a publish by my media application. When I write BLF, I am referring to Presence Dialog. So when my phones were resubscribing, they are being sent a 404. I tested the Grandstream and the Linksys, and both seem to have the problem resolved. I will test this with the Snom tomorrow and post my results. David

3 4

[Kamailio-Users] Somebody going to Astricon?
by Klaus Darilion 08 Oct '09

08 Oct '09

..so we should hang out for a beer :-)

4 3

[Kamailio-Users] help with custom module
by liran tal 07 Oct '09

07 Oct '09

Hey list, I've written a very simple module, just exporting one function I need which does something simple like checking a file and returns -1 or 1 if the file exist or not. The problem I'm facing is as follows: During call flow (extension1 calling extension2), it seems that the logic of the call flow is passed via that module and hangs the entire call process. And this happens without even calling that single exported function, but rather just by including the module itself via loadmodule. Log from the call flow: ... 8(3827) SIP Reply (status): 8(3827) version: <SIP/2.0> 8(3827) status: <100> 8(3827) reason: <Trying> 8(3827) parse_headers: flags=2 8(3827) Found param type 232, <branch> = <z9hG4bKde5a.32fd75d7.0>; state=9 8(3827) parse_via: next_via 8(3827) Found param type 232, <branch> = <z9hG4bK-d8754z-303e27391e65ac7d-1---d8754z->; state=6 8(3827) Found param type 235, <rport> = <21644>; state=16 8(3827) end of header reached, state=5 8(3827) parse_headers: Via found, flags=2 8(3827) parse_headers: this is the first via 8(3827) After parse_msg... * 8(3827) forward_reply: found module exec_custom, passing reply to it * 8(3827) DEBUG:destroy_avp_list: destroying list (nil) 8(3827) receive_msg: cleaning up ... that forward_reply thing has the module involved there for whatever reason, I've no idea. This is on openser1.0.1 I'd appreciate some feedback Regards, Liran.

2 1

[Kamailio-Users] PostgreSQL TLS support
by Olle E. Johansson 07 Oct '09

07 Oct '09

The db_postgres documentation doesn't say much about anything, really... Is there support for TLS connections and how do I configure it if it exist? Thanks, /O

2 4

[Kamailio-Users] LDAP and TLS
by Olle E. Johansson 07 Oct '09

07 Oct '09

Greetings, fellow Kamailians! Looking at the LDAP module for Kamailio, I see that we can use LDAPS: and get TLS connections. But where are the SSL configurations so I can use a client cert, or verify the server cert? /O

1 0

[Kamailio-Users] use as registrar only server
by Rajesh 07 Oct '09

07 Oct '09

I would like to know what all can be done using openser/kamailio as a registrar server alone. i.e. in the real world scenario how does openser can be useful as a registrar sever alone.I would like to make myself clear by pointing to the below example. i can integrate openser and asterisk and make openser as a registrar (only)server and can make use of asterisk for call functionality purposes.. Similarly, i would like to know where openser fits in (openser should only be used as a registrar server again). Your guidance is greatly appreciated. -- Regards, Rajesh

1 0

[Kamailio-Users] How can I access dialog attributes from within timeout route?
by Nathan Angelacos 07 Oct '09

07 Oct '09

Hi, In kamailio 1.5.2-notls, I'm trying to get the CallID of a dialog that has timed out: if (is_method( "INVITE" ) && !has_totag()) { $dlg_ctx(timeout_route) = 21; } ... route[21] { xlog ("L_WARN", "Dead dialog: $dlg(callid)"); exit; } However, syslog says: daemon.warn /usr/sbin/kamailio[6578]: Dead Dialog: <null> daemon.warn /usr/sbin/kamailio[6578]: WARNING:dialog:dlg_ontimeout: timeout for dlg with CallID \ '8b2eb5e3-20493c20-14e681d(a)192.168.1.1' and tags '87F489DA-922DF157' '2XrwB5OcEJ2zQvyiTUIh-l2Zd1wjL51M' dlg_handlers.c leads me to believe that the context is loaded before calling the route(21), and the dialog timeout handler is nice enough to report it in syslog just after my script fails. Am I missing something fundamental, or is this just not possible? Thanks!

2 8

[Serusers] SER and RTPproxy
by Jeff Russell 06 Oct '09

06 Oct '09

I Think I have thing configured correctly and yet I can not make things work correctly. This is what I am do I have SER on an internal (10.1.1.x) IP address I have RTPproxy on a different machine with a public IP address 206.75.15.x I am routing the call out over our firewall and things 'almost' work. SER sets up the call using the RTPproxy to set up the ports for the RTP stream, and the RTP stream is returned from the far end to the public IP address, however non of the RTP traffic is sent to the internal address, I am guessing it is waiting for an rtp stream from the internal address but neither SER nor the ip-phone send anything or even seem to know about the RTPproxy. Can anyone offer ANY suggestions as to how to direct the internal RTP stream to the RTPproxy address? Many thanks, Jeffr

2 2

Re: [Serusers] SER and RTPproxy
by Edson - Lists 06 Oct '09

06 Oct '09

Please, don't forget to copy the list... _this_ is a list... ;) Edson. Jeff Russell escreveu: > I think this is where the error is: > > 0(60402) ERROR: extract_body: message body has length zero > 0(60402) ERROR: force_rtp_proxy2: can't extract body from the message > > ideas? > > > > On 2009-10-06, at 11:08 AM, Edson - Lists wrote: > >> I would suggest You to use NGREP and sniff both legs of the >> communication flow... >> >> See what SER receives and responds, what's on the SDP body and look at >> syslog to understand what RTPproxy is doing. Maybe also take a look on >> what the internal client is sending and receiving. >> >> It should give You some hints. >> >> Edson. >> >> Jeff Russell escreveu: >>> I Think I have thing configured correctly and yet I can not make >>> things work correctly. >>> This is what I am do >>> I have SER on an internal (10.1.1.x) IP address >>> I have RTPproxy on a different machine with a public IP address >>> 206.75.15.x >>> I am routing the call out over our firewall and things 'almost' work. >>> SER sets up the call using the RTPproxy to set up the ports for the >>> RTP stream, and the RTP stream is returned from the far end to the >>> public IP address, however non of the RTP traffic is sent to the >>> internal address, I am guessing it is waiting for an rtp stream from >>> the internal address but neither SER nor the ip-phone send anything >>> or even seem to know about the RTPproxy. >>> Can anyone offer ANY suggestions as to how to direct the internal RTP >>> stream to the RTPproxy address? >>> Many thanks, >>> Jeffr >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Kamailio-Users] use of kamailo as a registrar (only)server
by Rajesh 06 Oct '09

06 Oct '09

I could implement openser as a registrar(only) server .Thanks to the team openser for the timely help.I would like to know what all can be done using openser/kamailio as a registrar server alone. i.e. in the real world scenario how does openser can be useful as a registrar sever alone.I would like to make myself clear by pointing to the below example. i can integrate openser and asterisk and make openser as a registrar (only)server and can make use of asterisk for call functionality purposes.. Similarly, i would like to know where openser fits in (openser should only be used as a registrar server again). Your responses can speed up my research work.. > > Saúl Ibarra wrote: > > You are doing save_location before authentication. You should first > challenge the REGISTER for authentication and then do save_location. > > 2009/10/6, Rajesh <Rajesh.sreedharan(a)siptech.com>: > > > Dear All, > I would like my kamailio server to act (only) as a > Registrar server. I have configured my kamilio for performing > authentication successfully, but this time i find that authentication is > not being done.Your views on my cfg is greatly appreciated. > > # ------------------ module loading ---------------------------------- > debug=3 > log_stderror=yes > mpath="/usr/local/lib/kamailio/modules/" > > loadmodule "db_mysql.so" > loadmodule "mi_fifo.so" > loadmodule "sl.so" > loadmodule "pv.so" > loadmodule "usrloc.so" > loadmodule "registrar.so" > loadmodule "textops.so" > loadmodule "uri_db.so" > > loadmodule "auth.so" > loadmodule "auth_db.so" > > port=5060 > # ----------------- setting module-specific parameters --------------- > > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > > # ----- auth_db params ----- > /* uncomment the following lines if you want to enable the DB based > authentication */ > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "db_url","mysql://openser:openserrw@localhost/openser") > modparam("auth_db", "load_credentials", "") > > # -- usrloc params -- > > modparam("usrloc", "db_url", > "mysql://openser:openserrw@192.168.64.75/openser") > > > modparam("usrloc", "db_mode", 3) > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > > # initial sanity checks -- too long messages > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > > if (!uri==myself) { > sl_send_reply("400", "Bad destination, only register > messages destined for this server are accepted"); > exit; > }; > > # the only valid message for the registrar is the register message > # other messages are answered with an error message > if (uri==myself) { > > if (method=="REGISTER") { > > sl_send_reply("100", "Trying"); > > if(!save("location")) { > sl_reply_error(); > }; > exit; > } else { > sl_send_reply("403", "Forbidden"); > exit; > }; > }; > > # Authentication route > > > if (is_method("REGISTER")) > { > # authenticate the REGISTER requests (uncomment to enable auth) > if (!www_authorize("192.168.64.75", "subscriber")) > { > www_challenge("192.168.64.75", "0"); > exit; > } > ###the following lines are uncommented > if ($au!=$tU) > { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } > else > { > # authenticate if from local subscriber (uncomment to enable auth) > if (from_uri==myself) > { > if (!proxy_authorize("", "subscriber")) > { > proxy_challenge("", "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 > } > } > return; > } > > > > -- > Regards, > Rajesh >

1 0

Re: [Kamailio-Users] kamailio as registrar server
by Saúl Ibarra 06 Oct '09

06 Oct '09

This should have gone to the list... :) On Tue, Oct 6, 2009 at 12:00 PM, Rajesh <Rajesh.sreedharan(a)siptech.com> wrote: > I could implement openser as a registrar(only) server .Thanks to the team > openser for the timely help.I would like to know what all can be done using > openser/kamailio as a registrar server alone. i.e. in the real world > scenario how does openser can be useful as a registrar sever alone.I would > like to make myself clear by pointing to the below example. i can integrate > openser and asterisk and make openser as a registrar (only)server and can > make use of asterisk for call functionality purposes.. Similarly, i would > like to know where openser fits in (openser should only be used as a > registrar server again). > Your responses can speed up my research work.. > > > Saúl Ibarra wrote: > > You are doing save_location before authentication. You should first > challenge the REGISTER for authentication and then do save_location. > > 2009/10/6, Rajesh <Rajesh.sreedharan(a)siptech.com>: > > > Dear All, > I would like my kamailio server to act (only) as a > Registrar server. I have configured my kamilio for performing > authentication successfully, but this time i find that authentication is > not being done.Your views on my cfg is greatly appreciated. > > # ------------------ module loading ---------------------------------- > debug=3 > log_stderror=yes > mpath="/usr/local/lib/kamailio/modules/" > > loadmodule "db_mysql.so" > loadmodule "mi_fifo.so" > loadmodule "sl.so" > loadmodule "pv.so" > loadmodule "usrloc.so" > loadmodule "registrar.so" > loadmodule "textops.so" > loadmodule "uri_db.so" > > loadmodule "auth.so" > loadmodule "auth_db.so" > > port=5060 > # ----------------- setting module-specific parameters --------------- > > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > > # ----- auth_db params ----- > /* uncomment the following lines if you want to enable the DB based > authentication */ > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "db_url","mysql://openser:openserrw@localhost/openser") > modparam("auth_db", "load_credentials", "") > > # -- usrloc params -- > > modparam("usrloc", "db_url", > "mysql://openser:openserrw@192.168.64.75/openser") > > > modparam("usrloc", "db_mode", 3) > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > > # initial sanity checks -- too long messages > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > > if (!uri==myself) { > sl_send_reply("400", "Bad destination, only register > messages destined for this server are accepted"); > exit; > }; > > # the only valid message for the registrar is the register message > # other messages are answered with an error message > if (uri==myself) { > > if (method=="REGISTER") { > > sl_send_reply("100", "Trying"); > > if(!save("location")) { > sl_reply_error(); > }; > exit; > } else { > sl_send_reply("403", "Forbidden"); > exit; > }; > }; > > # Authentication route > > > if (is_method("REGISTER")) > { > # authenticate the REGISTER requests (uncomment to enable auth) > if (!www_authorize("192.168.64.75", "subscriber")) > { > www_challenge("192.168.64.75", "0"); > exit; > } > ###the following lines are uncommented > if ($au!=$tU) > { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } > else > { > # authenticate if from local subscriber (uncomment to enable auth) > if (from_uri==myself) > { > if (!proxy_authorize("", "subscriber")) > { > proxy_challenge("", "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 > } > } > return; > } > > > > -- > Regards, > Rajesh > > > > > > -- > Regards, > Rajesh -- /Saúl http://www.saghul.net | http://www.sipdoc.net

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Re: [Kamailio-Users] fs_index.cxx:1: error: bad value (x86_64) for-march (mtune)= switch
by Iñaki Baz Castillo 06 Oct '09

06 Oct '09

¿?¿? El Lunes, 5 de Octubre de 2009, ?? escribió: > vendor_id : AuthenticAMD > cpu family : 16 > model : 2 > model name : Quad-Core AMD Opteron(tm) Processor 2350 > stepping : 2 > cpu MHz : 2010.303 > cache size : 512 KB > physical id : 1 > siblings : 4 > core id : 2 > cpu cores : 4 > fpu : yes > fpu_exception : yes > cpuid level : 5 > wp : yes > flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca > cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt > pdpe1gb rdtscp lm 3dnowext 3dnow pni monitor cx16 popcnt lahf_lm > cmp_legacy svm extapic cr8_legacy altmovcr8 abm sse4a misalignsse > 3dnowprefetch osvw bogomips : 4020.69 > TLB size : 1104 4K pages > clflush size : 64 > cache_alignment : 64 > address sizes : 48 bits physical, 48 bits virtual > power management: ts ttp tm stc 100mhzsteps hwpstate <NULL> > > > above is the info of the CPU. > > ************************************************************* > L Xue > Building 102, Room 102 > Department of Nuclear Physics > Shanghai Institute of Applied Physics, CAS > No. 2019, JiaLuo Roud > Shanghai, 201800 > P. R. China > Email:xueliang@sinap.ac.cn > MSN: xliang1984(a)hotmail.com > ************************************************************** > > > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org on behalf of Iñaki Baz Castillo > Sent: Mon 2009-10-5 23:37 > To: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] fs_index.cxx:1: error: bad value (x86_64) > for-march (mtune)= switch > > El Lunes, 5 de Octubre de 2009, Daniel-Constantin Mierla escribió: > > I am doing MC simulation at STAR. > > Is it a new VoIP service? > -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] kamailio as registrar server
by Rajesh 06 Oct '09

06 Oct '09

Dear All, I would like my kamailio server to act (only) as a Registrar server. I have configured my kamilio for performing authentication successfully, but this time i find that authentication is not being done.Your views on my cfg is greatly appreciated. # ------------------ module loading ---------------------------------- debug=3 log_stderror=yes mpath="/usr/local/lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "auth.so" loadmodule "auth_db.so" port=5060 # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url","mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # -- usrloc params -- modparam("usrloc", "db_url", "mysql://openser:openserrw@192.168.64.75/openser") modparam("usrloc", "db_mode", 3) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- too long messages if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!uri==myself) { sl_send_reply("400", "Bad destination, only register messages destined for this server are accepted"); exit; }; # the only valid message for the registrar is the register message # other messages are answered with an error message if (uri==myself) { if (method=="REGISTER") { sl_send_reply("100", "Trying"); if(!save("location")) { sl_reply_error(); }; exit; } else { sl_send_reply("403", "Forbidden"); exit; }; }; # Authentication route if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("192.168.64.75", "subscriber")) { www_challenge("192.168.64.75", "0"); exit; } ###the following lines are uncommented if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "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 } } return; } -- Regards, Rajesh

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[Serusers] Can SER 2.0.0-rc1 pstn.cfg file actually work with BYE from PSTN?
by Frank Durda IV 06 Oct '09

06 Oct '09

Does SER 2.0.0-rc1/examples/pstn.cfg or the much more complete one in Chapter 8 of "SER Getting Started" actually work when it comes to a BYE comming from the PSTN gateway? So far, it appears these two cfg files don't cope with the called party on the PSTN hangs up first, and it isn't clear how it could be made to work in anything more complex than a one-SIP-call-source one pstn-gateway configuration. Let's say I have four clients from which a call can originate and arrive at SER, with the caller source IP addresses of "A", "B", "C" and "D". "C" sends the INVITE of a call that SER determined is destined for the PSTN gateway, which is at IP address "P". Here, the INVITE, 100 Trying, 180/183, 200 OK and ACK messages all go to the right places and look more or less correct. If "C" sends a BYE to end the call, this also works correctly, with SER forwarding that on to "P" and the 200 OK being returned. So far, so good. However, if the called party on the PSTN hangs up first, "P" sends SER a BYE and SER simply sends the BYE right back to "P", whom responds with a "481" back. The originating party at "C" never finds out that the call has ended, because SER didn't pass the BYE on to the calling device. The SIP phone stays in the "Connected" state. In my case, the calling sites A, B, C and D don't include a Record-Route of their own, but sometimes have a Via: header. In either case, SER still doesn't send a BYE or re-INVITE coming from P to the correct destination of A, B, C or D. I don't know how I could force SER to send the BYE or INVITE on to the right place, given four possible call-initiator IP addresses, of which only "C" would know about this sample call. I don't believe I can stick in an extra Record-Route when handling the original INVITE that has the call source IP address in it (in addition to a Record-Route of my own), simply because I don't believe SER will let me dynamically select which senders IP address should go in the forged Record-Route, or if doing so would really help the situation later when that trail of crumbs might be helpful. Also, even though the pstn.cfg sample in the book has lots of steps to check for fraud before processing a re-INVITE coming from the PSTN (and the footnotes say that this is what these checks are for), I don't see how SER would know the right place to send the re-INVITE that came from the PSTN once it the authentication checks were done. It looks like SER will always send it to "P" regardless of which direction it should be sent, which doesn't help much, and puts any call using Session-Expires: to failure, when 50% of that time (plus 32.5 seconds) has passed. So, is there with a corrected pstn.cfg out there, or better still, has anybody actually made these two situations work, which are: (1) The called party on the PSTN hangs-up first which causes a BYE to be sent to SER. SER forwards the BYE to the right caller (or whatever the so-called proxy is supposed to do), AND (2) the SIP-PSTN gateway device sends a re-INVITE to SER and it goes somewhere that will generate an OK reply If someone has made it work, can they show an example, rather than just say "it should be possible" or "works for me"? Such responses are encouraging, but not that helpful. Thanks in advance!

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[Serusers] Is there a way to strip ";cic=nnnn" from INVITES?
by Frank Durda IV 06 Oct '09

06 Oct '09

I am receiving intermittent INVITES from a couple of sources where the INVITE includes a cic=nnnn. I need to strip this off before it reaches the PSTN gateway switch because that switch will try to use that value. However, it appears to be a pseudo-CIC and isn't an assigned carrier number so it can't be used for routing outside the network that stuck it in there. That network should be stripping it off before the call leaves their network, but they are not and the call has passed through a few entities by the time it reaches us, so we don't know whom to contact about the problem. Does SER have any mechanism that will strip this ONE item and leave the rest of the INVITE intact, eg so I can call more than one hard-coded number? The documentation I examined (iptel.org SIP Express Router v0.11.0 -- Admin's Guide, chapter 2.3, page 22) shows functions to delete or replace everything but attributes following a semi-colon and preceding the @. Is there something that is undocumented? Sample: (values altered) INVITE sip:9615551212;cic=9234@208.37.38.39;user=phone SIP/2.0 Need to change that to: INVITE sip:9615551212@208.37.38.39;user=phone SIP/2.0 (and leave any other parameters that might be present intact.) Thanks in advance for suggestions.

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[Kamailio-Users] new developer: Marius Zbihlei
by Henning Westerholt 05 Oct '09

05 Oct '09

Hi all, i'm happy to announce a new developer for the kamailio and sip-router project: Marius Zbihlei. Marius is a experienced C/C++ developer in our offices in Bukarest (Romania). He develops since a few years internal applications for the web hosting plattform of 1&1. Since app. a half year he is part of the team that handles the development and maintenance of our VoIP backend systems. Most of his work so far was targeted to our internal repositories, but Marius also already contributed a bunch of patches to the project. He'll support us with the maintenance of the kamailio code we contributed to the project in the past, and also work on new sip-router features in the future. Best regards, Henning -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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Re: [Kamailio-Users] fs_index.cxx:1: error: bad value (x86_64) for -march (mtune)= switch
by Daniel-Constantin Mierla 05 Oct '09

05 Oct '09

Hello, On 05.10.2009 17:29 Uhr, ?? wrote: > > hello， > > > I am doing MC simulation at STAR. I met this problem when I compile. > which mail list should I send? > had no idea what "mc simulation at star" is. Do you use kamailio sip server? http://www.kamailio.org If not, you are wrong here and I do not know what mailing list you should use. Daniel > Thanks. > > > xueliang > > ************************************************************* > L Xue > Building 102, Room 102 > Department of Nuclear Physics > Shanghai Institute of Applied Physics, CAS > No. 2019, JiaLuo Roud > Shanghai, 201800 > P. R. China > Email:xueliang@sinap.ac.cn > MSN: xliang1984(a)hotmail.com > ************************************************************** > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Mon 2009-10-5 16:21 > To: ?? > Cc: Users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] fs_index.cxx:1: error: bad value > (x86_64) for -march (mtune)= switch > > Hello, > > what application do you compile? Is not kamailio, so you probably sent > to wrong mailing list. > > Cheers, > Daniel > > On 04.10.2009 5:57 Uhr, ?? wrote: > > > > Hi , > > > > > > When I try to compile *.cxx on "Quad-Core AMD Opteron(tm) Processor > 2350" > > the follow happens: > > > > make[1]: Entering directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS' > > g++ -lpthread -lrt -Wall -g -I > > /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/include -L > > /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/lib -O3 -Wall -pipe > > -fverbose-asm -march=x86_64 -D_REENTRANT > > -DRTS_DAQMAN=\""172.16.0.1"\" -DTARGET_SYSTEM=\"LINUX\" > > -DPROJDIR=\"/RTScache\" -DINSTALL_SUFFIX=\"LINUX/x86_64\" > > -DRTS_PROJECT_STAR -DRTS_BIG_ENDIAN -DRTS_LOG_COLORED -g -I. -I../ > > -I../../include -I../../trg/include -c -o fs_index.o fs_index.cxx > > fs_index.cxx:1: error: bad value (x86_64) for -march= switch > > fs_index.cxx:1: error: bad value (x86_64) for -mtune= switch > > make[1]: *** [fs_index.o] Error 1 > > make[1]: Leaving directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS' > > > > How can I solve that > > Thanks and regards! > > > > > > xueliang > > > > > > ************************************************************* > > L Xue > > Building 102, Room 102 > > Department of Nuclear Physics > > Shanghai Institute of Applied Physics, CAS > > No. 2019, JiaLuo Roud > > Shanghai, 201800 > > P. R. China > > Email:xueliang@sinap.ac.cn > > MSN: xliang1984(a)hotmail.com > > ************************************************************** > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Kamailio (OpenSER) - Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- > Daniel-Constantin Mierla > * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin > * http://www.asipto.com/index.php/sip-router-masterclass/ > > > -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

2 1

asking odd mediaproxy,,
by Rakhmadhany Primananda 05 Oct '09

05 Oct '09

I've just setting SIP Express Router and Mediaproxy in 1 PC. Then, I'm setting in NAT Router that only activated for MASQUERADE like this: iptables -A POSTROUTING -s 172.17.2.0/24 -o eth1 -j MASQUERADE NB:I assume in network 10.100.0.0/24 (eth1) as public IP and 172.17.2.0/24as private IP. When I calling with x-lite and activated mediaproxy service, the connection running well with 2 direction (full duplex from 10.x.x.x to 172.x.x.x conversely).But,when I stopping mediaproxy service (for proving that voice packet if pass NAT will connect half duplex or maybe broken),apparently this connection still running well with 2 direction my question,, why this happened? are having trouble with mediaproxy or SIP Express Router? i'm confusing,, thanks very much,, :)

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[Kamailio-Users] Siremis v0.9.4 released
by Elena-Ramona Modroiu 05 Oct '09

05 Oct '09

Hello, Siremis - the web management interface for Kamailio SIP server - has been released with version 0.9.4. This is a patch release fixing the issues reported since v0.9.3 related to db connection samples in default configuration file. More details at: http://siremis.asipto.com Best regards, Ramona Modroiu

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[Kamailio-Users] Functions that don't accept pvars.
by Alex Balashov 05 Oct '09

05 Oct '09

It's very frustrating that some functions don't accept pseudovariables and there's no workaround. For example, in this case I need rtpproxy_offer() and rtpproxy_answer() to accept an IP address argument that is retrieved from a database. However, this won't work. rtpproxy_offer("", $var(src_ip)); # Config compilation error rtpproxy_offeR("", "$var(src_ip)"); # Results in $var(src_ip) # literally being placed in the # SDP body. I understand it is difficult to go back and update all legacy functions to accept PVs everywhere. But isn't it possible to provide a wrapper/compatibility function in the core that will parse a PV and generate as a result something that other functions can see as a string literal? -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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[Kamailio-Users] fs_index.cxx:1: error: bad value (x86_64) for -march (mtune)= switch
by ?? 05 Oct '09

05 Oct '09

Hi , When I try to compile *.cxx on "Quad-Core AMD Opteron(tm) Processor 2350" the follow happens: make[1]: Entering directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS' g++ -lpthread -lrt -Wall -g -I /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/include -L /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/lib -O3 -Wall -pipe -fverbose-asm -march=x86_64 -D_REENTRANT -DRTS_DAQMAN=\""172.16.0.1"\" -DTARGET_SYSTEM=\"LINUX\" -DPROJDIR=\"/RTScache\" -DINSTALL_SUFFIX=\"LINUX/x86_64\" -DRTS_PROJECT_STAR -DRTS_BIG_ENDIAN -DRTS_LOG_COLORED -g -I. -I../ -I../../include -I../../trg/include -c -o fs_index.o fs_index.cxx fs_index.cxx:1: error: bad value (x86_64) for -march= switch fs_index.cxx:1: error: bad value (x86_64) for -mtune= switch make[1]: *** [fs_index.o] Error 1 make[1]: Leaving directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS' How can I solve that Thanks and regards! xueliang ************************************************************* L Xue Building 102, Room 102 Department of Nuclear Physics Shanghai Institute of Applied Physics, CAS No. 2019, JiaLuo Roud Shanghai, 201800 P. R. China Email:xueliang@sinap.ac.cn MSN: xliang1984(a)hotmail.com **************************************************************

2 1

[Kamailio-Users] days to kamailio 3.0
by Daniel-Constantin Mierla 05 Oct '09

05 Oct '09

Hello, here is the roadmap planned to get to Kamailio 3.0: - this week testing using common git repository - end of this week have Kamailio 3.0 branched - 2 weeks testing of kamailio 3.0 branch and fix of discovered missing features from 1.5, packaging selected modules, name updates, installed tools, documentation updates, etc - 26 or 27 Oct day of release Comments/adjustments are welcome! I also think that we should look at releasing 1.5.3 since there were few important updates since 1.5.2. Cheers, Daniel -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

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SER/Kamailio 3.0 by what is kamctl replaced by?
by heinz huber 05 Oct '09

05 Oct '09

Hi, I've checked out the latest head from the git repository, compiled sip-router and installed it. How can i check how many users are registred at the proxy. In openser/kamailio days I used a simple 'kamctl ul show' and options to get some infos about the proxy 'internals'. I've read some lines about a ser_ctl command but can't find it my git working copy. I've followed the instructions on http://sip-router.org/wiki/migration/kamailio-3.0-config. OS: 2.6.18-5-686, debian 5.0.3 thx && br Andy -- Neu: GMX Doppel-FLAT mit Internet-Flatrate + Telefon-Flatrate für nur 19,99 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02

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Re: [Kamailio-Users] content about rtpproxy
by zhangchao00001 04 Oct '09

04 Oct '09

Hello everybody: I start rtpproxy as the root, on the same computer, rtpproxy 1.1 works well while rtpproxy 1.2 dosen't. >On 27.09.2009 2:56 Uhr, zhangchao00001 wrote: >> Hello everyone, i meet a problem when intergrating rtpproxy with >> kamailio. The two software start well, can communicate with each >> other, however, the rtpproxy always report "can't create listener" error. >> Dose anyone meet the same problem? >is the user that starts the rtpproxy has rights to create sockets? Is >there a firewall blocking higher ports? > >Cheers, >Daniel

1 0

[Serusers] simple blind call forwarding needed
by Martin Lane 02 Oct '09

02 Oct '09

SER Users: Thanks in advance for any help! I am running ser 0.9.6 and have a relatively simple cfg file. I have only four (4) digit extensions, no PSTN, no NAT and I do not care about authorization so my config file is basically the Hello World ser.cfg with my system parameters and a few statements in route that allowed me to create standing conference rooms and an echo application with sems. Now I need to implement call forwarding and I have been able to setup the MySQL ser usr_preferences table per chapter 10 of the 'SER - Getting Started' and the 'mySQL newbie? Problems with mySQL and SER?' documents. After adding the call forwarding functionality discussed in chapter 10 inside my config file any number dialed to a SIP device rings busy. All SIP devices register and can call the conference rooms or the echo application but they can not call each other. I will spare you my WireShark logs but can provide if that is needed. I'm looking for trouble shooting suggestions. Is there a way to print to the std i/o from inside the cfg file? This is the first time I needed to handle the INVITE message so I've included my INVITE Message Handler and the Call Forwarding Handler that I added for this effort: ... if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } if (method=="REGISTER") { route(2); break; }; ... route[3] { # ---------------------------------------------------------------------- # INVITE Message Handler # ---------------------------------------------------------------------- # Note: We are using this fuction only as a hook into the # blind call forwarding feature. Simply want to change the # R-URI and relay the message. if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); # Would love to do a printf here to see if this code is being hit!!! # Wireshark shows INVITE messages are being sent. # debug/printf("\n\n****** Inside route(3) ********\n\n"); # At this point the a blind call forwarding record was found and the # new destination was written in the R-URI. # DEBUG: try just calling route(1). # route(1); # Send to Call Forwarding Handler route(6); break; }; route(1); } route[6] { # ---------------------------------------------------------------------- # Blind Call Forwarding Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ---------------------------------------------------------------------- lookup("aliases"); if (!is_uri_host_local()) { if (!isflagset(22)) { append_branch(); }; route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{4}@") { route(1); break; }; sl_send_reply("404", "User Not Found"); break; }; # DEBUG: There are no alias' and we are not sending calls to other # networks. We may just need to call route(1) and not this # function. route(1); }

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[Serusers] Recent posting status
by ML Gmail 02 Oct '09

02 Oct '09

Hi: I posted a question to the list yesterday. I am a subscriber and the attributes for my profile have me included on all MSG traffic but I have not seen the MSG go out. Can you check on that? Should I resend the question? Thank you. Martin Lane

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Re: [Kamailio-Users] Siremis JSON error
by Elena-Ramona Modroiu 01 Oct '09

01 Oct '09

Luis Guaman wrote: > > Thanks Santiago, now it works great! > > siremis/metadata/Config.xml with changes: > > <?xml version="1.0" standalone="no"?> > > <Config> > > <DataSource> > > <Database Name="Default" Driver="Pdo_Mysql" Server="x.x.x.x" > DBName="openser" User="openser" Password="xxxx" /> > > <Database Name="Siremis" Driver="Pdo_Mysql" Server="x.x.x.x" > DBName="openser" User="openser" Password="xxxx" /> > > </DataSource> > > </Config> > so that was missing in the sample metadata/Config.xml file -- I did a quick 0.9.4 release to fix it. http://siremis.asipto.com/pub/downloads/siremis/siremis-0.9.4.tgz I will do a separate email to announce it. Best regards, Ramona > *De:* Santiago Gimeno [mailto:santiago.gimeno@gmail.com] > *Enviado el:* Jueves, 01 de Octubre de 2009 4:27 > *Para:* Luis Guaman > *CC:* users(a)lists.kamailio.org > *Asunto:* Re: [Kamailio-Users] Siremis JSON error > > Hi Luis, > > I ran into this problem too. In order to solve it I had to add a new > Database element to the siremis/metadata/Config.xml file with the Name > attribute set to Default and pointing to your openser database. I > don't know the reason of having two Databases elements pointing to the > same databases though. > > Best regards, > > Santi > > 2009/9/30 Luis Guaman <luis.guaman(a)interlancompu.com > <mailto:luis.guaman@interlancompu.com>> > > Hi, > > When we tried to run a Siremis Chart got the following error message: > > “Open Flash Chart > > JSON Parse Error [Syntax Error] > > Error at character 0, line 1: > > 0: <body style='font-family:Arial; font-size:12px; > background-color:#FCFCFC;'>Error m” > > Some help?? > > Atentamente, > > Luis Guamán > > Desarrollo de Negocios > > República 396 y Diego de Almagro > > Ph:+593.2.2526585 > > Cel:+593.9.4264354 > > Quito - Ecuador > > logoReducido > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > ------------------------------------------------------------------------ > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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[Serusers] blind call forwarding addition
by Martin Lane 01 Oct '09

01 Oct '09

SER Users: Thanks in advance for any help! I am running ser 0.9.6 and have a relatively simple cfg file. I have only four (4) digit extensions, no PSTN, no NAT and I do not care about authorization so my config file is basically the Hello World ser.cfg with my system parameters and a few statements in route that allowed me to create standing conference rooms and an echo application with sems. Now I need to implement call forwarding and I have been able to setup the MySQL ser usr_preferences table per chapter 10 of the 'SER - Getting Started' and the 'mySQL newbie? Problems with mySQL and SER?' documents. After adding the call forwarding functionality discussed in chapter 10 inside my config file any number dialed to a SIP device rings busy. All SIP devices register and can call the conference rooms or the echo application but they can not call each other. I will spare you my WireShark logs but can provide if that is needed. I'm looking for trouble shooting suggestions. Is there a way to print to the std i/o from inside the cfg file? This is the first time I needed to handle the INVITE message so I've included my INVITE Message Handler and the Call Forwarding Handler that I added for this effort: if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } if (method=="REGISTER") { route(2); break; }; ... route[3] { # ---------------------------------------------------------------------- # INVITE Message Handler # ---------------------------------------------------------------------- # Note: We are using this fuction only as a hook into the # blind call forwarding feature. Simply want to change the # R-URI and relay the message. if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); # Would love to do a printf here to see if this code is being hit!!! # Wireshark show INVITE messages are being sent. # debug/printf("\n\n****** Inside route(3) ********\n\n"); # At this point the a blind call forwarding record was found and the # new destination was written in the R-URI. # DEBUG: try just calling route(1). # route(1); # Send to Call Forwarding Handler route(6); break; }; route(1); } route[6] { # ---------------------------------------------------------------------- # Blind Call Forwarding Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ---------------------------------------------------------------------- lookup("aliases"); if (!is_uri_host_local()) { if (!isflagset(22)) { append_branch(); }; route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{4}@") { route(1); break; }; sl_send_reply("404", "User Not Found"); break; }; # DEBUG: There are no alias' and we are not sending calls to other # networks. We may just need to call route(1) and not this # function. route(1); }

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Re: [Kamailio-Users] Siremis JSON error
by Santiago Gimeno 01 Oct '09

01 Oct '09

Hi Luis, I ran into this problem too. In order to solve it I had to add a new Database element to the siremis/metadata/Config.xml file with the Name attribute set to Default and pointing to your openser database. I don't know the reason of having two Databases elements pointing to the same databases though. Best regards, Santi 2009/9/30 Luis Guaman <luis.guaman(a)interlancompu.com> > Hi, > > When we tried to run a Siremis Chart got the following error message: > > > > “Open Flash Chart > > > > JSON Parse Error [Syntax Error] > > Error at character 0, line 1: > > > > 0: <body style='font-family:Arial; font-size:12px; > background-color:#FCFCFC;'>Error m” > > > > Some help?? > > > > Atentamente, > > > > Luis Guamán > > Desarrollo de Negocios > > República 396 y Diego de Almagro > > Ph:+593.2.2526585 > > Cel:+593.9.4264354 > > Quito - Ecuador > > [image: logoReducido] > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] Book: SIP Security
by Daniel-Constantin Mierla 01 Oct '09

01 Oct '09

This book might be an interesting reading: http://by-miconda.blogspot.com/2009/09/book-sip-security.html Cheers, Daniel -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

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[Serusers] Quick question on pattern matching in the uri variable
by Frank Durda IV 30 Sep '09

30 Sep '09

The following piece of code is supposed to change a uri of 18885551212 to 8885551212 and change a uri of +18885551212 to 8885551212. The 18885551212 situation works fine, but in the +1 situation, the "if" test fails for some reason the code below falls to the "nothing to strip" log message. What am I missing? route[STRIP_LEADING_ONE] { if (method == "INVITE") { if (uri =~ "^sip:1.*" ) { #Leading 1? xlog("L_ERR", "Strip leading '1' - From URI:<%fu> From Tag:<%ft> Request's R-URI:<%ru> Before treatment To URI:<%tu>\n"); strip(1); } else if (uri =~ "^sip:+1.*" ) { #Leading +1? xlog("L_ERR", "Strip leading '+1' - From URI:<%fu> From Tag:<%ft> Request's R-URI:<%ru> Before treatment To URI:<%tu>\n"); strip(2); } else { xlog("L_ERR", "Nothing to Strip - From URI:<%fu> >From Tag:<%ft> Request's R-URI:<%ru> Before treatment To URI:<%tu>\n"); } } } Thanks in advance!

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Re: [Kamailio-Users] content about rtpproxy
by zhangchao00001 30 Sep '09

30 Sep '09

Hello everyone, i meet a problem when intergrating rtpproxy with kamailio. The two software start well, can communicate with each other, however, the rtpproxy always report "can't create listener" error. Dose anyone meet the same problem?

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Re: [Kamailio-Users] kamailio 1.4.3 slow
by Juan Perez 30 Sep '09

30 Sep '09

we finally found what was causing the prob, at least the prob is gone after doing this: In the syslog config file of the server where kamailio was working fine we found this entry: local0.none /var/log/messages we added it to the server with probs and problem solved. Since we were using the facility 0 to log the messages from kamailio apparently we need to add that entry listed above as well to make the logging work better. this is how we setup kamailio's logging, using the example from the web: # log messages with LOG_LOCAL0 in /var/log/ser.log local0.* -/var/log/proxy.log I do not know the reason why but it fixed our issue. I wanted to share with you. cheers and thank you a lot for your assistance jp ________________________________ From: Juan Perez <jperezsip2008(a)yahoo.com> To: Henning Westerholt <henning.westerholt(a)1und1.de>; users(a)lists.kamailio.org Cc: myoung(a)redmonsters.net Sent: Thursday, September 17, 2009 8:49:00 AM Subject: Re: [Kamailio-Users] kamailio 1.4.3 slow than you a lot Henning for your suggestions, I will try to do all of that but the lab stuff because we already have a lab setup but it works superb too, only on the new machine is when we have found the prob. in all our other scenarios kamailio runs super fast, processing a lot of calls per seconds and the cpu is 99-100 % idle. fortunately we can do this in the same machine because it has an amount of traffic controllable by us so at any given time we can take it out or put more or use sipp, I will post the results as soon as I have something new, thank you guys a lot again cheers jp ________________________________ From: Henning Westerholt <henning.westerholt(a)1und1.de> To: users(a)lists.kamailio.org Cc: Juan Perez <jperezsip2008(a)yahoo.com>; myoung(a)redmonsters.net Sent: Thursday, September 17, 2009 6:05:07 AM Subject: Re: [Kamailio-Users] kamailio 1.4.3 slow On Mittwoch, 16. September 2009, Juan Perez wrote: > In another scenario in our secondary site I am using another Dell Poweredge > 1950, with dual quad core almost at the same speed and 64 bits , same > version of CentOS and it is working like a charm. A lot of calls and CPU > idle almost 100% all the time. That is we decided to buy a more powerful > server to have a more robust platform in our main site but we haven't been > able to move forward with this because of the bad performance. Any other > ideas guys ? Hello Juan, as this looks more like a hardware issue to me i'd try to find issue that causes this particular server to be that slow. Somewhere it must spend this time. Some ideas: - Start kamailio with only one worker children, and then use strace to see what its doing when its hangs - enable debugging (only for a few seconds if you've production load on the system) and then look to the timestamps where its spend the time - use the benchmark module as Daniel already suggested - try to replicate the setup in a lab environment and then use a load tester (e.g. sipp) to give some traffic to the system. then change certain parts of the configuration and setup to find the culprit Cheers, Henning

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[Kamailio-Users] Siremis JSON error
by Luis Guaman 30 Sep '09

30 Sep '09

Hi, When we tried to run a Siremis Chart got the following error message: Open Flash Chart JSON Parse Error [Syntax Error] Error at character 0, line 1: 0: <body style='font-family:Arial; font-size:12px; background-color:#FCFCFC;'>Error m Some help?? Atentamente, Luis Guamán Desarrollo de Negocios República 396 y Diego de Almagro Ph:+593.2.2526585 Cel:+593.9.4264354 Quito - Ecuador logoReducido

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[Kamailio-Users] Kamailio not starting - no clue in syslog
by Anders 29 Sep '09

29 Sep '09

Hi, I get an error when trying to start Kamailio: ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I've searched around and found others with the same problem - but they usually found a lead to the problem in the syslog. All I get in the syslog when trying to start Kamailio is this line: Sep 28 05:34:01 server /USR/SBIN/CRON[10786]: (root) CMD (echo "call kamailio_cdrs();"|/usr/bin/mysql -u xxx -pyyy kamailio >/dev/null) Any ideas where and how to look? Thanks!

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[Kamailio-Users] DNS lookup for requests within dialog
by catalina oancea 28 Sep '09

28 Sep '09

Hi all, I tested the following scenario: astreisk kamailio provider(hostname) --------------------> dns lookup --------------------> REGISTER 1 REGISTER 1 <-------------------- <-------------------- 401 401 --------------------> dns lookup --------------------> REGISTER 2 REGISTER 2 (same dialog, increased cseq) It seems that kamailio does dns lookup again when receiving REGISTER 2, although both registers are part of the same dialog. Is this the expected behavior? Thanks, Catalina

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[Kamailio-Users] add_path_received and OBP problems
by Florian Meister 28 Sep '09

28 Sep '09

Hi, I'm trying to set up a HA-OBP for both natted and non-natted users. Atm my setup looks like that: [UAC] <---> (NAT) <---> [LB] <---> [REG + Proxy] <---> [PSTN GW] At the loadbalancer I use add_path and add_path_received to add a path-header to the request sent to the Registrar. (Of course only for register requests). I use a few parameters for telling the registrar module to use the path header. When looking in the mysql-db I can see the correct value in the path-field in the location-table. When getting an invite from the PSTN gateway for one of the UAC's the Proxy is adding a route-header according to the path-value in the mysql. That seems also correct. Now to my problem: The loadbalancer is sending out the INVITE to the private-ip, not to the public-nat-ip of the UAC. It seems that the received-parameter in the Route-header is somehow ignored. I am using the use_received parameter in the loadbalancer, but it does not work. In the following you can find the SIP trace (made on the loadbalancer) and the loadbalancer config, since I think there is the problem. Thanks, Florian Here is the SIP-trace of the INVITE: 09:49:07.785294 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 1101 E..i..@.?.Yb..,F..,+.....U.BINVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 69 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [PSTN Gateway] t=0 0 m=audio 19178 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 09:49:07.786459 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 445 E.....@.@.Z...,+..,F........SIP/2.0 100 Giving a try Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar] Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 09:49:07.786608 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:08.258116 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:09.258109 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:11.258107 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:15.023388 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 397 E.....@.?.\"..,F..,+......a.CANCEL sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To: <sip:+435572949012@[Proxy+Registrar]> CSeq: 102 CANCEL Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 09:49:15.024138 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 413 E.....@.@.[...,+..,F........SIP/2.0 200 canceling Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar] From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To: <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87 CSeq: 102 CANCEL Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 09:49:15.024576 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 493 E.. ..@.@.Z...,+..,F........SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar] Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 09:49:15.025180 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 433 E.....@.?.[...,F..,+.......MACK sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To: <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87 CSeq: 102 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 ----------------This is the config of my loadbalancer--------------------: ####### Global Parameters ######### debug=100 log_stderror=yes log_facility=LOG_LOCAL0 fork=no children=4 disable_tcp=yes port=5060 listen=[Loadbalancer] ####### Modules Section ######## mpath="/usr/lib/kamailio/modules/" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "nathelper.so" loadmodule "dispatcher.so" loadmodule "path.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs #modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #modparam("rr", "append_fromtag", 0) # -- nathelper modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy/rtpproxy.sock") modparam("nathelper", "received_avp", "$avp(i:80)") # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") # -- path params -- modparam("path", "use_received", 1) ####### Routing Logic ######## # main request routing logic route{ xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { xlog("L_ALERT", "_____HASTOTAG [$rm] from [$fu] to [$tu]\n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_ALERT", "_____LOOSE_ROUTE [$rm] from [$fu] to [$tu]\n"); t_relay(); exit; } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("L_ALERT", "_____NON-LOOSE, STATEFUL-ACK [$rm] from [$fu] to [$tu]\n"); # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { xlog("L_ALERT", "_____ACK_WITHOUT_TRANSACTION [$rm] from [$fu] to [$tu]\n"); # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing if (!is_method("REGISTER|MESSAGE")) xlog("L_ALERT", "_____RECORD_ROUTING [$rm] from [$fu] to [$tu]\n"); record_route(); #if ($rU==NULL) { # # request with no Username in RURI # sl_send_reply("484","Address Incomplete"); # exit; #} if (check_route_param("nat=yes")) { setbflag(6); } #if (isflagset(5) || isbflagset(6)) { route(5); #} /* example how to enable some additional event routes */ if (is_method("INVITE")) { t_on_reply("1"); t_on_failure("1"); } if(!ds_is_from_list()) { ds_select_domain("0", "0"); } if (!t_relay()) { sl_reply_error(); } } # Caller NAT detection route route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { add_path_received(); fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } else { add_path(); } return; } # RTPProxy control route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } onreply_route[1] { xdbg("incoming reply\n"); #if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { if (status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { #if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { if (is_method("INVITE")) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } }

3 13

Re: [Kamailio-Users] won't start - no error found in syslog
by Florian Meister 28 Sep '09

28 Sep '09

> There's rwx for root on /var/run - and I'm on with root trying to start > it... What about the user/group in the config (kamailio.conf) file ? Regards, Florian

1 0

[Kamailio-Users] won't start - no error found in syslog
by Anders 28 Sep '09

28 Sep '09

Hi, I get an error when trying to start Kamailio: ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I've searched around and found others with the same problem - but they usually found a lead to the problem in the syslog. All I get in the syslog when trying to start Kamailio is this line: Sep 28 05:34:01 server /USR/SBIN/CRON[10786]: (root) CMD (echo "call kamailio_cdrs();"|/usr/bin/mysql -u xxx -pyyy kamailio >/dev/null) Any ideas where and how to look? Thanks!

2 1

Re: [Kamailio-Users] content about kamailio start error
by zhangchao00001 27 Sep '09

27 Sep '09

Dear sir: I get the error during during processing of SIP message. The config file i used is below (the config works well on other kamailio server). By the way , the email system i used dosen't support:all the time please keep CC-ing to mailing list address ormaybe iadd the mailing list address in a wrong place, mails with a cc-ing added will be posted back. is it CC:users@lists.kamailio.org? debug=3 fork=yes log_stderror=yes port=5060 ####### Modules Section ######## mpath="/usr/local/lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "domain.so" loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 0) modparam("registrar", "method_filtering", 1) modparam("registrar", "append_branches", 0) modparam("registrar", "max_contacts", 10) modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 0) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("domain", "db_url", "mysql://root:123456@localhost/SIP") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://root:123456@localhost/SIP") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://root:123456@localhost/SIP") modparam("auth_db", "load_credentials", "") modparam("domain", "db_url", "mysql://root:123456@localhost/SIP") modparam("domain", "db_mode", 1) modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) modparam("presence|presence_xml", "db_url", "mysql://root:123456@localhost/SIP") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:26.28.29.253:5060") modparam("nathelper", "rtpproxy_sock", "udp:26.28.249.253:7722") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "rtpproxy_tout", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) modparam("path", "use_received", 1) ####### Routing Logic ######## route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(msg:len>=2048) { sl_send_reply("513","Message too big"); exit; } route(4); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); if(!uri==myself) { append_hf("P-hint: outbound\r\n"); if(is_method("REGISTER|MESSAGE")) { if(is_method("REGISTER")) { if(!add_path_received()) { sl_send_reply("503","Internal Path Error!"); exit(); } } } else { record_route(); } route(1); } # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } if (!lookup("location")) { switch ($retcode) { 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 setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { 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; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) /*if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; }*/ } else { # authenticate if from local subscriber (uncomment to enable auth) /*if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($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 }*/ } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } if (t_check_status("486|408")) { sethostport("26.28.249.125:5060"); append_branch(); # do not set the missed call flag again t_relay(); } } 在2009-09-25，"Daniel-Constantin Mierla" <miconda(a)gmail.com> 写道： >Hello, > >do keep mailing list CC-ed for every mail, otherwise I will stop replying. > >On 25.09.2009 3:33 Uhr, zhangchao00001 wrote: >> Dear sir: >> thank you for your reply. >> The Kamailio -V result : >> version: kamailio 1.5.2-notls (i386/linux) >> flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, >> SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> svnrevision: unknown >> @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ >> main.c compiled on 10:23:04 Sep 23 2009 with gcc 4.3.0 >> when there's no user, it reports no error, when a UAC register, it >> reports the error below: >> Sep 25 09:24:21 [3323] ERROR:core:build_req_buf_from_sip_req: out of >> pkg memory >> Sep 25 09:24:21 [3323] ERROR:tm:print_uac_request: no more shm_mem >> Sep 25 09:24:21 [3323] ERROR:tm:t_forward_nonack: failure to add branches >> Sep 25 09:24:21 [3323] ERROR:tm:_reply_light: failed to allocate shmem >> buffer >> Sep 25 09:24:21 [3323] ERROR:tm:relay_reply: no more share memory >> what should i do to fix it? >does it happen after startup, during processing of SIP message? Is a >stress test? What is the list of loaded modules? If you can share the >config, would be good. > >Daniel > >-- >Daniel-Constantin Mierla >* Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >* http://www.asipto.com/index.php/sip-router-masterclass/ > 网易邮箱用户购物独享现金返还网易邮箱用户购物独享现金返还网易邮箱用户购物独享现金返还 "中国制造",讲述中国60年往事 "中国制造",讲述中国60年往事

1 0

[Kamailio-Users] Need Assistance Tracking Down A SIP Signaling Problem
by Geoffrey Mina 25 Sep '09

25 Sep '09

Hello, I am wondering if anyone can help me determine what the problem with my SIP signaling is here. I have attached two Excel files (can provide other formats if someone doesn't have Excel). The environments in both scenarios are almost identical... although I am dealing with two different carriers upstream. In my environment, I have Kamailio (1.5) sitting in front of a multitude of Asterisk machines. I am using the dispatcher module to distribute INVITE requests across the network. I am doing some interop with a new carrier and we are struggling a bit with some looping scenarios. They are sending invites to my Kamailio server, I am forwarding to asterisk. The two XLS files (Success.xls) and (Fail.xls) have sip_trace information for the transactions. As you can see in the Fail.xls, the ACK to the 200 OK gets looped back to the Kamailio server and never ends up reaching the Asterisk machine. In the Success.xls, this doesn't happen. The only thing I can see that's different is the Success carrier appears to be replacing the RURI in the ACK with the value of the Contact header... in the failed one, the RURI of the ACK is the IP of my Kamailio gateway. I have read through the RFC 3261 to try and figure out what is breaking this, but haven't been able to. I basically need to determine if the signalling is correct... what is wrong with my config... and if the signalling is wrong, what the carrier is doing incorrectly. Thanks in advance! Geoff

4 9

[Kamailio-Users] Diameter
by Ghaith ALKAYYEM 24 Sep '09

24 Sep '09

Hello list, Does anybody have any idea or recommendation about integrating diameter with SER? As far as you know the available module is obsolete, i'm trying to implement the module "CDP" available with OpenIMS: http://www.openimscore.org/docs/ser_ims/CDP.html Thank you

2 1

[Kamailio-Users] Unable to lookup host in c= line,
by Tincho ylm 24 Sep '09

24 Sep '09

Hi all! I'm trying to make a realtime environment openser-asterisk. At this moment, everything is working fine but I have problems with VM feature. What I'm doing is: A call B If B is busy and returns 486 Openser send this to Asterisk. failure_route[1] { if(!t_was_cancelled()) { if (t_check_status("487|486")) { rewritehostport("200.xx.xx.xx:5060"); append_branch(); xlog("L_INFO","---> To Voicemail...\n"); setflag(10); route(1); exit; } if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } } Asterisk cant handle it because this error: chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' I'm using: Kamailio 1.4.4 Rtpproxy Asterisk 1.4.22 Is it a rtpproxy issue? it can fix it? If I make a ethereal capture, I can see in the MESSAGE BODY this: connection inrfomation (c): IN IP4 200.xx.xx.xx200.xx.xx.xx What can I do? thanks in advance.

2 2

[Serusers] Search trough status line for specific reason phrase
by Josip Djuricic 23 Sep '09

23 Sep '09

Dear, is there any way to search inside Status-Line of response message? I have strange b2bua that always returns SIP 500 just with different reason phrase. I need to match on them so that I can rewrite failure messages. I thought about something like this: failure_route[1] { if (t_check_status("500")) { #here I need to search trough status line for specific reasons, but have no idea how to do that if (search("^Specific message")) { t_reply("480", "Temporarily Unavailable"); break; } } } Thank you J.

1 0

[Kamailio-Users] H323 - SIP Signalling gateway for Kamailio
by General Lee 23 Sep '09

23 Sep '09

Hi, I'm looking for a H323 - SIP signalling gateway which I can use to bridge video calls between my H323 Gatekeeper and Kamailio system. Has anyone done anything like this before? Are you able to suggest any suitable h323 - SIP video gateway solutions I could use? Thanks -- View this message in context: http://www.nabble.com/H323---SIP-Signalling-gateway-for-Kamailio-tp25530990… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 3

[Serusers] SER as a stateless proxy server
by Muhammad Ali 23 Sep '09

23 Sep '09

Hi, How SER can be configured as a stateless proxy server? Best regards M Ali

1 0

[Kamailio-Users] content about kamailio start error
by zhangchao00001 23 Sep '09

23 Sep '09

Hello everyone, i have some questions about kamailio. After i install rtpproxy 0.3version, the kamailio server can't work any more. The following is the error message. Dose any one know the reason? dose the rtpproxy 0.3 change something that kamailio need? Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module registrar Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module textops Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module uri_db Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module siputils Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module xlog Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module acc Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Sep 23 11:07:57 [4925] ERROR:core:build_req_buf_from_sip_req: out of pkg memory Sep 23 11:07:57 [4925] ERROR:tm:print_uac_request: no more shm_mem Sep 23 11:07:57 [4925] ERROR:tm:t_forward_nonack: failure to add branches Sep 23 11:07:57 [4925] ERROR:tm:_reply_light: failed to allocate shmem buffer Sep 23 11:07:57 [4925] ERROR:tm:relay_reply: no more share memory Sep 23 11:07:57 [4925] ERROR:tm:_reply_light: failed to allocate shmem buffer

2 1

[Kamailio-Users] Comparing AVPs to user variables
by Alex Balashov 23 Sep '09

23 Sep '09

Is there a known bug in Kamailio 1.4.1 that causes comparisons like this to be inaccurate? get_profile_size("people", "$avp(s:person_id)", "$var(u_call_count)"); ... if($var(u_call_count) >= $avp(s:call_limit)) In this case, $var(u_call_count) was 3 and $avp(s:call_limit) was 26 - I printed both values beforehand. Yet, the comparison was evaluated as true. I worked around the problem by storing the profile size in an AVP as well and comparing them instead, and it seems to have solved the problem: if($avp(s:u_call_count) >= $avp(s:call_limit)) However, I was wondering if this is a known problem in 1.4.1 and possibly 1.4.x. Thanks! -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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[Kamailio-Users] Stupid softswitch sending a CANCEL with "To tag" in parallel forking
by Iñaki Baz Castillo 22 Sep '09

22 Sep '09

Hi, I've experimenting an absurd issue when a softswitch calls to Kamailio and receives two 180 (with different To tags) due to parallel forking made by Kamailio. When the softswitch receives the second 180 (with the second To tag) it sends a ridiculous CANCEL with To tag matching the second To tag received (yes, a CANCEL with To tag !!!). Of course Kamailio replies "403 Why do you send a CANCEL with To tag? please read the RFC 3261". Unfortunatelly the softswitch didn't go to the school and ignores the 403 by repeating the CANCEL. And the worst, when *any* of the ringing phones answerw, the softswitch ignores the 200 so the call is never established. IMHO there is nothing I can do to "fix" this stupid issue. I've tried to reply 200 for the CANCEL but the softswitch is not happy (it seems that it expects the 487 from the canceled branch). And of course, if I do "t_relay()" for the CANCEL then Kamailio cancels both branches (the correct behavior). The only hyper exotic "solution" I can imagine is the following: - Kamailio receives the wrong CANCEL with To tag "2". - Reply 200 for the CANCEL to make the softswitch happy. - Reply 487 with To tag "2" to make the softswitch happy (IMHO this is not possible). - Generate a spoofed BYE for the second branch (that with To tag "2"). Of course, this is not possible with dialog module as it doesn't handle parallel forking. Any suggestion? (yes, I would like to kill the softswitch). Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

[Kamailio-Users] new feature: update sip msg content in request route
by Daniel-Constantin Mierla 21 Sep '09

21 Sep '09

Hello, I just committed a new function in modules_k/textops - msg_apply_changes(). Hopefully the name is suggestive enough - the function applies the changes done to the content of sip message - e.g., if you added a new header, once you use the function, that header will become visible in config file. Therefore the function tries to overcome the overhead of looping various messages back to sip server just because modifications were not visible once done. Another benefit is ability to access latest value for a field (being header or body content) in case of need to hashes or signatures for security purposes. The function can be used only in request route for now. Be carefully when using it, since it changes the expectation you had so far of always working on initially received sip message. Once the function is sued, initial content is lost forever. Since it builds a new buffer and re-parses, it is not very recommended to use it extensively. Example of usage: append_hf("My-Header: yes\r\n"); if(msg_apply_changes()) { # msg buffer has a new content if(is_present_hf("My-Header")) { # will get always here } } Testing and feedback is welcome! Cheers, Daniel

2 1

[Kamailio-Users] questions about TLS
by zhangchao00001 21 Sep '09

21 Sep '09

Hello everybody: I've have some questions about tls connection. How to build the tls connection between two proxy? there's just one function is_peer_verified() in tlsops module. Is it that we set the paramer, the proxy will connect by itself or we should build the connection in the config file explicitly? could someone give me a example? Proxy1<----TLS--->Proxy2

2 1

[Kamailio-Users] room does not exists in IMC
by Rajesh 21 Sep '09

21 Sep '09

Dear Experts, I have been trying to configure IMC for quite some time now..when ti try to create the chatroom (using my xlite phone), using the command #create q, i get the following error. *ERROR:imc:imc_handle_message: room [q] does not exist! Sep 17 10:25:36 [2562] ERROR:imc:imc_manager: failed to handle 'message' * Iam posting my cfg below.. Your sujjestion would be of immense help to me.. # # $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "imc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr 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) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* 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", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") #-------tm params----- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) #-------imc params----- modparam("imc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("imc", "hash_size", 8) modparam("imc", "imc_cmd_start_char", "#") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } #if(uri=~"sip:q.*@") if(uri=~"sip:q@192.168.64.75") { # IMC - message xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { 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 setflag(2); route(1); } route[1] { /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($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 } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); } failure_route[1] { if (t_was_cancelled()) { exit; } } - Regards, Rajesh

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[Kamailio-Users] allow_address and "Error while converting IP address string to in_addr"
by Marcello Lupo 21 Sep '09

21 Sep '09

Hi to all, i'm using openser-1.2.2 and i'm getting the error: "Error while converting IP address string to in_addr" while using the allow_address function. I tried both with: allow_address("$(ru{uri.host})", "$rp") allow_address("$rd", "$rp") I made debugging printng content of both $rd and $(ru{uri.host}) and both return the correct value. Don't know anymore where to search about it. Any help? Thank you, regards, Marcello

2 4

registration error:500 - Server Internal Error
by Rakhmadhany Primananda 19 Sep '09

19 Sep '09

when i create account in x-lite,then i found error code like this: Registration error:500 - Server Internal Error i'm using SIP server with SER 0.9.6. what's the problem with x-lite? thanks,,

1 0

[Kamailio-Users] PCRE / Regex Module
by Brandon Armstead 18 Sep '09

18 Sep '09

Hello, I have the following pcre regex method call: pcre_match("$hdr(From)", "(?i)^(restricted|anonymous|private|unknown|unavailable)") to match calls that are sent without a proper ANI in a case-insensitive manner, however it does not seem to return true for the following example From Header: >From Header: "Unavailable" <sip:Restricted@192.168.47.68<sip%3ARestricted(a)192.168.47.68> >;tag=gK07327e23 Any thoughts / input is appreciated, and as always thank you for your help! Sincerely, Brandon Armstead

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Re: [Kamailio-Users] Kamailio speed dial
by Elena-Ramona Modroiu 17 Sep '09

17 Sep '09

Hi, please do not forget to CC the list. Your xlog statements print $tu - to header uri, not $ru, request-uri, which is the one updated by speed dial. Make the update and if you still get same result either is an error printed by sd_lookup() in log or the table is not populated properly. Send over the result of the mysql query: select * from speed_dial where sd_username='51' Regards, Ramona Mythili D wrote: > Hi, Good morning....I have seen your mail..Thank you very much for > your reply.. > I have followed your suggestions. > > if(uri=~"sip:[0-9]{2}@.*"){ > > xlog("L_INFO","mylog: test1.\n"); > > xlog("L_INFO", "SIP Request: method [$rm] from [$fu] to [$tu]\n"); > > #sd_lookup("speed_dial", "sip:$au@$fd"); > > sd_lookup("speed_dial"); > > xlog("L_INFO","mylog: test2.\n"); > > xlog("L_INFO", "SIP Request: method [$rm] from [$fu] to [$tu]\n"); > > } > > I called fromm 1002 to 51. > Then i got the output as : > mylog test1. > SIP Request: method [INVITE] from [sip:1002@192.168.64.94 > <mailto:sip%3A1002@192.168.64.94>] to [sip:51@192.168.64.94 > <mailto:sip%3A51@192.168.64.94>] > mylog: test2. > SIP Request: method [INVITE] from [sip:1002@192.168.64.94 > <mailto:sip%3A1002@192.168.64.94>] to [sip:51@192.168.64.94 > <mailto:sip%3A51@192.168.64.94> > > > Here, see the control goes to the function sd_lookup("speed_dial") and > it must take the value 5001 instead of 51 and sholud make call to 5001. > and after mylog:test2 message, it sholud return the message with OK. > But see it is not like that. So, I think, error might be with the > sd_lookup(" "). Please clarify my doubt. > > Thanks.... > Regards, > Mythili. > On Mon, Sep 14, 2009 at 2:27 PM, Elena-Ramona Modroiu > <ramona(a)asipto.com <mailto:ramona@asipto.com>> wrote: > > Hi, > > I see in one config you use $au@$fd for lookup, that is wrong > unless you have the two digits in username of auth header. Better > use sd_lookup() without second parameter. > > Then print $ru in xlog() before and after sd_lookup() -- that will > reveal what is looked up and if the function changed the r-uri. > > Regards, > Ramona > > > Mythili D wrote: > > Hiii...I installed kamailio 1.5.1 and mysql 5.0.37 properly. > Authnetication is working properly. Now i'm going to configure > the feature "speed dial". > I tried with two config files. I'm facing the same problem > with two config files.. > 1st config file > ----------------------------------------------------------- > # > # $Id: speeddial.cfg 4585 2008-08-06 08:20:30Z klaus_darilion $ > # > # sample config script to use speeddial module > # > debug=3 > log_stderror=yes > log_facility=LOG_LOCAL0 > fork=yes > children=4 > # ----------- global configuration parameters > ------------------------ > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > port=5060 > # ------------------ module loading > ---------------------------------- > loadmodule "/usr/local/lib/kamailio/modules/sl.so" > loadmodule "/usr/local/lib/kamailio/modules/tm.so" > loadmodule "/usr/local/lib/kamailio/modules/rr.so" > loadmodule "/usr/local/lib/kamailio/modules/maxfwd.so" > loadmodule "/usr/local/lib/kamailio/modules/usrloc.so" > loadmodule "/usr/local/lib/kamailio/modules/registrar.so" > loadmodule "/usr/local/lib/kamailio/modules/textops.so" > loadmodule "/usr/local/lib/kamailio/modules/db_mysql.so" > loadmodule "/usr/local/lib/kamailio/modules/speeddial.so" > loadmodule "/usr/local/lib/kamailio/modules/mi_fifo.so" > loadmodule "/usr/local/lib/kamailio/modules/xlog.so" > # ----------------- setting module-specific parameters > --------------- > modparam("speeddial", "db_url", > "mysql://openser:openserrw@localhost/openser") > modparam("speeddial", "use_domain", 1) > # -- mi_fifo params -- > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") > # -- usrloc params -- > modparam("usrloc", "db_mode", 0) > # -- rr params -- > # add value to ;lr param to make some broken UAs happy > modparam("rr", "enable_full_lr", 1) > # ------------------------- request routing logic > ------------------- > # main routing logic > route{ > # initial sanity checks > if (!mf_process_maxfwd_header("10")) > { > sl_send_reply("483","Too Many Hops"); > exit; > }; > if (msg:len >= max_len ) > { > sl_send_reply("513", "Message too big"); > exit; > }; > if (!method=="REGISTER") record_route(); > if (loose_route()) > { > if (!t_relay()) > { > sl_reply_error(); > }; > exit; > }; > if (!uri==myself) > { > if (!t_relay()) > { > sl_reply_error(); > }; > exit; > }; > if (uri==myself) > { > if (method=="REGISTER") > { > save("location"); > exit; > }; > if(uri=~"sip:[0-9]{2}@.*"){ > xlog("L_INFO","mylog: test1.\n"); > sd_lookup("speed_dial"); > #sd_lookup("speed_dial", "sip:$au@$fd"); > xlog("L_INFO","mylog: test2.\n"); > } > lookup("aliases"); > if (!uri==myself) > { > if (!t_relay()) > { > sl_reply_error(); > }; > exit; > }; > if (!lookup("location")) > { > sl_send_reply("404", "Not Found"); > exit; > }; > }; > if (!t_relay()) > { > sl_reply_error(); > }; > } > -------------------------------------------------------------------- > I'm using the table speed_dial. My data in table is > username=5001 > domain=192.168.64.94 > sd_username=51 > sd_domain=192.168.64.94 > I dialled 51 from useragent 1001. the output is not found. > but it is displaying test1 and test2 messages from my cfg file > which i gave, i.e., it is entring into loop and function > sd_lookup("speed_dial"). But the problem with speed_dial i > think so... > Please help me...... > 2nd config file > --------------------------------------------------------------------- > # > # $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $ > # > # Kamailio (OpenSER) SIP Server - basic configuration script > # - web: http://www.kamailio.org <http://www.kamailio.org/> > # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ > # > # Direct your questions about this file to: > <users(a)lists.kamailio.org <mailto:users@lists.kamailio.org> > <mailto:users@lists.kamailio.org > <mailto:users@lists.kamailio.org>>> > > # > # Refer to the Core CookBook at > http://www.kamailio.org/dokuwiki/doku.php > # for an explanation of possible statements, functions and > parameters. > # > # There are comments showing how to enable different features > in th econfig > # file. Such commented code starts with #X# where X is a > letter to identify > # a feature. Delete entire #X# if you want to enable that > feature. Next are > # sed commands that help you enable such features. > # > # *** To enamble mysql execute: > # sed -i 's///g' kamailio.cfg > # > # *** To enamble authentication execute: > # - enable mysql > # sed -i 's///g' kamailio.cfg > # - add users using 'kamctl' > # > # *** To enamble persistent user location execute: > # - enable mysql > # sed -i 's/#u#//g' kamailio.cfg > # > # *** To enamble presence server execute: > # - enable mysql > # sed -i 's/#p#//g' kamailio.cfg > # > # *** To enamble nat traversal execute: > # sed -i 's/#n#//g' kamailio.cfg > # - install RTPProxy: http://www.rtpproxy.org > <http://www.rtpproxy.org/> > # - start RTPProxy: > # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 > # > # *** To enhance accounting execute: > # - enable mysql > # sed -i 's/#c#//g' kamailio.cfg > # - add following columns to database > # 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 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 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 ''; > # > ####### Global Parameters ######### > debug=3 > log_stderror=yes > log_facility=LOG_LOCAL0 > fork=yes > children=4 > /* uncomment the following lines to enable debugging */ > #debug=6 > #fork=no > #log_stderror=yes > /* uncomment the next line to disable TCP (default on) */ > #disable_tcp=yes > /* uncomment the next line to enable the auto temporary > blacklisting of > not available destinations (default disabled) */ > #disable_dns_blacklist=no > /* uncomment the next line to enable IPv6 lookup after IPv4 dns > lookup failures (default disabled) */ > #dns_try_ipv6=yes > /* uncomment the next line to disable the auto discovery of > local aliases > based on revers DNS on IPs (default on) */ > #auto_aliases=no > /* uncomment the following lines to enable TLS support > (default off) */ > #disable_tls = no > #listen = tls:your_IP:5061 > #tls_verify_server = 1 > #tls_verify_client = 1 > #tls_require_client_certificate = 0 > #tls_method = TLSv1 > #tls_certificate = > "/usr/local/etc/kamailio/tls/user/user-cert.pem" > #tls_private_key = > "/usr/local/etc/kamailio/tls/user/user-privkey.pem" > #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" > port=5070 > /* 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:192.168.1.2:5060 <http://192.168.1.2:5060/> > <http://192.168.1.2:5060 <http://192.168.1.2:5060/>> > > ####### Modules Section ######## > #set module path > mpath="/usr/local/lib/kamailio/modules/" > /* uncomment next line for MySQL DB support */ > loadmodule "db_mysql.so" > loadmodule "mi_fifo.so" > loadmodule "sl.so" > loadmodule "tm.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "usrloc.so" > loadmodule "registrar.so" > loadmodule "textops.so" > loadmodule "uri_db.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "acc.so" > loadmodule "speeddial.so" > /* uncomment next lines for MySQL based authentication support > NOTE: a DB (like db_mysql) module must be also loaded */ > loadmodule "auth.so" > loadmodule "auth_db.so" > /* uncomment next line for aliases support > NOTE: a DB (like db_mysql) module must be also loaded */ > #loadmodule "alias_db.so" > /* uncomment next line for multi-domain support > NOTE: a DB (like db_mysql) module must be also loaded > NOTE: be sure and enable multi-domain support in all used modules > (see "multi-module params" section ) */ > #loadmodule "domain.so" > /* uncomment the next two lines for presence server support > NOTE: a DB (like db_mysql) module must be also loaded */ > #p#loadmodule "presence.so" > #p#loadmodule "presence_xml.so" > #n#loadmodule "nathelper.so" > # ----------------- setting module-specific parameters > --------------- > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > # ----- rr params ----- > # add value to ;lr param to cope with most of the UAs > modparam("rr", "enable_full_lr", 1) > # do not append from tag to the RR (no need for this script) > modparam("rr", "append_fromtag", 0) > # ----- rr 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) > # ----- uri_db params ----- > /* by default we disable the DB support in the module as we do > not need it > in this configuration */ > modparam("uri_db", "use_uri_table", 0) > modparam("uri_db", "db_url", "") > # ----- acc params ----- > /* what sepcial events should be accounted ? */ > modparam("acc", "early_media", 1) > modparam("acc", "report_ack", 1) > modparam("acc", "report_cancels", 1) > /* 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", "failed_transaction_flag", 3) > modparam("acc", "log_flag", 1) > modparam("acc", "log_missed_flag", 2) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > /* uncomment the following lines to enable DB accounting also */ > #c#modparam("acc", "db_flag", 1) > #c#modparam("acc", "db_missed_flag", 2) > #c#modparam("acc", "db_url", > #c# "mysql://openser:openserrw@localhost/openser") > #c#modparam("acc", "db_extra", > #c# > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > # ----- usrloc params ----- > /* uncomment the following lines if you want to enable DB > persistency > for location entries */ > #u#modparam("usrloc", "db_mode", 0) > #u#modparam("usrloc", "db_url", > #u# "mysql://openser:openserrw@localhost/openser") > # ----- auth_db params ----- > /* uncomment the following lines if you want to enable the DB > based > authentication */ > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "db_url", > "mysql://openser:openserrw@localhost/openser") > modparam("auth_db", "load_credentials", "") > # ----- alias_db params ----- > /* uncomment the following lines if you want to enable the DB > based > aliases */ > #modparam("alias_db", "db_url", > # "mysql://openser:openserrw@localhost/openser") > # ----- domain params ----- > /* uncomment the following lines to enable multi-domain detection > support */ > #modparam("domain", "db_url", > # "mysql://openser:openserrw@localhost/openser") > #modparam("domain", "db_mode", 1) # Use caching > # ----- multi-module params ----- > /* uncomment the following line if you want to enable > multi-domain support > in the modules (dafault off) */ > #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) > # ----- presence params ----- > /* uncomment the following lines if you want to enable presence */ > #p#modparam("presence|presence_xml", "db_url", > #p# "mysql://openser:openserrw@localhost/openser") > #p#modparam("presence_xml", "force_active", 1) > #p#modparam("presence", "server_address", > "sip:192.168.1.2:5060 <http://192.168.1.2:5060/> > <http://192.168.1.2:5060 <http://192.168.1.2:5060/>>") > # -- nathelper > #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722 > <http://127.0.0.1:7722/> <http://127.0.0.1:7722 > <http://127.0.0.1:7722/>>") > > #n#modparam("nathelper", "natping_interval", 30) > #n#modparam("nathelper", "ping_nated_only", 1) > #n#modparam("nathelper", "sipping_bflag", 7) > #n#modparam("nathelper", "sipping_from", > "sip:pinger@kamailio.org <mailto:sip%3Apinger@kamailio.org> > <mailto:sip%3Apinger@kamailio.org > <mailto:sip%253Apinger@kamailio.org>>") > > #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") > #n#modparam("usrloc", "nat_bflag", 6) > ####### Routing Logic ######## > # main request routing logic > route{ > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > } > # > # NAT detection > route(4); > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > if (is_method("BYE")) { > setflag(1); # do accounting ... > setflag(3); # ... even if the transaction fails > } > route(1); > } else { > if (is_method("SUBSCRIBE") && uri == myself) { > # in-dialog subscribe requests > route(2); > exit; > } > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # non loose-route, but stateful ACK; must be an ACK after a > 487 or e.g. 404 from upstream server > t_relay(); > exit; > } else { > # ACK without matching transaction ... ignore and discard.\n"); > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > #initial requests > # CANCEL processing > if (is_method("CANCEL")) > { > if (t_check_trans()) > t_relay(); > exit; > } > t_check_trans(); > # authentication > route(3); > # record routing > if (!is_method("REGISTER|MESSAGE")) > record_route(); > # account only INVITEs > if (is_method("INVITE")) { > setflag(1); # do accounting > } > if (!uri==myself) > /* replace with following line if multi-domain support is used */ > ##if (!is_uri_host_local()) > { > append_hf("P-hint: outbound\r\n"); > # if you have some interdomain connections via TLS > ##if($rd=="tls_domain1.net <http://tls_domain1.net/> > <http://tls_domain1.net <http://tls_domain1.net/>>") { > ## t_relay("tls:domain1.net <http://domain1.net/> > <http://domain1.net <http://domain1.net/>>"); > ## exit; > ##} else if($rd=="tls_domain2.net <http://tls_domain2.net/> > <http://tls_domain2.net <http://tls_domain2.net/>>") { > ## t_relay("tls:domain2.net <http://domain2.net/> > <http://domain2.net <http://domain2.net/>>"); > > ## exit; > ##} > route(1); > } > # requests for my domain > if( is_method("PUBLISH|SUBSCRIBE")) > route(2); > if (is_method("REGISTER")) > { > if (!save("location")) > sl_reply_error(); > exit; > } > if ($rU==NULL) { > # request with no Username in RURI > sl_send_reply("484","Address Incomplete"); > exit; > } > # apply DB based aliases (uncomment to enable) > ##alias_db_lookup("dbaliases"); > if(uri=~"sip:[0-9]{2}@.*"){ > xlog("L_INFO","mylog: test1.\n"); > sd_lookup("speed_dial", "sip:$au@$fd"); > #sd_lookup("speed_dial"); > xlog("L_INFO","mylog: test2.\n"); > } > if (!lookup("location")) { > switch ($retcode) { > 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 > setflag(2); > route(1); > } > route[1] { > #n# if (check_route_param("nat=yes")) { > #n# setbflag(6); > #n# } > #n# if (isflagset(5) || isbflagset(6)) { > #n# route(5); > #n# } > /* example how to enable some additional event routes */ > if (is_method("INVITE")) { > #t_on_branch("1"); > t_on_reply("1"); > t_on_failure("1"); > } > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > # Presence route > /* uncomment the whole following route for enabling presence > server */ > route[2] > { > #p# if (!t_newtran()) > #p# { > #p# sl_reply_error(); > #p# exit; > #p# }; > #p# > #p# if(is_method("PUBLISH")) > #p# { > #p# handle_publish(); > #p# t_release(); > #p# } > #p# else > #p# if( is_method("SUBSCRIBE")) > #p# { > #p# handle_subscribe(); > #p# t_release(); > #p# } > #p# exit; > > # if presence enabled, this part will not be executed > if (is_method("PUBLISH") || $rU==null) > { > sl_send_reply("404", "Not here"); > exit; > } > return; > } > # Authentication route > /* uncomment the whole following route for enabling > authentication */ > route[3] { > if (is_method("REGISTER")) > { > # authenticate the REGISTER requests (uncomment to enable auth) > if (!www_authorize("", "subscriber")) > { > www_challenge("", "0"); > exit; > } > if ($au!=$tU) > { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > # authenticate if from local subscriber (uncomment to enable auth) > if (from_uri==myself) > { > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > exit; > } > if (is_method("PUBLISH")) > { > if ($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 > } > } > return; > } > # Caller NAT detection route > /* uncomment the whole following route for enabling Caller NAT > Detection */ > route[4]{ > #n# force_rport(); > #n# if (nat_uac_test("19")) { > #n# if (method=="REGISTER") { > #n# fix_nated_register(); > #n# } else { > #n# fix_nated_contact(); > #n# } > #n# setflag(5); > #n# } > return; > } > # RTPProxy control > /* uncomment the whole following route for enabling RTPProxy > Control */ > route[5] { > #n# if (is_method("BYE")) { > #n# unforce_rtp_proxy(); > #n# } else if (is_method("INVITE")){ > #n# force_rtp_proxy(); > #n# } > #n# if (!has_totag()) add_rr_param(";nat=yes"); > return; > } > branch_route[1] { > xdbg("new branch at $ru\n"); > } > onreply_route[1] { > xdbg("incoming reply\n"); > #n# if ((isflagset(5) || isbflagset(6)) && > status=~"(183)|(2[0-9][0-9])") { > #n# force_rtp_proxy(); > #n# } > #n# if (isbflagset(6)) { > #n# fix_nated_contact(); > #n# } > } > failure_route[1] { > #n# if (is_method("INVITE") > #n# && (isbflagset(6) || isflagset(5))) { > #n# unforce_rtp_proxy(); > #n# } > if (t_was_cancelled()) { > 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 > <http://192.168.2.100:5060/> <http://192.168.2.100:5060 > <http://192.168.2.100:5060/>>"); > > ## append_branch(); > ## # do not set the missed call flag again > ## t_relay(); > ##} > } > ------------------------------------------------------------------- > Here i'm using the table speed_dial. My data in table is > username=5001 > domain=192.168.64.94 > sd_username=51 > sd_domain=192.168.64.94 > I dialled 51 from useragent 1001. the output is not found. > but it is displaying test1 and test2 messages from my cfg file > which i gave, i.e., it is entring into loop and function > sd_lookup("speed_dial"). But the problem with speed_dial i > think so > Problem is same in two files... > Please help me... > Thanks in advance.... > regards, > Mythili. > > ------------------------------------------------------------------------ > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > >

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Re: [Kamailio-Users] RTPproxy pre-filling callee // media address
by Nick Lemberger 17 Sep '09

17 Sep '09

I thought those flags/ip address settings was the change the SDP part of the SIP message. I tried your suggestion anyway, but it didn't work. What I'm really looking for is how to change what Kamailio is telling RTPProxy via the socket. -Nick >>> On 9/17/2009 at 8:31 AM, in message <a950b2030909170631s24d5dab4pd989c7a2e5846315(a)mail.gmail.com>, Laszlo <laszlo(a)voipfreak.net> wrote: > 2009/9/17 Nick Lemberger <Nick.Lemberger(a)lkfd.net> > >> Sone background: >> I'm trying to use Kamailio as a multi-homed proxy in front of a softswitch, >> one NIC has a public IP, the other a private. The softswitch is in the >> private network and all SIP messaging goes through a 'virtual IP' on the >> softswitch which allows for statefull failover between the softswitch CPUs. >> The media on the other had goes to an address stated in the SDP message. >> >> I don't have any problem with the SIP messaging, it flows as expected. >> Also, the RTP stream from my UAC to Kamailio/RTPproxy is working as >> expected as well (I can record audio on this leg). >> >> There is a problem when RTP proxy sets up the leg on the internal side of >> the network, I see the following in the logs: >> Sep 17 07:56:36 voip.lakefield.net rtpproxy[20388]: [ID 408714 mail.info] >> INFO:handle_command: pre-filling callee's address with >> 192.168.249.40:33020 >> >> I'm wondering what variable Kamailio gets the 192.168.249.40 address from - >> this is the virtual IP for SIP messaging. In the SDP body of those messages >> from the softswitch, I see that it's supposed to use 192.168.249.8 as the >> media IP (and I see that it's setting up the correct port information, >> source port of 33020 in this instance, but with the wrong IP). Where can I >> tell Kamailio what IP should be telling RTPProxy? >> >> Thanks, >> -Nick >> <http://lists.openser-project.org/cgi-bin/mailman/listinfo/users> > > > you can try using force_rtp_proxy with IP and some flags( o and/or c) > > http://www.kamailio.org/docs/modules/1.5.x/nathelper.html#id2468157 > > -Laszlo

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[Kamailio-Users] RTPproxy pre-filling callee // media address
by Nick Lemberger 17 Sep '09

17 Sep '09

Sone background: I'm trying to use Kamailio as a multi-homed proxy in front of a softswitch, one NIC has a public IP, the other a private. The softswitch is in the private network and all SIP messaging goes through a 'virtual IP' on the softswitch which allows for statefull failover between the softswitch CPUs. The media on the other had goes to an address stated in the SDP message. I don't have any problem with the SIP messaging, it flows as expected. Also, the RTP stream from my UAC to Kamailio/RTPproxy is working as expected as well (I can record audio on this leg). There is a problem when RTP proxy sets up the leg on the internal side of the network, I see the following in the logs: Sep 17 07:56:36 voip.lakefield.net rtpproxy[20388]: [ID 408714 mail.info] INFO:handle_command: pre-filling callee's address with 192.168.249.40:33020 I'm wondering what variable Kamailio gets the 192.168.249.40 address from - this is the virtual IP for SIP messaging. In the SDP body of those messages from the softswitch, I see that it's supposed to use 192.168.249.8 as the media IP (and I see that it's setting up the correct port information, source port of 33020 in this instance, but with the wrong IP). Where can I tell Kamailio what IP should be telling RTPProxy? Thanks, -Nick

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[Kamailio-Users] kamailio 1.4.3 slow
by Juan Perez 17 Sep '09

17 Sep '09

hello I installed kamailio in a pc with dual quad core @ 3.3GHz, 4 GB RAM and see that kamailio is very slow answering to INVITEs from the inbound carriers. The DEBUG is set to 3 and the idle CPU lowers only to 87% or 90 %, which is still decent but anyways the performance is very bad. This is the first time that I see this, I have it installed in other machines with less resources, same OS [CentOS 5.2 64 bits] and it is performing very very good. I know that a high debug level is a enemy but in my case is set to 3 like in the other system, and it is using the same configuration file. What other thing may affect the performance? I have made sure that it is not a network issue because the captures are taken in the same machine and I see multiples INVITEs hitting it and no response at all until more than 2 seconds I would appreciate any help or suggestions cheers jp

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[Serusers] ser 0.9.8 bugfix release
by Andrei Pelinescu-Onciul 17 Sep '09

17 Sep '09

There is a new bugfix release for the old 0.9.x ser (if you are using a 2.x version or sip-router, please disregard this message). The release is made from the top of the rel_0_9_0 branch (so if you already have a ser version compiled from the top of rel_0_9_0 in the last 10 month you don't need to upgrade). It contains 3 more fixes when compared to 0.9.7: - tm: fix delete_cell() when the transaction is referenced - core: fix memory leak in subst_user - acc: misspeled function name and wrong check You can either get it from cvs (v_0_9_8 tag or the top of rel_0_9_0 branch) or you can download a tarball with the sources: http://ftp.iptel.org/pub/ser/0.9.8/src/ser-0.9.8_src.tar.gz. There are no binary packages provided, only sources. Upgrading from older 0.9.x versions should only involve re-compiling and re-installing ser. There are no changes in the config file format or in the DB schemes. Andrei

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[Serusers] HowTo add manually ContactURI to registered user
by Evgeny Miloslavsky 17 Sep '09

17 Sep '09

Hi All. I have sip:ABCD@SomeDomain.com user registered successfully at SER. Now I want manually to add another ContactURI to the same user. Is there any possibility to do so? Thanks in advance Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 Office: 972-74-7170072

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[Kamailio-Users] sip_resolvehost: NATPR query timed out
by Joao Gomes Pereira 16 Sep '09

16 Sep '09

Hello Im getting this error in my Kamailio-1.5.2: Sep 16 15:43:08 sip kamailio[25298]: ERROR:core:sip_resolvehost: NATPR query timed out, no server could be reached, stop resolving (Connection timed out) Sep 16 15:43:08 sip kamailio[25298]: CRITICAL:core:mk_proxy: could not resolve hostname: "sip.server.pt" Sep 16 15:43:08 sip kamailio[25298]: ERROR:tm:uri2proxy: bad host name in URI <sip:sip.server.pt> Sep 16 15:43:08 sip kamailio[25298]: ERROR:tm:t_forward_nonack: failure to add branches Why cant Kamailio resolve the name? Could this be causing my clients SIP registrations to fail? Thanks regards Joao Pereira

2 4

[Kamailio-Users] avp_db_query to SELECT a DOUBLE colomn
by BERGANZ François 16 Sep '09

16 Sep '09

Hello, I need help. I try to select a DOUBLE(10,4). Kamailio report <null>. Example: avp_db_query("select CREDIT from subscriber where client='$avp(s:679)'", "$avp(i:513)"); $avp(i:513) is null! But in my database, it is 15.0000 Know you how to do ? Cordialement, BERGANZ François cid:image001.gif@01C8F7CD.6BC1D2C0 <http://www.acropolistelecom.net/> http://www.acropolistelecom.net P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] Job Offer: VoIP System Administrator
by Andreas Granig 15 Sep '09

15 Sep '09

Sipwise is currently hiring a VoIP System Administrator for an interesting position based in Vienna/Austria. You have strong skills in Linux system administration (Monitoring and Alerting using SNMP/Nagios/Cacti/MRTG, Scripting in Perl/M4/sh), a deep understanding of highly available system deployments and good knowledge regarding SIP (preferably Kamailio, Sems, Asterisk)? We offer you a challenging position to help our team further improving our Kamailio-based Class5 Softswitches, deploying them at customer sites and supporting our customers (large DSL and Cable Providers throughout Europe) and sales teams with technical details. If you are interested, please send me an email for further information. Best regards, Andreas

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[Kamailio-Users] SIP Router Development Meeting - Oct 2, 2009, Berlin, Germany
by Daniel-Constantin Mierla 15 Sep '09

15 Sep '09

Hello, I want to introduce SIP Router Development Meeting 2009 planned for Friday, October 2 in Berlin, Germany. More details about the event at: http://sip-router.org/2009/09/14/development-meeting-2009/ Shortly, the goal of the meeting is to analyze the evolution so far, discuss the plans for the future and socialize within the community. The integration between SER and Kamailio is 99% done (only seas module needs some work), first major release based on SIP router core is due in about one month: Kamailio 3.0, today being the date of going into testing phase. Participation is free for anybody upon registration via email at: registration(a)lists.sip-router.org Feel free to submit suggestions for discussion topics, presentations and event extensions. A hacking session is already proposed, now just show your interest. For general questions about the meeting (and/or hacking session), please address them to: sr-dev(a)lists.sip-router.org See you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] Kamailio speed dial
by Mythili D 14 Sep '09

14 Sep '09

Hiii...I installed kamailio 1.5.1 and mysql 5.0.37 properly. Authnetication is working properly. Now i'm going to configure the feature "speed dial". I tried with two config files. I'm facing the same problem with two config files.. 1st config file ----------------------------------------------------------- # # $Id: speeddial.cfg 4585 2008-08-06 08:20:30Z klaus_darilion $ # # sample config script to use speeddial module # debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 # ----------- global configuration parameters ------------------------ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/kamailio/modules/sl.so" loadmodule "/usr/local/lib/kamailio/modules/tm.so" loadmodule "/usr/local/lib/kamailio/modules/rr.so" loadmodule "/usr/local/lib/kamailio/modules/maxfwd.so" loadmodule "/usr/local/lib/kamailio/modules/usrloc.so" loadmodule "/usr/local/lib/kamailio/modules/registrar.so" loadmodule "/usr/local/lib/kamailio/modules/textops.so" loadmodule "/usr/local/lib/kamailio/modules/db_mysql.so" loadmodule "/usr/local/lib/kamailio/modules/speeddial.so" loadmodule "/usr/local/lib/kamailio/modules/mi_fifo.so" loadmodule "/usr/local/lib/kamailio/modules/xlog.so" # ----------------- setting module-specific parameters --------------- modparam("speeddial", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("speeddial", "use_domain", 1) # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (!uri==myself) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if(uri=~"sip:[0-9]{2}@.*"){ xlog("L_INFO","mylog: test1.\n"); sd_lookup("speed_dial"); #sd_lookup("speed_dial", "sip:$au@$fd"); xlog("L_INFO","mylog: test2.\n"); } lookup("aliases"); if (!uri==myself) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; }; if (!t_relay()) { sl_reply_error(); }; } -------------------------------------------------------------------- I'm using the table speed_dial. My data in table is username=5001 domain=192.168.64.94 sd_username=51 sd_domain=192.168.64.94 I dialled 51 from useragent 1001. the output is not found. but it is displaying test1 and test2 messages from my cfg file which i gave, i.e., it is entring into loop and function sd_lookup("speed_dial"). But the problem with speed_dial i think so... Please help me...... 2nd config file --------------------------------------------------------------------- # # $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's///g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's///g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # 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 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 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 ''; # ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5070 /* 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:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "speeddial.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr 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) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* 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", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("acc", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ #u#modparam("usrloc", "db_mode", 0) #u#modparam("usrloc", "db_url", #u# "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if(uri=~"sip:[0-9]{2}@.*"){ xlog("L_INFO","mylog: test1.\n"); sd_lookup("speed_dial", "sip:$au@$fd"); #sd_lookup("speed_dial"); xlog("L_INFO","mylog: test2.\n"); } if (!lookup("location")) { switch ($retcode) { 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 setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($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 } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE") #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { 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(); ##} } ------------------------------------------------------------------- Here i'm using the table speed_dial. My data in table is username=5001 domain=192.168.64.94 sd_username=51 sd_domain=192.168.64.94 I dialled 51 from useragent 1001. the output is not found. but it is displaying test1 and test2 messages from my cfg file which i gave, i.e., it is entring into loop and function sd_lookup("speed_dial"). But the problem with speed_dial i think so Problem is same in two files... Please help me... Thanks in advance.... regards, Mythili.

2 1

[Kamailio-Users] Does forward() DNS SRV failover?
by Arturo Díaz Almagro 14 Sep '09

14 Sep '09

Hello, I am using Kamailio 1.4.1 and I have to relay REGISTER messages to several other proxies (2). In one case I need to forward all request/responses and in the other case I don not need to propagate the responses. In the first case I am using forward() and in the other case t_replicate(). What I have realized from my test is that forward() does not perform DNS SRV failover as t_relay does, but if I use t_relay, the t_replicate does not work. Is that a bug in forward()? Is there any other way to relay messages, in the case explained above? Thanks -- Arturo Díaz

2 1

[Kamailio-Users] passing parameters in cfg for IMC
by Rajesh 14 Sep '09

14 Sep '09

Dear All, Iam trying to configure IMC. I find that different parameters are passed in the sample cfg and in the README. I would like to know if the following two parameters are essential inorder to configure IMC or can i run IMC without passing the two parameters. modparam("imc", "rooms_table", "rooms") modparam("imc", "rooms_table", "members") I would also like to know the use of these parameters. Thanks Rajesh

2 1

[Kamailio-Users] monioring and alert
by ALAEDDINE abbech 11 Sep '09

11 Sep '09

Hello all, I want have an alert when number of simultaneous call in kamailio/openser reach 200call/second. Who have an idea to do this. regards, alasupcom

3 3

[Kamailio-Users] new module: number portability functionality
by Henning Westerholt 11 Sep '09

11 Sep '09

Hi all, i just commited several parts related to number portability functionality to the sr repository. This can be used as a source for routing informations related to the ownership of the number. This functionality consists of several parts. 1. pdb server This server loads serialized routing data from the disk and stores it in memory. It then listens on an UDP Port for requests containing a number and returns ID of the carrier which owns the number. This server uses the same datastructure as the carrierroute module and provides a really good performance. According our experience it usually consumes only a few percent CPU, even you use only one server for your complete callrouting cluster. 2. pdb connector module This module connects the sip-router server to the pdb server. It supports load-balancing and aggressive timeouts. Normally it not need more than a few ms to query the remote server and return the reply to the configuration script. 3. pdb tool (data compiler) This tool provides the functionality to compile the carrier and number informations into the binary data format the pdb server expect. It supports optimizing the generated trie structure, so that for example the complete number to carrier mapping for germany (app. 150 million numbers) don't need more than a few hundred megabytes. You can also combine not interesting carriers in order to save even more space and get better performance. You find the module in the modules/pdb directory, the server and tool is in utils/pdbt. This directory also contains documentation (README for module, utils/pdbt/docs/* for data format and network protocol). We decided to develop a custom solution as none of the alternatives that were evaluated (like enum over DNS, databases e.g. berkeley DB, some commercial products) provides the wanted performance, failure tolerance and storage efficieny. This module has been in production use since some time, so if you find some bugs they are most probably introduced from me because of the sr adaption. The original implementation was done from Hardy Kahl. Best regards, Henning Westerholt -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[Kamailio-Users] Kamailio 1.5.1 + WeSip
by Chandrakant Solanki 11 Sep '09

11 Sep '09

Hello I have install kamailio-1.5.1-tls_src now i would like to install WeSip... How can I ... and can I installed WeSip into same PC in which I installed kamailio... -- Regards, Chandrakant Solanki

2 1

[Kamailio-Users] error message ...
by Rajesh 11 Sep '09

11 Sep '09

IAm trying to configure IMC in kamailio.I created two tables in the database named, imc_rooms and imc_members.IN my xlite phone i give the command q:sip@192.168.64.75 ('q' is used as the default chatroom name in .cfg). when i give the command , # create room_name, i get the following error, ERROR:imc:imc_handle_message: room [q] does not exist! Sep 9 12:28:25 [2632] ERROR:imc:imc_manager: failed to handle 'message' Kindly guide me.. Thanks

3 6

[Kamailio-Users] kamailio speed dial
by mythili ram 11 Sep '09

11 Sep '09

Hiii...I installed kamailio 1.5.1 and mysql 5.0.37 properly. Authnetication is working properly. Now i'm going to configure the feature "speed dial". I tried with two config files. I'm facing the same problem with two config files.. 1st config file ----------------------------------------------------------- # # $Id: speeddial.cfg 4585 2008-08-06 08:20:30Z klaus_darilion $ # # sample config script to use speeddial module # debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 # ----------- global configuration parameters ------------------------ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/kamailio/modules/sl.so" loadmodule "/usr/local/lib/kamailio/modules/tm.so" loadmodule "/usr/local/lib/kamailio/modules/rr.so" loadmodule "/usr/local/lib/kamailio/modules/maxfwd.so" loadmodule "/usr/local/lib/kamailio/modules/usrloc.so" loadmodule "/usr/local/lib/kamailio/modules/registrar.so" loadmodule "/usr/local/lib/kamailio/modules/textops.so" loadmodule "/usr/local/lib/kamailio/modules/db_mysql.so" loadmodule "/usr/local/lib/kamailio/modules/speeddial.so" loadmodule "/usr/local/lib/kamailio/modules/mi_fifo.so" loadmodule "/usr/local/lib/kamailio/modules/xlog.so" # ----------------- setting module-specific parameters --------------- modparam("speeddial", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("speeddial", "use_domain", 1) # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (!uri==myself) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if(uri=~"sip:[0-9]{2}@.*"){ xlog("L_INFO","mylog: test1.\n"); sd_lookup("speed_dial"); #sd_lookup("speed_dial", "sip:$au@$fd"); xlog("L_INFO","mylog: test2.\n"); } lookup("aliases"); if (!uri==myself) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; }; if (!t_relay()) { sl_reply_error(); }; } -------------------------------------------------------------------- I'm using the table speed_dial. My data in table is username=5001 domain=192.168.64.94 sd_username=51 sd_domain=192.168.64.94 I dialled 51 from useragent 1001. the output is not found. but it is displaying test1 and test2 messages from my cfg file which i gave, i.e., it is entring into loop and function sd_lookup("speed_dial"). But the problem with speed_dial i think so... Please help me...... 2nd config file --------------------------------------------------------------------- # # $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's///g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's///g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # 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 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 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 ''; # ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5070 /* 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:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "speeddial.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr 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) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* 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", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("acc", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ #u#modparam("usrloc", "db_mode", 0) #u#modparam("usrloc", "db_url", #u# "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if(uri=~"sip:[0-9]{2}@.*"){ xlog("L_INFO","mylog: test1.\n"); sd_lookup("speed_dial", "sip:$au@$fd"); #sd_lookup("speed_dial"); xlog("L_INFO","mylog: test2.\n"); } if (!lookup("location")) { switch ($retcode) { 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 setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($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 } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE") #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { 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(); ##} } ------------------------------------------------------------------- Here i'm using the table speed_dial. My data in table is username=5001 domain=192.168.64.94 sd_username=51 sd_domain=192.168.64.94 I dialled 51 from useragent 1001. the output is not found. but it is displaying test1 and test2 messages from my cfg file which i gave, i.e., it is entring into loop and function sd_lookup("speed_dial"). But the problem with speed_dial i think so Problem is same in two files... Please help me... Thanks in advance.... regards, Mythili. _________________________________________________________________ News, sports, entertainment and fine living…learn the ropes on MSN India http://in.msn.com

1 0

[Kamailio-Users] room does not exits
by Rajesh 11 Sep '09

11 Sep '09

Dear All, I am configurig IMC and i am curently facing an issue. When i send a mesage to the charroom named q (the default nmae is q), i get an error message which says, " ERROR:imc:imc_handle_message: room [q] does not exist! ERROR:imc:imc_manager: failed to handle 'message' " i created two tables in my database.Iam sure that iam doing something silly which is invoking this error message . Kindly guide me with my cfg and let me know if I need to make any changes in my cfg. Iam posting my cfg below. -- ****************************************************************************************************************************************** # # $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's///g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's///g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # 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 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 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 ''; # ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "imc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr 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) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* 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", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("acc", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ #u#modparam("usrloc", "db_mode", 2) #u#modparam("usrloc", "db_url", #u# "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #-------tm params----- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) #-------imc params----- modparam("imc", "db_url", "mysql://openser:openserrw@192.168.64.75/openser") #the below lines are additionally added for imc #modparam("imc", "rooms_table", "imc_rooms") #modparam("imc", "rooms_table", "imc_members") modparam("imc", "hash_size", 8) modparam("imc", "imc_cmd_start_char", "#") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if(uri=~"sip:q.*@") #if(uri=~"sip:q@192.168.64.75") { # IMC - message xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { 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 setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($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 } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE") #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { 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(); ##} } Thanks

1 0

[Kamailio-Users] monitoring tool
by Chandrakant Solanki 11 Sep '09

11 Sep '09

Hello I want to know that .... Is there any monitoring tool for kamailio .. which shows user details numbers of user connected @ that time... which automatically updated every 2/5 minutes... I knew about siremis... is there any other tools..??? -- Regards, Chandrakant Solanki

2 1

[Kamailio-Users] Textops / Search / Regex
by Brandon Armstead 10 Sep '09

10 Sep '09

Hello, I currently have the following search method use: search("(restricted|anonymous|Anonymous|Private|private|Unknown|unknown)") unfortunately this is catching cases where there is a Diversion header i.e. Diversion: < sip:1234@someHost > ; reason = unknown So I attempted to do an assertion: search("(restricted|anonymous|Anonymous|Private|private|Unknown|(?<!reason=)unknown)") However it does not seem that Kamailio is completely PCRE compatible? thus server fail.... / parse error Does anyone have a solution so I can keep the same matching, except exclude matching "unknown|Unknown" when reason=unknown, however match it in all other conditions? Thank you for any and all help ahead of time. Sincerely, Brandon

2 2

[Kamailio-Users] tls and notls kamailio
by Mark Ryden 10 Sep '09

10 Sep '09

Hello, I am trying to build kamailio from source. I am a newbie to kamailio. In the download area of 1.5.2, http://www.kamailio.org/pub/kamailio/1.5.2/src/, there are two files: kamailio-1.5.2-notls_src.tar.gz 3.3M kamailio-1.5.2-tls_src.tar.gz 3.4M I have a question: what is the difference, in short, between these two tar balls ? Regards, Mark

2 1

[Serusers] Trouble with Record-Route
by Frank Durda IV 10 Sep '09

10 Sep '09

I have a situation where SER sits between two independent networks, one being an isolated network and the other being the rest of the Internet. The isolated network is also a PSTN gateway with only one switch on that network. SER has two physical interfaces, and uses force_send_socket and rewritehost to make sure things go out the right interfaces. The problem is when the called network hangs up first and generates a BYE* first, if SER didn't add a record-route, the called network switch bypasses SER and attempts to send directly to the calling IP address, which doesn't work because the two networks cannot be reached directly. If I add just one record-route in SER, the called network now knows how to get back to SER to tell it about the BYE, and SER passes that BYE on to the calling network. So far so good. Unfortunately, this also causes the called network to add Record-Route to responses it sends out, like 183/200 with its own IP address. Those reach the SER without problem, but SER doesn't add its own Record-Route to the reply, so when this reaches the calling network, they only have a Record-route that points to an IP address they cannot possibly reach. I tried a number of things to get around this, and all failed. 1. Tried to add my own Record-Route to the reply, but SER doesn't allow this. Won't even start. 2. Tried to remove the Record-Route from the reply, but SER also doesn't allow this. Won't even start. 3. Tried to add both IP addresses for the two interfaces to by using two Record-Routes at the INVITE. This doesn't fix the response messages, and actually causes looping behavior or "I'm terribly sorry" responses. (I tried this with and without the "enable_double_rr" parameter set and it didn't seem to make any difference.) By altering the order of the two Record-Route you are pretty-much guaranteed to confuse one party or the other as each needs the Record-Routes listed in the opposite order. 4. Tried some combinations of the above with loose_route() but the usual outcome is SER sending messages back to itself rather than to the right place and eventually the dreaded "I'm terribly sorry" or "481 Transaction" gets sent out, usually to nobody in particular. So, are there any working examples out there for a SER that has two network interfaces, each in its own isolated network that correctly route messages between the two sides? I need SER to not only add Record-Routes when appropriate but to strip them off or replace them as the replies pass back through and back to the initiator. The network isolation was a security thing, not a NAT thing (although private IP space is in use), but it is not possible for a message from one network to reach the other without passing through SER, and some commercial equipment I am working with ignores the source of a message and only sends the response to wherever the record-route(s) say it should go and if no record-route, it honors the Via, while other gear ignored both and only replies to the source of the message. *The problems affecting called-party BYEs also causes major grief when re-INVITEs occur, which fail to make it to the calling network (or more likely, the 200 OK replies fail to make it back), causing the call to die after 32 seconds. Any examples would be greatly appreciated.

3 2

[Kamailio-Users] One Way Audio problem
by Chandrakant Solanki 10 Sep '09

10 Sep '09

hello I am using kamailio with asterisk as gateway... I found one way audio problem... Please find attached log on pastbin... http://pastebin.com/f25e3f34a Here is my configuration file listen = sipstaging.testing.com port = 5060 mpath = "/lib/kamailio/modules" #children = 8 debug = 7 fork = yes #group = "openser" #user = "openser" #disable_tcp = no #log_facility = LOG_DAEMON log_facility = LOG_LOCAL0 log_stderror = no #tcp_children = 4 #mhomed = no #server_signature = yes #sock_group = "openser" #sock_mode = 0600 #sock_user = "openser" #unix_sock = "/tmp/openser.sock" #unix_sock_children = 1 #reply_to_via = no #sip_warning = no #check_via = no dns = no rev_dns = no #disable_core_dump = no #dns_try_ipv6 = yes #dns_use_search_list = yes #Chandrakant #loadmodule "exec.so" #loadmodule "presence.so" #modparam("presence", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") #modparam("presence", "force_active", 1 ); #modparam("presence", "max_expires", 3600 ) #modparam("presence", "server_address", "sip: voicemailserver.testingpbx.com:5060" ) #loadmodule "presence_mwi.so" loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "pv.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 3) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 3) modparam("tm", "fr_inv_timer", 35) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) #modparam("tm", "noisy_ctimer", 1) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) modparam("tm", "fr_inv_timer_avp", "$avp(s:callee_fr_inv_timer)") loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") #modparam("mi_fifo", "fifo_mode", 0660) #modparam("mi_fifo", "fifo_group", "openser") #modparam("mi_fifo", "fifo_user", "openser") #modparam("mi_fifo", "reply_dir", "/tmp/") #modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "textops.so" #loadmodule "uri.so" loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "db_mysql.so" modparam("db_mysql", "ping_interval", 300) modparam("db_mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 1) modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=uuid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") #modparam("uri_db", "db_table", "uri") #modparam("uri_db", "user_column", "username") #modparam("uri_db", "domain_column", "domain") #modparam("uri_db", "uriuser_column", "uri_user") #modparam("uri_db", "subscriber_table", "subscriber") #modparam("uri_db", "subscriber_user_column", "username") #modparam("uri_db", "subscriber_domain_column", "domain") #modparam("uri_db", "use_uri_table", 0) #modparam("uri_db", "use_domain", 1) loadmodule "avpops.so" modparam("avpops", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 1) modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") loadmodule "lcr.so" modparam("lcr", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "gw_uri_avp", "$avp(i:1400)") modparam("lcr", "ruri_user_avp", "$avp(i:1402)") #modparam("lcr", "contact_avp", "$avp(1401)") modparam("lcr", "fr_inv_timer_avp", "$avp(s:fr_inv_timer_avp)") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "$avp(s:rpid)") loadmodule "alias_db.so" modparam("alias_db", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") modparam("alias_db", "use_domain", 0) loadmodule "uac.so" modparam("uac", "rr_store_param", "vsf") modparam("uac", "from_restore_mode", "auto") modparam("uac", "from_passwd", "s1p:Wiz4rd!") loadmodule "acc.so" modparam("acc", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") # flag to record to db modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # flag to log to syslog modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) # use extra accounting to record caller and callee username/domain # - take them from From URI and R-URI modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") #loadmodule "nat_traversal.so" ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO","***** IN route[0]\n"); xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); xlog("L_INFO","***** OUT route[0.1]\n"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); xlog("L_INFO","***** OUT route[0.2]\n"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } # allow NOTIFY messages only from the Asterisk server # if (is_method("NOTIFY")) { # if (!t_relay()) { # sl_reply_error(); # exit; # } # } if (is_method("SUBSCRIBE")) { # challenge/response if (!www_authorize("", "subscriber")) { www_challenge("", "0"); xlog("L_INFO","***** OUT route[0.3]\n"); exit; } # notify the Asterisk server to update the UAs MWI. I # believe most UAs that support SUBSCRIBE will # periodically send a SUBSCRIBE message. These cause # the Asterisk server to update their MWI. # xlog("L_INFO","******************************** $ru\n" ); # exec_msg("/usr/bin/mwi/subscribe $ru"); exit; xlog("L_INFO","***** OUT route[0.4]\n"); } if(is_method("CANCEL") || is_method("BYE")) { unforce_rtp_proxy(); } if(loose_route()) { # if(!has_totag()) # { # # xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # sl_send_reply("403", "Initial Loose-Routing Rejected"); # exit; # } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(8); } if(is_method("REGISTER")) { route(7); } if(is_method("INVITE")) { route(9); } if(is_method("CANCEL") || is_method("ACK")) { route(13); } route(14); xlog("L_INFO","***** OUT route[0]\n"); } ######################################################################## # Request route 'clear-usr-preferences-caller' ######################################################################## route[1] { xlog("L_INFO","***** IN route[1]\n"); xlog("L_INFO", "Clear caller preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:caller_cli)/g"); avp_delete("$avp(s:clir)/g"); xlog("L_INFO","***** OUT route[1]\n"); } ######################################################################## # Request route 'clear-usr-preferences-callee' ######################################################################## route[2] { xlog("L_INFO","***** IN route[2]\n"); xlog("L_INFO", "Clear callee preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:callee_fr_inv_timer)/g"); avp_delete("$avp(s:cfu)/g"); avp_delete("$avp(s:cfc)/g"); xlog("L_INFO","***** OUT route[2]\n"); } ######################################################################## # Request route 'usr-preferences-caller' ######################################################################## route[3] { xlog("L_INFO","***** IN route[3]\n"); route(1); xlog("L_INFO", "Load caller preferences for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load caller avps avp_db_load("$avp(s:caller_uuid)", "*"); avp_copy("$avp(s:cli)", "$avp(s:caller_cli)/d"); if(is_avp_set("$avp(s:clir)/n") && avp_check("$avp(s:clir)", "eq/i:1")) { # mark for anonymization setflag(28); } xlog("L_INFO","***** OUT route[3]\n"); } ######################################################################## # Request route 'usr-preferences-callee' ######################################################################## route[4] { xlog("L_INFO","***** IN route[4]\n"); xlog("L_INFO", "Load callee preferences for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load callee avps avp_db_load("$avp(s:callee_uuid)", "*"); if(is_avp_set("$avp(s:cfu)/s")) { xlog("L_INFO", "Call-forward-unconditional to '$avp(s:cfu)' found - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:caller_uuid)/g"); avp_copy("$avp(s:callee_uuid)", "$avp(s:caller_uuid)/d"); avp_pushto("$ru", "$avp(s:cfu)"); route(3); route(10); xlog("L_INFO","***** OUT route[4.1]\n"); exit; } if(is_avp_set("$avp(s:ringtimeout)/n")) { xlog("L_INFO", "Setting ring timeout to $avp(s:ringtimeout) secs - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_copy("$avp(s:ringtimeout)", "$avp(s:callee_fr_inv_timer)/d"); } xlog("L_INFO","***** OUT route[4]\n"); } ######################################################################## # Request route 'clir' ######################################################################## route[5] { xlog("L_INFO","***** IN route[5]\n"); if(isflagset(28) && !isflagset(27)) { setflag(27); xlog("L_INFO", "Anonymize caller - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); uac_replace_from("Anonymous","sip:anonymous@anonymous.invalid"); if(is_present_hf("Privacy")) { remove_hf("Privacy"); } append_hf("Privacy: id\r\n"); } xlog("L_INFO","***** OUT route[5]\n"); } ######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[6] { xlog("L_INFO","***** IN route[6]\n"); if(isflagset(22)) { unforce_rtp_proxy(); } xlog("L_INFO","***** OUT route[6]\n"); } ######################################################################## # Request route 'base-route-register' ######################################################################## route[7] { xlog("L_INFO","***** IN route[7]\n"); sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); xlog("L_INFO","***** OUT route[7.1]\n"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); xlog("L_INFO","***** OUT route[7.2]\n"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); xlog("L_INFO","***** OUT route[7.3]\n"); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_INFO","***** OUT route[7]\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[8] { xlog("L_INFO","***** IN route[8]\n"); if(is_present_hf("P-Asserted-Identity")) { remove_hf("P-Asserted-Identity"); } if(is_present_hf("Remote-Party-ID")) { remove_hf("Remote-Party-ID"); } if(is_avp_set("$avp(s:caller_cli)/s")) { if(!isflagset(28)) { xlog("L_INFO", "Set caller CLI '$avp(s:caller_cli)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); append_hf("P-Asserted-Identity: <$avp(s:caller_cli)>\r\n"); } } route(5); if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { unforce_rtp_proxy(); } } xlog("L_INFO","***** OUT route[8]\n"); exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[9] { xlog("L_INFO","***** IN route[9]\n"); sl_send_reply("100", "Trying"); if(from_gw()) { $avp(s:caller_uuid)="0"; xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); $avp(s:ringtimeout)=1000; route(3); if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(51); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); xlog("L_INFO","***** OUT route[9.1]\n"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); xlog("L_INFO","***** OUT route[9.2]\n"); exit; } route(3); if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(10); } # route(3); # if(nat_uac_test("19")) # { # fix_nated_contact(); # setbflag(6); # } # route(10); xlog("L_INFO","***** OUT route[9]\n"); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[10] { xlog("L_INFO","***** IN route[10]\n"); if(alias_db_lookup("dbaliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route(2); if(!is_domain_local("$rd")) { setflag(20); $avp(s:callee_uuid) = "0"; route(12); } avp_delete("$avp(s:callee_uuid)"); avp_db_query("select uuid from subscriber where username = '$rU'", "$avp(s:callee_uuid)"); if(is_avp_set("$avp(s:callee_uuid)/s")) { xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); } else { $avp(s:callee_uuid) = "0"; xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_INFO", ">>>>>>>>>>>>>>>>>>>>LENGTH - STORED = $var(X) RURI=$rU len=$(rU{s.len}) \n"); if($rU=="*86") { route(12); } else { if($(rU{s.len})<=7) { route(11); } else { route(12); } } } xlog("L_INFO","***** OUT route[10]\n"); exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[11] { xlog("L_INFO","***** IN route[11]\n"); route(4); revert_uri(); seturi("sip:*89@openser.org"); rewritehostport("voicemailserver.testingpbx.com:5060"); route(8); #if(!lookup("location")) #{ # xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # sl_send_reply("404", "User Offline"); # route(50); #} #else #{ # xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # #route(8); # route(50); #} xlog("L_INFO","***** OUT route[11]\n"); exit; } ############################## # TEST ROUTE 51 ########################### route[51] { if(alias_db_lookup("dbaliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route(2); if(!is_domain_local("$rd")) { setflag(20); $avp(s:callee_uuid) = "0"; route(12); } avp_delete("$avp(s:callee_uuid)"); avp_db_query("select uuid from subscriber where username = '$rU'", "$avp(s:callee_uuid)"); if(is_avp_set("$avp(s:callee_uuid)/s")) { xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(52); } else { $avp(s:callee_uuid) = "0"; xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(12); } exit; } ########################## #TEST 52 ############################## route[52] { route(4); if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(8); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[12] { xlog("L_INFO","***** IN route[12]\n"); if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(8); xlog("L_INFO","***** OUT route[12.1]\n"); exit; } if(!isflagset(23)) { if(uri =~ "^sip:\*86@.*") { xlog("L_INFO", "User accessing Voicemail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); rewritehostport("voicemailserver.testingpbx.com:5060"); route(8); } # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded else if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); xlog("L_INFO","***** OUT route[12.2]\n"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); xlog("L_INFO","***** OUT route[12.3]\n"); exit; } setflag(21); t_on_failure("1"); route(8); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); xlog("L_INFO","***** OUT route[12]\n"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[13] { xlog("L_INFO","***** IN route[13]\n"); t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_INFO","***** OUT route[13.1]\n"); exit; } if(!t_relay()) { sl_reply_error(); } xlog("L_INFO","***** OUT route[13]\n"); exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[14] { xlog("L_INFO","***** IN route[14]\n"); xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); xlog("L_INFO","***** OUT route[14]\n"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[15] { xlog("L_INFO","***** IN route[15]\n"); if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); xlog("L_INFO","***** OUT route[15.1]\n"); exit; } xlog("L_INFO","***** OUT route[15]\n"); } ######################################################################## #Send the call out ######################################################################## route[16] { xlog("L_INFO","***** IN route[16]\n"); #send the call outward if(method=="INVITE" && !isflagset(10)) { t_on_failure("2"); }; if (!t_relay()) { xlog("L_WARN", "ERROR: t_relay failed"); sl_reply_error(); }; xlog("L_INFO","***** OUT route[16]\n"); } ######################################################################## # Request route 'Voicemail and Asterisk for further processing' ######################################################################## failure_route[3] { xlog("L_INFO","***** IN failure route[3]\n"); if(!t_was_cancelled()) { revert_uri(); rewritehostport("voicemailserver.testingpbx.com:5060"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "No Answer Sending the call to Voicemail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(10); route(8); } xlog("L_INFO","***** OUT failure route[3]\n"); } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO","***** IN onreply route[1]\n"); xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_INFO","***** OUT onreply route[1]\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO","***** IN onreply route[2]\n"); xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } xlog("L_INFO","***** OUT onreply route[2]\n"); exit; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO","***** IN failure route[1]\n"); xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(15); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); xlog("L_INFO","***** OUT failure route[1.1]\n"); exit; } t_on_failure("3"); route(8); xlog("L_INFO","***** OUT failure route[1]\n"); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { xlog("L_INFO","***** IN failure route[2]\n"); if(!t_was_cancelled()) { revert_uri(); seturi("sip:*87@openser.org"); rewritehostport("voicemailserver.testingpbx.com:5060"); append_branch(); xlog("L_INFO","Call to Voicemail"); setflag(10); route(10); } if(t_check_status("422|481|487")) { xlog("L_INFO", "Final reply - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); xlog("L_INFO","***** OUT failure route[2.1]\n"); exit; } t_on_branch("1"); xlog("L_INFO", "Redirect from UAC intercepted - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); if($avp(s:cfc) != NULL) { avp_pushto("$ru", "$avp(s:cfc)"); setflag(29); append_branch(); t_on_branch("1"); xlog("L_INFO", "CFC detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); } route(15); route(6); xlog("L_INFO","***** OUT failure route[2]\n"); } ######################################################################## # Branch route 'cfc-drop-local' ######################################################################## branch_route[1] { xlog("L_INFO","***** IN branch route[1]\n"); if(is_domain_local("$rd")) { xlog("L_INFO", "Dropping local branch - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); drop(); } xlog("L_INFO","***** OUT branch route[1]\n"); } -- Regards, Chandrakant Solanki

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[Kamailio-Users] How Kamailio Generate INVITE
by ALAEDDINE abbech 10 Sep '09

10 Sep '09

Hello all How can i generate an INVITE from Kamailio. I want generate many INVITE using parametre from data base. Who have an idea for this pb. Thanks Regards Aladin

3 2

[Kamailio-Users] kam 1.5 generating 407 on new branch
by Robert McGilvray 10 Sep '09

10 Sep '09

I'm pulling two destinations out of a sql database that I'd like to parallel fork on. I rewrite the main branches RURI to the first result that's returned then append the second result as a branch using assignment to $br. The first call always rings correctly but the second hangs and it looks like it's sending a 407 to my UAC which already authenticated on the initial INVITE. if ($avp(s:alias_counter) == 0) { # Replace the main branch URI with the first result of the DB $ru = "sip:" + $dbr(result=>[0,0]) + "@" + $dbr(result=>[0,1]); } else { # For all other rows append them as branches $br = "sip:" + $dbr(result=>[$avp(s:alias_counter),0]) + "@" + $dbr(result=>[$avp(s:alias_counter),1]); } Sep 9 14:20:36 [12535] DBG:core:parse_msg: SIP Reply (status): Sep 9 14:20:36 [12535] DBG:core:parse_msg: version: <SIP/2.0> Sep 9 14:20:36 [12535] DBG:core:parse_msg: status: <407> Sep 9 14:20:36 [12535] DBG:core:parse_msg: reason: <Proxy Authentication Required> Do I need to bypass my authentication route for new branches or am going about this wrong? In openser 1.3 using enum for forking I don't have to do any special logic for bypassing proxy_auth so I assumed it'd be similar. Thanks Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

2 1

Re: [Kamailio-Users] statistics not working
by Henning Westerholt 10 Sep '09

10 Sep '09

On Mittwoch, 9. September 2009, Anders wrote: > Thanks for the VERY quick answer! Hi Anders, np. :) Please always CC to the user list. > server:/# kamctl fifo get_statistics inuse_transactions > database engine 'MYSQL' loaded > Control engine 'FIFO' loaded > entering fifo_cmd get_statistics inuse_transactions > tm:inuse_transactions = 23 > > FIFO command was: > :get_statistics:openser_receiver_1853 > > inuse_transactions > > That looks alright, doesn't it? It looks ok, but perhaps the additional debugging output you got confuse the subsequent commands you want to run? Can you check your kamctlrc file? There is something like this: ## verbose - debug purposes - default '0' # VERBOSE=1 Perhaps it works when you deactivate this setting? Henning

2 2

[Kamailio-Users] statistics not working
by Anders 09 Sep '09

09 Sep '09

Hi list, I'm pretty green, so please bare with me if the questions are common-sense to some of you - I ask here only because I was unable to find or understand the answer elsewhere, like it the very technical documentation or somewhere else on the web. I suspect that something is wrong with the configuration somewhere, because I can't get much out of the statistics. Just look at the responses from these commands: > server:/etc/munin/plugins# kamctl fifo get_statistics active_dialogs | > cut -d" " -f3 'MYSQL' > 'FIFO' > get_statistics > 0 > was:\n:get_statistics:openser_receiver_6342\nactive_dialogs\n\n > server:/etc/munin/plugins# kamctl fifo get_statistics early_dialogs | > cut -d" " -f3 'MYSQL' > 'FIFO' > get_statistics > 0 > was:\n:get_statistics:openser_receiver_6580\nearly_dialogs\n\n > server:/etc/munin/plugins# kamctl fifo get_statistics > inuse_transactions | cut -d" " -f3 'MYSQL' > 'FIFO' > get_statistics > 0 > was:\n:get_statistics:openser_receiver_6593\ninuse_transactions\n\n > server:/etc/munin/plugins# kamctl fifo get_statistics database engine > 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd > get_statistics 400 Too few or too many arguments FIFO command was: > :get_statistics:openser_receiver_6610 Doesn't look right, does it? Where do you suggest that I start looking? Thanks!

2 1

[Kamailio-Users] Freezing for Kamailio 3.0 - Monday, Sep 14, 2009
by Daniel-Constantin Mierla 09 Sep '09

09 Sep '09

Hello, who ever has brand new features to push into upcoming Kamailio 3.0 has to do it this week. The plan is o freeze the code and start the testing phase at the beginning of next week, with bug hunting as main activity. As usual, work on additional tools and documentation can go on during testing. If discovered, important missing features from kamailio 1.5 will be considered bugs. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] error message with imc
by Rajesh 09 Sep '09

09 Sep '09

Hi , I have beeen configuring imc for quite some time and finally i got some error message that is related to IMC. thanks to you for helping me reach this point. With the error message, i think i should be able to rectify it with your help.My error message reads the following , ERROR:imc:imc_handle_message: room [q] does not exist! ERROR:imc:imc_manager: failed to handle 'message' . kindly guide me how to rectify the error.. -- Regards, Rajesh

3 4

/var/log/syslog in mediaproxy
by Rakhmadhany Primananda 09 Sep '09

09 Sep '09

hi everybody,, I'm using mediaproxy 1.7.2 and ser-0.9.6,, this is my output from syslog after I installed mediaproxy and running the process : root@serbuntu:/# cat /var/log/syslog | grep proxy > Sep 3 13:38:12 serbuntu mediaproxy[2214]: Listening for commands on local socket `/var/run/mediaproxy.sock' Sep 3 13:38:12 serbuntu mediaproxy[2214]: Listening for remote commands on `10.100.0.45:25060' Sep 3 13:38:12 serbuntu mediaproxy[2214]: Remote commands are allowed from: 10.100.0.45, 127.0.0.1 Sep 3 13:38:12 serbuntu mediaproxy[2214]: Using IP address `10.100.0.45' for the RTP/RTCP proxy Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Traceback (most recent call last): Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File"/usr/local/mediaproxy/proxydispatcher.py", line 87, in <module> Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: dispatcher.start() Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 114, in start Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: mp = MediaproxyCommandHandler() Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 232, in __init__ Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: raise RuntimeError, "couldn't create server (%s)" % why[1] Sep 3 13:38:13 serbuntu proxydispatcher[2217]: RuntimeError Sep 3 13:38:13 serbuntu proxydispatcher[2217]: : Sep 3 13:38:13 serbuntu proxydispatcher[2217]: couldn't create server (Address already in use) Sep 3 13:38:13 serbuntu proxydispatcher[2217]: exiting. Sep 3 14:08:01 serbuntu mediaproxy[2214]: received signal 15 Sep 3 14:09:09 serbuntu mediaproxy[4987]: Listening for commands on local socket `/var/run/mediaproxy.sock' Sep 3 14:09:09 serbuntu mediaproxy[4987]: Listening for remote commands on `10.100.0.45:25060' Sep 3 14:09:09 serbuntu mediaproxy[4987]: Remote commands are allowed from: 10.100.0.45, 127.0.0.1 Sep 3 14:09:09 serbuntu mediaproxy[4987]: Using IP address `10.100.0.45' for the RTP/RTCP proxy Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Traceback (most recent call last): Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/proxydispatcher.py", line 87, in <module> Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: dispatcher.start() Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 114, in start Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: mp = MediaproxyCommandHandler() Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 232, in __init__ Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: raise RuntimeError, "couldn't create server (%s)" % why[1] Sep 3 21:01:19 serbuntu proxydispatcher[7498]: RuntimeError Sep 3 21:01:19 serbuntu proxydispatcher[7498]: : Sep 3 21:01:19 serbuntu proxydispatcher[7498]: couldn't create server (Address already in use) Sep 3 21:01:19 serbuntu proxydispatcher[7498]: exiting. my question: what's the meaning of serbuntu proxydispatcher[7498]: raise RuntimeError, "couldn't create server (%s)" % why[1] and serbuntu proxydispatcher[7498]: couldn't create server (Address already in use) ? is it a bug or error? and this is my mediaproxy.ini configuration (mediaproxy and SER running in one machine) : root@serbuntu:/# cat /usr/local/mediaproxy/mediaproxy.ini > ; ; Configuration file for MediaProxy ; [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock group = ser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = ser listen = 10.100.0.45 allow = 10.100.0.45 proxyIP = 10.100.0.45 portRange = 60000:65000 ;TOS = 0xb8 ;idleTimeout = 60 ;holdTimeout = 3600 ;forceClose = 0 [Accounting] dbaccounting = off [Database] user = dbuser password = dbpass host = dbhost database = radius table = radacct thankq,, regards,, dhany

2 1

[Kamailio-Users] what does the following line mean
by Rajesh 08 Sep '09

08 Sep '09

Dear Experts, Iam configuring IMC and i would like to know what does the following line mean. xdbg("script: message from [$fu] chatroom000 [$ru] msg [$rb]\n"); .I get error in this line when i start kamailio. what does $fu ,$ru , $rb mean. Thanks

3 2

[Kamailio-Users] issues with IMC module
by Rajesh 08 Sep '09

08 Sep '09

Dear Experts, I have been trying to configure IMC for quite some time.I would like to know the reason why am i not gettin the module working.I understand that the command to create chatroom is #create roomname, but iam not very sure as to where do i add the create chat room command .Kindly take time to go through my cfg file and tel me where do i need to add the command to create chatroom. debug=4 # debug level (cmd line: -dddddddddd) fork=no # don't fork log_stderror=yes # log to stderr (cmd line: -E) children=0 # number of children check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5066 listen=192.168.64.75 #alias="stefan.lo" # ------------------ module loading ---------------------------------- mpath="/usr/local//lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "textops.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "imc.so" loadmodule "tm.so" loadmodule "xlog.so" loadmodule "registrar.so" loadmodule "usrloc.so" loadmodule "mi_fifo.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- registrar params -- modparam("registrar", "default_expires", 120) # -- registrar params -- modparam("usrloc", "db_mode", 0) # -- imc params -- modparam("imc", "imc_cmd_start_char", "#") modparam("imc|usrloc", "db_url", "mysql://openser:openserrw@192.168.64.75/openser") # -- tm params -- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; # registrations - if(is_method("REGISTER")) { save("location"); exit; } #imc##if(uri=~"sip:q.*@") if(uri=~"sip:chatroom000@192.168.64.75") { # IMC - message ##imc## xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); xdbg("script: message from [$fu] chatroom000 [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { ##edited as below for imc xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); xdbg("script: message not for room, from [$fu] chatroom000 [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; } -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

1 0

[Serusers] STUN server software?
by SIP 07 Sep '09

07 Sep '09

Anyone know any STUN server software that supports XOR_MAPPED_ADDRESS functionality? Open source or commercial is acceptable. N.

2 1

[Kamailio-Users] kamailio 1.4 INVITE authentication bypass
by Asim Riaz 07 Sep '09

07 Sep '09

Hi List, I am using kamailio 1.4 and authenticating INVITE if the source ip address is not in trusted table but one of the IP which is not in the trusted table was able to bypass INVITE authentication, . I don’t have SIP traces saved from the called but when that was happening I could see that the INVITE didn’t have auth credentials but caller was able to bypass authentication and was sending calls to my upstream gateway. Caller’s IP is definitely not in the trusted table, I am just wondering is it something wrong in my script or similar issue has reported before ; Thanks in Advance Asim route[2] { xlog("L_INFO", "[ROUTE-2] Received initial INVITE from $si\n"); setflag(2); setflag(3); if(is_from_local()) { if(!allow_trusted()) { xlog("L_INFO", "[ROUTE-2 !] Issuing proxy challenge\n"); if(!proxy_authorize("", "subscriber")) { proxy_challenge("", "1"); exit; } else if(!check_from()) { xlog("L_INFO", "[ROUTE-2 !] From URI denied\n"); sl_send_reply("403", "Forbidden"); exit; } } else { xlog("L_INFO", "[ROUTE-2 !] From URI domain not local - denied\n"); sl_send_reply("403", "Forbidden"); exit; } } consume_credentials(); xlog("L_INFO", "[ROUTE-2 ->] Authentication credentials valid\n"); if(nat_uac_test("1")) { xlog("L_INFO", "[ROUTE-2 ->] RFC1918 contact found - fixing up\n"); fix_nated_contact(); force_rport(); setbflag(7); } if(nat_uac_test("8") && search("Content-Type: application/sdp")) { xlog("L_INFO", "[ROUTE-2 ->] RFC1918 SDP endpoint found - fixing up\n"); fix_nated_sdp("10"); } # Apply outbound translations and figure out where to route the call. route(4); # this route the calls to upstream gateway. } These messages i was getting in syslog [ROUTE-2] Received initial INVITE from xxx.xxx.xxx.xxx(Caller_IP) ERROR:auth:consume_credentials: no authorized credentials found (error in scripts) [ROUTE-2 ->] Authentication credentials valid [ROUTE-4] Applying outbound translations to: 0022334455 [ROUTE-4 ->] Translated RURI user part to: 22334455 [ROUTE-4 ->] Gateway election: my_upstream_gateway [ROUTE-5] Accounting translation: sip:0022334455@my_upstream_gateway [ROUTE-2 ->] Relaying

3 9

[Kamailio-Users] imc related issues
by Rajesh 07 Sep '09

07 Sep '09

Hi, Iam trying to configure IMC. as i was told that this works fine, on a users system,I would like to know if i need to integrate it to my kamailio.cfg file or "can i use it all alone".. My cfg file is as folllows. debug=5 # debug level (cmd line: -dddddddddd) fork=no # don't fork log_stderror=yes # log to stderr (cmd line: -E) children=0 # number of children check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5063 listen=192.168.64.75 #alias="stefan.lo" # ------------------ module loading ---------------------------------- mpath="/usr/local//lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "textops.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "imc.so" loadmodule "tm.so" loadmodule "xlog.so" loadmodule "registrar.so" loadmodule "usrloc.so" loadmodule "mi_fifo.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- registrar params -- modparam("registrar", "default_expires", 120) # -- registrar params -- modparam("usrloc", "db_mode", 0) # -- imc params -- modparam("imc", "imc_cmd_start_char", "#") modparam("imc|usrloc", "db_url", "mysql://openser:openserrw@192.168.64.75/openser") # -- tm params -- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; # registrations - if(is_method("REGISTER")) { save("location"); exit; } #imc##if(uri=~"sip:q.*@") if(uri=~"sip:chatroom000@192.168.64.75") { # IMC - message ##imc## xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); xdbg("script: message from [$fu] chatroom000 [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { ##edited as below for imc xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); xdbg("script: message not for room, from [$fu] chatroom000 [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; } -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

1 0

Re: [Kamailio-Users] unknown column 'name' in field list
by Henning Westerholt 07 Sep '09

07 Sep '09

On Montag, 7. September 2009, Rajesh wrote: > > can you do issue a mysql "desc imc_rooms;" on your openser/ kamailio > > DB to check if the table is correctly setup? And do you configured a > > database url for the imc module? > > Thanks Henning for your prompt reply. > > I did give the command desc imc_rooms and found that the table do exist, > indeed i created them manually before setting up imc module.What else > could be the cause of the problem. Hi Rajesh, please always CC to the list. As hinted in some other mail, perhaps the module use not the correct DB URL? I just checked it on my system with 1.5 branch, it loads ok: Sep 7 12:56:14 ca ../kamailio[9312]: INFO:core:init_mod: initializing module imc Sep 7 12:56:14 ca ../kamailio[9312]: DBG:imc:mod_init: db_url=mysql://XXXX:XXXX@localhost/openser/43/0xb7dd4cac Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:find_mod_export: found <db_bind_api> in module db_mysql [../modules/] Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_bind_mod: using db bind api for db_mysql Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_do_init: connection 0x816ca10 not found in pool Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_new_connection: opening connection: mysql://xxxx:xxxx@localhost/openser Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_new_connection: connection type is Localhost via UNIX socket Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_new_connection: protocol version is 10 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_new_connection: server version is 5.0.75-0ubuntu10.2 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_new_result: allocate 28 bytes for result set at 0x816c8f8 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: 3 columns returned from the query Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_allocate_columns: allocate 12 bytes for result names at 0x816caa8 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_allocate_columns: allocate 12 bytes for result types at 0x816c988 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[0] at 0x816cb60 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x816cb60)[0]=[name] Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[1] at 0x816cb40 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x816cb40)[1]=[domain] Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[2] at 0x816c858 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x816c858)[2]=[flag] Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: use DB_INT result type Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_convert_rows: no rows returned from the query Sep 7 12:56:14 ca ../kamailio[9312]: INFO:imc:add_from_db: the query returned no result Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing 3 columns Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing RES_NAMES[0] at 0x816cb60 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing RES_NAMES[1] at 0x816cb40 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing RES_NAMES[2] at 0x816c858 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing result names at 0x816caa8 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing result types at 0x816c988 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_rows: freeing 0 rows Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_result: freeing result set at 0x816c8f8 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:find_cmd_export_t: found <load_tm>(0) in module tm [../modules/] Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:pool_remove: removing connection from the pool Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:init_mod: registering stats for imc Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:init_mod: register MI for imc Sep 7 12:56:14 ca ../kamailio[9312]: INFO:core:init_mod: initializing module maxfwd [..] I'd suggest that you increase the debug level to '4' in your cfg, and take a look to the output if the DB credentials are correct. Regards, Henning

1 0

[Kamailio-Users] Could some body pls provide me with a sample cfg file for IMC
by Rajesh 07 Sep '09

07 Sep '09

Dear All, I havae been trying to configure imc for quite some time..If any of you have configured IMC successfully, kindly send me the cfg file .Iam pretty sure that it would help me greatly and reduce my time taken to implement the module,IMC.I would appreciate your efforts for sending me the cfg file Thanks in advance!!!

3 2

[Kamailio-Users] unknown column 'name' in field list
by Rajesh 07 Sep '09

07 Sep '09

Dear All, Am configuring imc for kamailio and i would like to know , why am i getting the error on starting kamailio.My error is as follows " ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'name' in 'field list' Sep 7 14:23:25 [4034] ERROR:core:db_do_query: error while submitting query Sep 7 14:23:25 [4034] ERROR:imc:add_from_db: failed to querry table Sep 7 14:23:25 [4034] ERROR:imc:mod_init: failed to get information from db Sep 7 14:23:25 [4034] ERROR:core:init_mod: failed to initialize module imc Sep 7 14:23:25 [4034] ERROR:core:main: error while initializing modules " -- Thanks in advance

2 1

[Kamailio-Users] Siremis Login
by Davis Doe 06 Sep '09

06 Sep '09

Dear alberto, I am newbie to kamailio pabx and followed the steps by step and i have a problem by accessing the web interface. error: "You don't have permission to access /siremis/bin/ on this server." please help me out. davis

2 1

[Serusers] SER doesn't want to restart
by inge 04 Sep '09

04 Sep '09

Hello all, I have done some modification in my ser.cfg. And when I decided to restart, it failed. Here the output from "messages" : Sep 3 06:44:57 solo /usr/local/sbin/ser[23168]: Maxfwd module- initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: permissions - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: Default allow file (/usr/local/etc/ser/permissions.allow) not found => empty rule set Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: Default deny file (/usr/local/etc/ser/permissions.deny) not found => empty rule set Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: AVPops - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: CHECK_UA - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_row(): No memory left Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_rows(): Error while converting row #4207 Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_result(): Error while converting rows Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: store_result(): Error while converting result Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: preload_udomain(): Error while doing db_query Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: register_udomain(): Error while preloading domain 'location' Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: domain_fixup(): Error while registering domain Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: ERROR: fix_expr : fix_actions error Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: CHECK_UA - destroing module I had thinking that my modification were bad, so I leaved them. But SER failed again with the same output. I put some comments before the lines which used "check_ua" module, and then it worked ! I had never experience a problem with "check_ua" before that. Can someone help me to understand why it failed ?

2 2

[Kamailio-Users] NAT kamailio
by san la 04 Sep '09

04 Sep '09

Hi! I want out of Internet and NAT kamailio registration extension from the internet but do not know how. You can help me letters.! can send your configuration file kamailio.cfg to see also. Thank!

2 1

[Kamailio-Users] t_realy in failure_route?
by Ricardo Martinez 03 Sep '09

03 Sep '09

Hello I have a question. I hope that someone could clarify this to me. I have the next situation. I'm using a t_on_failure route along with the LCR module. So I have something like this : failure_route[1] { if (t_was_cancelled()) { xlog("L_INFO", "[$ci] $rm:$ru t_was_cancelled in failure_route\n"); exit; } if ( t_check_status("486") ) { xlog("L_INFO","[$ci] 486 Received\n"); } if (t_check_status("603")) { xlog("L_INFO","[$ci] 603 Received\n"); if (!next_gw()) { t_reply("503", "Service not available, no more gateways"); xlog("L_INFO", "[$ci] No hay mas gateways para $rm:$ou\n"); exit; } else { xlog("L_INFO", "[$ci] $rm cambiando $ou por $ru en failure_route"); } t_on_failure("1"); } t_relay(); } Maybe I'm misunderstanding how the failure_route works but with this configuration I was expecting that If I have a 603 from the callee I will do the "next_gateway" configuration. And If I have a 486 I will put a line in the log and then continue with the t_relay. So.. the 486 will be passed to the caller, but with the final answer from the caller ( ACK ) to the 486.. and I'm getting this error... Sep 2 16:43:41 ERROR:tm:t_forward_nonack: no branch for forwarding Sep 2 16:43:41 ERROR:tm:w_t_relay: t_forward_nonack failed So it seems that tries to forward the ACK with t_relay? Caller Proxy Callee ------> INVITE -----> INVITE <------ 486 -------> ACK <------ 486 -----> (error?) ACK But... When I modified the next lines in the failure_route if ( t_check_status("486") ) { xlog("L_INFO","[$ci] 486 Received\n"); exit; } And add the "exit" parameter, i don't have the above error and the "486" is passed back to the caller as "expected". Can someone explain me this? Thanks in advance.. Regards Ricardo Martinez.-

2 4

[Serusers] Need a way to delete a URI parameter if present from the INVITE
by Frank Durda IV 03 Sep '09

03 Sep '09

A given system is sending me INVITEs of the form: INVITE sip:6663338888@111.222.33.44;cic=9876;rn=6663331212;npdi SIP/2.0 that includes some URI parameters, eg the things after the semi-colon. In this case, the cic=nnnn value is really some internal routing number and not a real PSTN CIC value (like 0288 for AT&T). So when this eventually reaches a switch on the PSTN, that switch sees that someone has already determined the CIC and it attempts to route using what really isn't a CIC at all. I would like to strip this "cic=nnnn" out of the call within SER, and while there are a number of documented rewriteuri functions for altering the URI in SER, there doesn't seem to be anything that can be used to get at URI parameters. For the above, I would like to trim it to: INVITE sip:6663338888@111.222.33.44;rn=6663331212;npdi SIP/2.0 as the rn= and npdi values are sane and can be used by the destination switch. Is there such a beast, or is the only options to add a function/module or use the exec to invoke an external program to do the job? Thanks in advance.

2 3

[Serusers] /var/log/syslog mediaproxy
by Akhmad Dhany 03 Sep '09

03 Sep '09

hi everybody,, this is my output from syslog after I installed mediaproxy and running the process : root@serbuntu:/# cat /var/log/syslog | grep proxy Sep 3 13:38:12 serbuntu mediaproxy[2214]: Listening for commands on local socket `/var/run/mediaproxy.sock' Sep 3 13:38:12 serbuntu mediaproxy[2214]: Listening for remote commands on `10.100.0.45:25060' Sep 3 13:38:12 serbuntu mediaproxy[2214]: Remote commands are allowed from: 10.100.0.45, 127.0.0.1 Sep 3 13:38:12 serbuntu mediaproxy[2214]: Using IP address `10.100.0.45' for the RTP/RTCP proxy Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Traceback (most recent call last): Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File "/usr/local/mediaproxy/proxydispatcher.py", line 87, in <module> Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: dispatcher.start() Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 114, in start Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: mp = MediaproxyCommandHandler() Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 232, in __init__ Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: raise RuntimeError, "couldn't create server (%s)" % why[1] Sep 3 13:38:13 serbuntu proxydispatcher[2217]: RuntimeError Sep 3 13:38:13 serbuntu proxydispatcher[2217]: : Sep 3 13:38:13 serbuntu proxydispatcher[2217]: couldn't create server (Address already in use) Sep 3 13:38:13 serbuntu proxydispatcher[2217]: exiting. Sep 3 14:08:01 serbuntu mediaproxy[2214]: received signal 15 Sep 3 14:09:09 serbuntu mediaproxy[4987]: Listening for commands on local socket `/var/run/mediaproxy.sock' Sep 3 14:09:09 serbuntu mediaproxy[4987]: Listening for remote commands on `10.100.0.45:25060' Sep 3 14:09:09 serbuntu mediaproxy[4987]: Remote commands are allowed from: 10.100.0.45, 127.0.0.1 Sep 3 14:09:09 serbuntu mediaproxy[4987]: Using IP address `10.100.0.45' for the RTP/RTCP proxy Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Traceback (most recent call last): Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/proxydispatcher.py", line 87, in <module> Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: dispatcher.start() Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 114, in start Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: mp = MediaproxyCommandHandler() Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 232, in __init__ Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: raise RuntimeError, "couldn't create server (%s)" % why[1] Sep 3 21:01:19 serbuntu proxydispatcher[7498]: RuntimeError Sep 3 21:01:19 serbuntu proxydispatcher[7498]: : Sep 3 21:01:19 serbuntu proxydispatcher[7498]: couldn't create server (Address already in use) Sep 3 21:01:19 serbuntu proxydispatcher[7498]: exiting. my question: what's the meaning of serbuntu proxydispatcher[7498]: raise RuntimeError, "couldn't create server (%s)" % why[1] and serbuntu proxydispatcher[7498]: couldn't create server (Address already in use) ? is it a bug or error? thankq,, regards,, dhany

1 0

[Kamailio-Users] Siremis with postgres
by Karl Heinz Wolf 03 Sep '09

03 Sep '09

I just tried to get Siremis working with postgres. First I had problems because I created the table and columns with uppercase letters and the behaviour of mysql and postgres is different in this aspect so the Siremis queries weren't successful. Now I created the user access table the following way: CREATE TABLE ob_users ( sysid serial NOT NULL, userid varchar(15) NOT NULL, password varchar(15) default NULL, PRIMARY KEY (sysid) ); I also modified Config.xml: Driver="Pdo_Pgsql" instead of Driver="Pdo_Mysql" and Siremis seems to work fine with postgres. cheers, karl heinz

1 0

SER doesn't want to restart
by inge 03 Sep '09

03 Sep '09

Hello all, I have done some modification in my ser.cfg. And when I decided to restart, it failed. Here the output from "messages" : Sep 3 06:44:57 solo /usr/local/sbin/ser[23168]: Maxfwd module- initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: permissions - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: Default allow file (/usr/local/etc/ser/permissions.allow) not found => empty rule set Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: Default deny file (/usr/local/etc/ser/permissions.deny) not found => empty rule set Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: AVPops - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: CHECK_UA - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_row(): No memory left Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_rows(): Error while converting row #4207 Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_result(): Error while converting rows Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: store_result(): Error while converting result Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: preload_udomain(): Error while doing db_query Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: register_udomain(): Error while preloading domain 'location' Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: domain_fixup(): Error while registering domain Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: ERROR: fix_expr : fix_actions error Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: CHECK_UA - destroing module I had thinking that my modification were bad, so I leaved them. But SER failed again with the same output. I put some comments before the lines which used "check_ua" module, and then it worked ! I had never experience a problem with "check_ua" before that. Can someone help me to understand why it failed ?

1 0

[Kamailio-Users] support for 'include' in cfg file
by Daniel-Constantin Mierla 03 Sep '09

03 Sep '09

Hello, just committed to GIT support for 'include' in config file. Hopefully this helps in nicer structuring and maintenance of your configs. The syntax is simple: include path_to_file The path_to_file can be relative or absolute. If it is not absolute path, first attempt is to locate it relative to current directory, and if fails, relative to directory of the file that includes it. There is no restriction where include can be used or what can contain - any part of config file is ok. There is a limit of maximum 10 includes in depth, otherwise you can use as many includes as you want. Reporting of the cfg file syntax errors prints now the file name for easier troubleshooting. Example: route { ... include /sr/checks.cfg ... } --- /sr/checks.cfg --- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } --- Testing and feedback is much appreciated. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

5 13

[Kamailio-Users] imc issues
by Rajesh 03 Sep '09

03 Sep '09

On 03.09.2009 12:00 Uhr, Rajesh wrote: Yes i did cretae.. i completed authentication and presence module successfully.. iam caught up with this.. am posting my cfg file below. kindly correct me if iam wrong. ************************************************************************************************************************************************** ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 ##fork changed to NO manually from yes fork=yes #fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local//etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local//etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local//etc/kamailio/tls/user/user-calist.pem" port = 5067 /* 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:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local//lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "textops.so" loadmodule "usrloc.so" loadmodule "maxfwd.so" loadmodule "imc.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "registrar.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "domain.so" loadmodule "presence.so" loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" ##insert function tested here # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr 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) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* 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", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc","db_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url","mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") ###registrar params added for imc # -- registrar params -- modparam("registrar", "default_expires", 120) # -- registrar params -- modparam("usrloc", "db_mode", 0) # -- imc params -- modparam("imc", "imc_cmd_start_char", "#") modparam("imc|usrloc", "db_url", "mysql://openser:openserrw@192.168.64.75/openser") ##the following three lines were added seperately for imc modparam("imc", "rooms_table","imc_rooms") modparam("imc", "rooms_table","imc_members") modparam("imc", "hash_size", 8) # -- tm params -- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ modparam("alias_db", "db_url","mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ modparam("presence|presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:192.168.64.75:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ insert_hf("To-username: test"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } ########The following set of lines were added for implementing imc # registrations - ##the following line has been added for creating chat rooms ##insert_hf (#create chat-000) if(is_method("REGISTER")) { save("location"); exit; } ##if(uri=~"sip:q*@") ## if(uri=~"sip:chat0000@192.168.64.75") ##the below line has been edited by Rajesh. if(uri=~"sip:chat0000@192.168.64.75") { # IMC - message #imc xdbg("script: message from [$fu ] r-uri [$ru ] msg [$rb]\n"); ##the above two lines are edited for imc by Rajesh. xdbg("script: message from [$fu ] chatroom000 [$ru ] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { xdbg("script: message not for room, from [$fu ] r-uri [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; #######The above lines were added for implementing imc #insert function tested below insert_hf("Test1"); # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { ## insert functioin tested below insert_hf("Test_in_ACK"); if ( t_check_trans() ) { # non 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.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { #insert_hf("To-username: test"); setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain insert_hf ("Test2"); if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { 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 setflag(2); route(1); } ##insert function added below ##insert_hf ("Test4"); route[1] { ##insert function added below insert_hf ("Test5"); #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { insert_hf ("Test5"); ##tried insert function here. #t_on_branch("1"); ##tried here insert_hf("To-username: test"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { ##tried here for insert function. #insert_hf("Test_Presence"); #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("192.168.64.75", "subscriber")) { www_challenge("192.168.64.75", "0"); exit; } ###the following lines are uncommented if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($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 } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE) #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { 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(); ##} } ************************************************************************************************************************************************** Daniel-Constantin Mierla wrote: Hello, have you created the db tables: kamdbctl create? Cheers, Daniel On 03.09.2009 11:44 Uhr, Rajesh wrote: Dear Team, Iam configuring imc for kamailio-1.5 and while iam trying to load the module, i get the following error .Kindly guide me with a solution ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'name' in 'field list' Sep 3 12:24:20 [18583] ERROR:core:db_do_query: error while submitting query Sep 3 12:24:20 [18583] ERROR:imc:add_from_db: failed to querry table Sep 3 12:24:20 [18583] ERROR:imc:mod_init: failed to get information from db Sep 3 12:24:20 [18583] ERROR:core:init_mod: failed to initialize module imc Sep 3 12:24:20 [18583] ERROR:core:main: error while initializing modules --

1 0

[Kamailio-Users] DIVERSION or UAC_REDIRECT or BOTH
by Omar Mendoza 03 Sep '09

03 Sep '09

I got confused: Using Kamailio 1.4.3 uac_redirect is been used to handle diversion, also the draft expired there is anew one http://tools.ietf.org/html/draft-levy-sip- diversion-10, is this used together with the diversion module, or no need at all. or is diversion module when you have customers registered in your system? I am do not have phones registered in my system at all, i use mainly the CR module, and i need to allow diversion headers. which one to use, or both?! Also i am scare to use the diversion due the CDRs will create a new record for that, who i will charge the call for?! :) and if i use UAC_REDIRECT should i be able to use the avp $dip and the following, from ROUTE to add to the accounting? Diversion header URI $di - reference to Diversion header URI Diversion "privacy" parameter $dip - reference to Diversion header “privacy” parameter value Diversion "reason" parameter $dir - reference to Diversion header “reason” parameter value __________________________ O the Great

3 4

[Kamailio-Users] problems with imc
by Rajesh 03 Sep '09

03 Sep '09

Dear Team, Iam configuring imc for kamailio-1.5 and while iam trying to load the module, i get the following error .Kindly guide me with a solution ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'name' in 'field list' Sep 3 12:24:20 [18583] ERROR:core:db_do_query: error while submitting query Sep 3 12:24:20 [18583] ERROR:imc:add_from_db: failed to querry table Sep 3 12:24:20 [18583] ERROR:imc:mod_init: failed to get information from db Sep 3 12:24:20 [18583] ERROR:core:init_mod: failed to initialize module imc Sep 3 12:24:20 [18583] ERROR:core:main: error while initializing modules -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

3 2

[Kamailio-Users] Problem configuring imc
by Rajesh 02 Sep '09

02 Sep '09

Deat Team, I would like to know how to embedd the message #create chatroom into the message body of the SIP message. I cameacross a post, wher i was told that instead of SIP,XMPP is better used for the purpose of instant messaging conference.It would be greatly helpful to me, if u can explain it to me -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

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Re: [Kamailio-Users] question about htable size
by jh＠tutpro.com 02 Sep '09

02 Sep '09

Edson Gellert Schubert writes: > http://lists.kamailio.org/pipermail/users/2009-August/024463.html ok, looks like size has the same meaning as in lcr module, i.e., it just affects the size of hash table, not number of entries stored in it. -- juha

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[Kamailio-Users] question about htable size
by jh＠tutpro.com 02 Sep '09

02 Sep '09

htable module readme says abotu htable param: size - number specifying the size of hash table. The number of entries in the table is 2^size does it mean the size of the hash table or how many entries can be stored in hash table? in lcr module, for example, lcr_hash_size param specifies the former and there is no limit on how many entries a hash table can contain: Defines the size of hash table used to store <prefix, from_pattern, priority> tuples. Hashing is done based on prefix. Larger value means less collisions with other prefixes. Hash size value should be a power of 2. it is the same with htable size param? -- juha

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[Kamailio-Users] job offer: VoIP developer, VoIP admin
by Henning Westerholt 02 Sep '09

02 Sep '09

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have two open positions in Karlsruhe, Germany. For more informations (requirements, how to apply..) please refer to the openings on our job page (in german). Some facts about our VoIP system can be found at e.g. [1], please feel free to contact me per mail as well. VoIP Operation, system administration position: http://www.unitedinternet.de/xml/ui/JobsJobDetail;?__frame=_top&__lf=JobsAp… VoIP Software development, R&D position: http://www.unitedinternet.de/xml/ui/JobsJobDetail;?__frame=_top&__lf=JobsAp… Best regards, Henning Westerholt [1] http://www.kamailio.org/events/2009-linuxtag/presentation.pdf -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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Re: [Kamailio-Users] how to create chatrooms in imc
by Iñaki Baz Castillo 02 Sep '09

02 Sep '09

2009/9/2 Rajesh <Rajesh.sreedharan(a)siptech.com>: > Thanks again for your time. Please, don't send private mails. No free consultance service here. > Could you tell me how to add send messages via xmpp protocol.. Hum... use a XMPP client for that. I don't understand your question. > i was also asked to send cc to mailing list..is > users(a)lists.openser-project.org the mail address to which i need to mark cc > to The Kamailio users maillist is users(a)lists.kamailio.org. Use it instead of OpenSER's maillist as Kamailio is the new name of the project. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] how to create chatrooms in imc
by Daniel-Constantin Mierla 02 Sep '09

02 Sep '09

always cc to mailing list, private messages are ignored unless they are used for advanced troubleshooting where privacy is required. On 02.09.2009 12:32 Uhr, Rajesh wrote: > Dear Daniel, > Thankyou so much for your post. I read the > "readme" file and i am still unclear. I would like to know where to > add the command to create the chat room.Do i need to add it in the > .cfg file.Kindly excuse me if my questions are vague as am little new > to this Product. yes, in the config file you have to call imc_manager(): http://kamailio.org/docs/modules/1.5.x/imc.html#id2452973 Cheers, Daniel > > Daniel-Constantin Mierla wrote: >> Hello, >> >> On 02.09.2009 12:05 Uhr, Rajesh wrote: >>> Dear Openser Team, >>> I am curretnly configuring imc >>> module for kamailio and i would like to know(i) how to create chat >>> rooms , (ii)how to add users to the current chatrooms . One mail >>> from you can save a lot of time for me. >> >> see here: >> http://kamailio.org/docs/modules/1.5.x/imc.html#id2467570 >> >> With default config options for imc module, the command to create a >> chat room is: >> #create chatroom1 >> >> See more in the readme. >> >> Cheers, >> Daniel >> >>> >>> -- >>> Thanks & kind Regards, >>> Rajesh Sreedharan >>> Software Trainee-Wireless & Mobile Technology >>> >>> SIPTECH Solutions Limited >>> G4, Elnet Software City, CPT Road, >>> Taramani, Chennai - 600 113, India. >>> Tel : +91 (44) 2254 1473/ 74 >>> Extn.:206 >>> Fax : +91 (44) 2254 1475 >>> E-Mail :Rajesh.Sreedharan@Siptech.com >>> <mailto:Rajesh.Sreedharan@Siptech.com?> >>> Website : www.siptech.com <http://www.siptech.com/> >>> <http://www.siptech.com> >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> > > > -- > Regards, > Rajesh Sreedharan > Software Trainee-Wireless & Mobile Technology > > SIPTECH Solutions Limited > G4, Elnet Software City, CPT Road, > Taramani, Chennai - 600 113, India. > Tel : +91 (44) 2254 1473/ 74 > Extn.:206 > Fax : +91 (44) 2254 1475 > E-Mail :Rajesh.Sreedharan@Siptech.com > <mailto:Rajesh.Sreedharan@Siptech.com?> > Website : www.siptech.com <http://www.siptech.com/> > <http://www.siptech.com> -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] how to create chatrooms in imc
by Rajesh 02 Sep '09

02 Sep '09

Dear Openser Team, I am curretnly configuring imc module for kamailio and i would like to know(i) how to create chat rooms , (ii)how to add users to the current chatrooms . One mail from you can save a lot of time for me. -- Thanks & kind Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

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[Kamailio-Users] kindly approve my request
by Rajesh 02 Sep '09

02 Sep '09

Deat Team, Kindly approve my request for posting my questions on openser/kamailio on ur webpage. -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

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[Kamailio-Users] registrar lookups without reason
by jh＠tutpro.com 02 Sep '09

02 Sep '09

when analyzing the weird problem, where request uri gets rewritten with aor's contact even when there is no command in config file that would do it, i noticed these lines in syslog: Aug 31 16:41:50 /usr/sbin/kamailio[23139]: DBG:core:rewrite_uri: rewriting Request-URI with 'sip:foo.bar@ip-address:5060;transport=UDP' Aug 31 16:41:50 opensipg /usr/sbin/kamailio[23139]: DBG:registrar:lookup: 'foo.bar@ip-address' Not found in usrloc Aug 31 16:41:50 opensipg /usr/sbin/kamailio[23139]: DBG:registrar:lookup: 'foo.bar@ip-address' Not found in usrloc any one has any idea why lookup was called even when it is not in the script? this is getting very weird. looks like kamailio is executing statements on its own. -- juha

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[Serusers] CANCEL passed through SER is occasionally ignored by destination switch
by Frank Durda IV 02 Sep '09

02 Sep '09

I have a situation where CANCEL messages sent by the caller through SER are sometimes ignored by the destination switch. These events are infrequent, and seem to be ignored only if they are received in a narrow window between the 100 Trying and the 183 Session Progress or 180 Ringing message. The destination switch eventually emits a 183/180 and even a 200 OK as the called party answers the call of a long-departed caller. I don't believe that this is a race condition, as the called numbers are sometimes wireless parties, who have 4-10 seconds of time between 100 Trying/IAM and 180/183/ACM responses, and I have seen ignored cancels at the beginning and middle of that lengthy period. So a race condition for that large a period with such an infrequent occurrence seems unlikely. The destination (SIP to ISUP gateway) equipment vendor points out that in the dozen samples of this occurring that the branch tag of the CANCEL didn't match the one shown in the INVITE, and per RFC 3261, their decision to ignore the CANCEL seems to be the right one. In particular, section 17.2.3 criteria 1. This immediately made me think of SER and its habit of saving time and memory by not remembering some or all call state unless you tell it to, via various knobs that are turned off by default. However, I can't explain why most cancels that arrive in the same point in the call setup apparently do meet criteria and only a small number don't. It would seem that SER would have mismatched branch tags all the time or never, depending on the knob settings. Any ideas? Thanks in advance.

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[Kamailio-Users] SIP Redirect server
by Meftah Tayeb 01 Sep '09

01 Sep '09

hello all please could anyone help me to use Kamailio a a redirect server from server1.com that is the kamailio redirect server to server2.com that is the registrare server? thanks for any help __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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[Kamailio-Users] Symmetric NAT
by David 01 Sep '09

01 Sep '09

Hey, So I have setup a Kamailio server, it works well with most devices, but works terribly with Symmetric NAT. If you haven't already guessed, I am using STUN for NAT traversal. Could some people share there experiences with Symmetric NAT ? I see a lot of places saying that STUN is not good with it, but I haven't found a lot of suggestions for how to handle it. Thanks, David

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[Kamailio-Users] LCR questions ( ping and flags columns )
by Ricardo Martinez 01 Sep '09

01 Sep '09

Hello list. I have a couple of questions about the LCR module. The first one is about the "ping" column in the gw table. I guess this column stores the "results" of execute a ping to the gateways in the LCR.gw table isn't?. So how this "ping" is executed? It is just a simple ICMP message? Or maybe is an OPTIONS sip message to the gateway?. I saw in the lcr_mod.c file some ping_method, ping_interval variables, but in the README file there is nothing about the "ping". Could someone clarify this to me this? Second. In the mysql "gw" table there is a column called "flags". According to the README file it said that this column can store gateways "capabilities". For example, and I hope that someone could guide me here, can i store the maximum channels available per gateway?, and use it to select or not select a gateway according the gateway channels used?. I hope that someone could help me here. Thank in advance, Regards, Ricardo Martinez.-

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[Kamailio-Users] Kamailio wins Best of Open Source award.
by Alex Balashov 31 Aug '09

31 Aug '09

http://www.infoworld.com/d/open-source/best-open-source-software-awards-200… -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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[Kamailio-Users] watchers and active_watchers tables
by catalina oancea 31 Aug '09

31 Aug '09

Hi, Could someone please explain the usage of the watchers table? I understand the usage active_watchers, but why the watchers table too? Thanks, Catalina

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[Kamailio-Users] Issue in starting openser-0.9.5 for embedded application
by Ratheendran R 31 Aug '09

31 Aug '09

Hi All, I am using openser-0.9.5 version for embedded application,I am able to sucessfully cross compile(gcc 2.95) it.however the application fails to start. The above SER version is selected based on its compatibility with the cross compiler. below are my observations 1.if I run './openserctl start' from the command line I get the below message printf: No such file or directory PID file /var/run/openser.pid does not exist -- OpenSER start failed 2. If I run './openser' from command line I get the below message Listening on udp: 192.168.1.22 [192.168.1.22]:5060 tcp: 192.168.1.22 [192.168.1.22]:5060 Aliases: Too much shared memory demanded: 33554432 Please guide me in fixing this issue. Thanks in advance. Ratheendran

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[Kamailio-Users] rewriteuser( ) with avp
by Alex R.S.M 31 Aug '09

31 Aug '09

Hi, I'm trying to change "username" field of URI in the INT message. When using rewriteuser() function with avp variable as below, the username is changed to $avp(i:8) itself instead of the value of the avp. rewriteuser("$avp(i:8)"); I've tried all sorts of diff formats and none of them replace username with the value of avp variable.: rewriteuser($avp(i:8)); rewriteuser("avp(i:8)"); is this a bug? Thanks, A-RS _________________________________________________________________ Windows Live: Keep your friends up to date with what you do online. http://windowslive.com/Campaign/SocialNetworking?ocid=PID23285::T:WLMTAGL:O…

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[Kamailio-Users] scope of AVPs (AVP defined when execution of script starts)
by jh＠tutpro.com 29 Aug '09

29 Aug '09

i have always thought that AVPs exits per transaction. that does not seem to be the case, however, because when i put in the beginning of my script xlog("L_INFO", "Caller lcr_id $avp(CALLER_LCR_ID_AVP)\n"); this gets printed to syslog: Aug 29 11:34:34 localhost /usr/sbin/sip-router[10210]: INFO: Caller lcr_id 2 is this the indented behavior? do i really need to delete all AVPs in the beginning of the script in order to make sure that they don't have any values left from processing of previous message? -- juha

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[Kamailio-Users] Phones that support TLS with certificate
by catalina oancea 27 Aug '09

27 Aug '09

Hello, I am trying to find the phones that can use tls with certificate, I mean that the phone doesn't accept incoming calls unless the server certificate is signed by a public authority or manually added to a list. For now my conclusions are: -Snom 360 - works with TLS. But although snom wiki states that a certificate is needed, incoming calls are accepted even if the certificate is not uploaded. -SJPhone - same as snom -Bria - this is the only phone I could find that works only if the certificate is loaded. Has anybody used other phones with tls and can give me a piece of advice? also, maybe I am not configuring Snom right and that's why it doesn't need a certificate, any ideas? Thanks Catalina

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[Kamailio-Users] Perl module and memory leak in append_branch
by James Puckett 26 Aug '09

26 Aug '09

I have followed a bit of the discussions regarding the problems with memory management between the perl modules and the core kamailio software. I need to get append_branch working from within the perl module so that it doesn't cause a memory leak. It's been a while since I did any serious straight C programming, but if someone can point me to the correct way to handle memory allocated with the module so that it gets freed in the core I'd be glad to take this problem on. Specifically, I'm using something that looks like: $m->append_branch("<sip:".$cdpn."\@".$peers{$_}."?From=%3Csip:". $sourcenumber."@".$destIp."%3E;tag=".$fromtag.">", $qval); This leaves about 190 bytes per call unfreed. Thanks. James Puckett - Engineers' Consulting Group Phone 229-316-0012 - Fax 229-316-1490

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