- sr-users - kamailio.org

Avoiding to send 603 DECLINED to un-trusted IPs - Is that possible with Kamilio?
by Bruce B 23 Jul '11

23 Jul '11

Hi everyone, Brand new to kamailio and was told that Asterisk is not able to NOT send 603 DECLINED for anonymous SIP Invites. Is it possible to set Kamilio to absorb SIP invites from all over the world and only respond to those requests whose IPs are registered as PEERS and to not respond back AT ALL to those IPs that are not registered peers? Thanks, Bruce

1 0

Siremis & Asterisk
by Roman Dmytriv 22 Jul '11

22 Jul '11

Hello, I have my Kamailio super node up and running on Debian 6.0 machine and have few questions: I'm planning to install Asterisk and Siremis on top of it. Which I should do first if it matters? Do you have any guides documentations on how to do it? I was testing Jitsi and found out that I can only use IP for registrar not FQDN. For proxy I can use FQDN no problem. What could possibly cause that? Regards Roman

2 2

dispatcher - call load distribution algorithm
by Giuseppe Carella 21 Jul '11

21 Jul '11

Hi all, I'm using kamailio like dispatcher, for INVITE requests. The client is the IMS bench as UAC, and the backend node are sip servlet container, in which is running an UAS servlet. Between the client and kamailio there is the OpenIMSCore. I was using the hash over callid like algorithm, and all was going well. Now I'm trying to use the last algorithm, based on the call load of the destination backend, and the ACK message of the same dialog is sent to another instance. I read on the documentation that there is no SIP dialog tracking depending on other modules. This is my configuration: # main request routing block route { if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; } # select from first dst group by round-robin if(!ds_select_dst("1", "10")) { sl_send_reply("500", "No destination available"); exit; } t_on_failure("RTF_DISPATCH"); if(!t_relay()) { sl_reply_error(); exit; } } # dispatcher failure routing block failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # select next destination only for local timeout if (t_branch_timeout() & !t_branch_replied()) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); t_relay(); exit; } } } Thanks and best regards, Giuseppe

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Kamailio presence server problem with PostgreSQL
by Roberto Fichera 21 Jul '11

21 Jul '11

Hi all, I've setup for testing purpose an xcap server with Kamailio v3.1.4 backed with a postgresql v8.4.8 all in 64bits versions. On that configuration, I'm getting the errors below and consequently the presence doesn't correctly reports the status. Does anyone has the same problem? Best regards, Roberto Fichera. Jul 19 10:31:52 roberto /usr/sbin/kamailio[2005]: INFO: pua_usrloc [ul_publish.c:211]: should not send ul publish Jul 19 10:31:52 roberto /usr/sbin/kamailio[2015]: ERROR: <script>: ===== xhttp: xcap-caps : <null> Jul 19 10:31:52 roberto /usr/sbin/kamailio[2017]: ERROR: <script>: ===== xhttp: resource-lists : sip:roberto@192.168.2.92 Jul 19 10:31:52 roberto /usr/sbin/kamailio[2017]: ERROR: <script>: ===== xhttp: get sip:roberto@192.168.2.92 => /xcap-root/resource-lists/users/sip:roberto@192.168.2.92/index Jul 19 10:31:52 roberto /usr/sbin/kamailio[2021]: ERROR: <script>: ===== xhttp: pres-rules : sip:roberto@192.168.2.92 Jul 19 10:31:52 roberto /usr/sbin/kamailio[2021]: ERROR: <script>: ===== xhttp: get sip:roberto@192.168.2.92 => /xcap-root/pres-rules/users/sip:roberto@192.168.2.92/presrules Jul 19 10:31:52 roberto /usr/sbin/kamailio[1999]: ERROR: presence [subscribe.c:711]: Missing or unsupported event header field value Jul 19 10:31:52 roberto /usr/sbin/kamailio[1999]: ERROR: presence [subscribe.c:714]: #011event= message-summary Jul 19 10:31:52 roberto /usr/sbin/kamailio[2004]: INFO: presence [notify.c:1588]: NOTIFY sip:roberto@192.168.2.92 via sip:roberto@192.168.2.92:5090;transport=udp;registering_acc=192_168_2_92 on behalf of sip:roberto@192.168.2.92 for event presence.winfo Jul 19 10:31:52 roberto /usr/sbin/kamailio[2005]: INFO: presence [notify.c:1588]: NOTIFY sip:roberto@192.168.2.92 via sip:roberto@192.168.2.92:5090;transport=udp;registering_acc=192_168_2_92 on behalf of sip:mattia@192.168.2.92 for event presence *Jul 19 10:31:52 roberto /usr/sbin/kamailio[1999]: ERROR: db_postgres [km_dbase.c:427]: invalid query, execution aborted Jul 19 10:31:52 roberto /usr/sbin/kamailio[1999]: ERROR: db_postgres [km_dbase.c:428]: driver error: PGRES_FATAL_ERROR, ERROR: null value in column "sender" violates not-null constraint#012* *Jul 19 10:31:52 roberto /usr/sbin/kamailio[1999]: WARNING: db_postgres [km_dbase.c:473]: unexpected result returned*

5 17

xcap server - http load–balancer: xcap authentication problem
by laura testi 21 Jul '11

21 Jul '11

Hello all, I’m using two Kamailio Presence Servers that are also XCAP Servers for the presence and resources management and the Kamailio Dispatcher in order to balance the SIP requestes coming from SIP Client. Then I’m trying to configure a HTTP load–balancer for the xcap/http traffic. I’m testing both Apache and INginx HTTP proxy but I’m facing with a problem related with the Xcap authentication: when a client sends the first xcap request to the http load-balancer, it forwards the request to one of the configured Xcap Server that replies with the 401 response in order to let the client authenticates. So, the client sends the second request with the Digest authentication to the http load-balancer. But, sometimes, the http load-balancer forwards the authenticate request to the second Xcap Server that is not able to authenticate the request. Then the client receive the “Unauthorized” response… Could you please give us some hints about this problem? Thanks in advance, laura

4 11

dns_srv_lb
by Jon Bonilla 21 Jul '11

21 Jul '11

Hi all I'm running a kamailio 3.1 based proxy which actually relays REGISTER requests to a domain "example.com" I've setup a DNS SRV record for that domain which returns two hosts with same prio and weight. I want kamailio to balance the requests to both servers 50%. Taking a look to ./doc/dns.txt I've setup these configuration options: dns_try_ipv6=no dns_retr_time=1 dns_retr_no=1 dns_use_search_list=no use_dns_cache=on use_dns_failover=on dns_srv_lb=on If I understood right the last setting would have to make kamailio work as I described, balancing the requests between both servers. But my logs say that kamailio is actually using only one of the ip addresses (host2.example.com) to send the REGISTER requests. Here's the output of sercmd dns.view: sercmd> dns.view { name: _sip._udp.example.com type: SRV size (bytes): 232 reference counter: 1 permanent: no expires in (s): 31364 last used (s): 446 negative entry: no rr name: host1.example.com rr port: 5060 rr priority: 0 rr weight: 0 rr expires in (s): 31364 rr name: host2.example.com rr port: 5060 rr priority: 0 rr weight: 0 rr expires in (s): 31364 } { name: host2.example.com type: A size (bytes): 128 reference counter: 1 permanent: no expires in (s): 592964 last used (s): 446 negative entry: no rr ip: 192.168.0.103 rr expires in (s): 592964 } { name: host1.example.com type: A size (bytes): 128 reference counter: 1 permanent: no expires in (s): 592964 last used (s): 11835 negative entry: no rr ip: 192.168.0.102 rr expires in (s): 592964 } Any suggestions? What am I doing wrong? Thanks, Jon

2 2

Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64
by Coca 21 Jul '11

21 Jul '11

Hi List, I have a Kamailio3.1 server and RTPProxy running in WAN. The calls between UA will automatically terminated in Callee UA side 36s after connected, while no one sends a BYE. While Kamailio and UA are in LAN at all , everything is just working well. Is it my rtpproxy doesn't working correctly or something else is wrong? What can I do to fix it. Any hint?? BTW, Kamailio is installed following official guide http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git The kamailio.cfg wasn't changed at all except for below: ----------------------------------------------------------------------------------------- 1) adding the following lines: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT 2)uncommenting the line below in route[REGISTRAR], setbflag(FLB_NATSIPPING); 3)change rtpproxy port corresponding modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") ----------------------------------------------------------------------------------------- and my rtpproxy1.2.1 was installed by apt-get install rtpproxy, with 1) /etc/default/rtpproxy changed into: # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 LISTEN_ADDR=xx.xx.xx.xx # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" 2) and started by rtpproxy -l xx.xx.xx.xx -s udp:localhost:22222 -u kamailio Your help will be great appreciated. Coca

1 0

Kamailio: 500 Server error occurred (19/SL).
by Max Doronin 21 Jul '11

21 Jul '11

Hello guys, I have strange behavior on my Kamailio-1.5.1-notls instance. I have LCR and TM modules enabled.For a particular destination I have 4 routes to try. When I call to non-existing number, it tries - route 1 (404 Not Found) - route 2 (404 Not Found) - route 3 (503 Service unavailable) And the problem is with the last 503. Kamailio ACKs it, immediately drops these 2 lines into syslog Jul 13 04:14:47 /usr/local/sbin/kamailio[23572]: ERROR:tm:t_forward_nonack: no branch for forwarding Jul 13 04:14:47 /usr/local/sbin/kamailio[23572]: ERROR:tm:w_t_relay: t_forward_nonack failed And sends SIP/2.0 500 Server error occurred (19/SL). To the call originator. 30 microseconds later it sends SIP/2.0 500 Service Unavailable. to the originator again Latter 500 looks like the relayed original "503 Service unavailable". I think so because the Reason header is the same: Reason: SIP;cause=503;text="Service Unavailable";icodetext="NoCircuitAvailable";iintcode=10034;isubsystem=3. Obviously call is terminated. I noticed that that last peer rewrites my Via headers like this: Original: == Via: SIP/2.0/UDP 187.45.214.132;branch=z9hG4bK3dac.fae9b793.2. Via: SIP/2.0/UDP 192.168.2.33:5622;received=49.49.59.23;branch=z9hG4bK-d8754z-a6ba40014f97fb7c-1---d8754z-;rport=23832. == Rewritten == Via: SIP/2.0/UDP 187.45.214.132;received=187.45.214.132;branch=z9hG4bK3dac.fae9b793.2,SIP/2.0/UDP 192.168.2.33:5622;received=49.49.59.23;branch=z9hG4b == I tried to modify via1_matching parameter (1 -> 0) but no changes. My questions are: - What can be the reason of that 500 with 19/SL - Can kamailio properly handle this 1 line Via header? - What can be the reason of that "t_forward_nonack: no branch for forwarding"? Regards, Max

2 4

(no subject)
by Eliezer Croitoru 20 Jul '11

20 Jul '11

1 0

Re: [SR-Users] IMC module unresolved issues
by Ozren Lapcevic 20 Jul '11

20 Jul '11

Hi, did anyone run into similar problems or have successfully set up IMC module with mysql? Can you please share your know-how on the subject? Besides issue described below, I also had several other issues, but I think they all might be related to the fact that my rooms and members are only stored in cache, but not in the database. To repeat, this is really strange behavior because all other modules that I'm using write to the database without a problem. Best regards Ozren On Mon, May 30, 2011 at 6:28 PM, Ozren Lapcevic <ozren.lapcevic(a)gmail.com>wrote: > Hi all, > > I'm trying to configure IMC module and get it to work properly but I've run > into problems with saving IMC module data to mysql database. > > I'm using following code provided in IMC module documentation: > > modparam("imc", "db_url", DBURL) > > ## route block ### > if(is_method("MESSAGE") && (uri=~ "sip:chat-[0-9]+@") || (uri=~ > "sip:chat-manager@")) { > xlog("L_INFO", "going to imc: F [$fu] T [$tu] B [$rb]\n"); > > if(imc_manager()) > sl_send_reply("200", "ok - imc_manager()"); > else > sl_send_reply("500", "command error"); > exit; > } > > I am creating chat room with following message sent to chat-manager(a)abc.hr: > "#create chat-111" > > DEBUG log returns following: > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > INFO: <script>: going to imc: F [sip:oz@abc.hr] T [sip:chat-manager@abc.hr] > B [#create chat-111] > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc.c:525]: found command > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_cmd.c:146]: command: [create] > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_cmd.c:151]: parameter 0=[chat-111] > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_cmd.c:171]: new room [chat-111] > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_cmd.c:186]: added room uri= sip:chat-111@abc.hr > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_mng.c:322]: [uri]= sip:oz@abc.hr > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_mng.c:326]: [user]= oz > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_cmd.c:196]: added the owner as the first member [ > sip:oz@abc.hr] > > "./kamctl fifo imc_list_rooms" returns following: > ROOM:: URI=sip:chat-111@abc.hr MEMBERS=0 OWNER=sip:oz@abc.hr > > "./kamctl fifo imc_list_members sip:chat-111@abc.hr" returns following: > ROOM:: sip:chat-111@abc.hr NR_OF_MEMBERS=1 > MEMBER:: sip:oz@abc.hr > > However, that data is not persisted/saved in the imc_rooms and imc_members > tables in the database. Any clues what might be the problem here? > > Any help is greatly appreciated. > > Best regards, > Ozren Lapcevic > > PS > I'm runing Kamailio 3.1.4 with default cfg file with enabled mysql, > multidomain, usrlocdb, auth and NAT. I can register clients, establish > calls, send IMs, etc. imc_rooms and imc_members tables were created when > kamailio was installed. I've also checked imc-create.sql script and tables > are correctly created. > > The same DBURL used for imc is also used for auth_db, domain and usrloc > modules and I haven't experienced any problems with those modules and > database connection/persistence. domain, location, version and subscriber > tables are correctly populated. > > I'm not getting any errors from imc_manager(), it returns 200 OK. I can't > trace any errors in DEBUG related to problems with mysql database. >

2 7

how do i set up outbound on iptel
by Bill Peebles 20 Jul '11

20 Jul '11

i really want to find out how. I dont have a home phone.

2 1

Project conference in Berlin, Sep 02, 2011
by Daniel-Constantin Mierla 20 Jul '11

20 Jul '11

Hello, the details for organizing the 10 years SER conference in Berlin were settled. After the initial announcement the interest was quite high, by now we have a good number of attendees and speakers. We are still open for new speakers, with the target of showing use cases of Kamailio or SER. The event will be hosted by FhG Fokus institute, where SER project was started. It will be a day full of presentations, followed by an evening social networking event. We hope to be able to provide the evening event for free to everyone attending, for that reason we look for sponsorships - if you are interested to sponsor, contact us and we will provide more details about available options. Registration is now open for attendees. Note that the number of seats is limited, don't wait for the last minute to register. More details, including the registration form can be accessed on the event's web page: * http://sip-router.org/10-years-ser/ Looking forward to meeting you in Berlin, enjoy a great summer till then, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Pipelimit Table
by Nathaniel L Keeling 20 Jul '11

20 Jul '11

I have installed Kamailio 3.1.0 and I am trying to test the pipelimit module with postgresql. I ran the kamdbctl to create the tables but it did not create the default table for pipelimit. Is there a script someplace that I am missing or do I need to create my own script to create the tables? I have search the directories and did not find a script for the tables. Thanks Nathaniel L Keeling

2 1

Change aliases table without a restart of Kamailio
by Morten Isaksen 19 Jul '11

19 Jul '11

Hi, Is there any way to reload the content of the aliases table without restarting Kamailio? I have this in my cfg file if (lookup("location")) { xlog("L_WARN", "Callee is online - - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(TO_USER_SIPCHECKS); route(RELAY); } else if (lookup("aliases")) { xlog("L_WARN", "Callee is in aliases - - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(TO_USER_SIPCHECKS); route(RELAY); } else { xlog("L_WARN", "Callee is offline - - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); t_newtran(); t_reply("486", "Busy here"); exit; } And changes in the aliases table does is ignored until I restart Kamailio. -- Morten Isaksen

2 5

Syntax error - custom parameter
by Skyler 19 Jul '11

19 Jul '11

Hi all, I'm getting this error when using custom parameters and I don't why. when I use #!define WITH_ASTERISK the script throws the below error and when I remove the #!define WITH_ASTERISK the script loads without errors. I'm running kamailio-3.1.4-2.1.i386.rpm on Centos 5.6 Does anyone know why this might be happening? parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 16: syntax error 0(2907) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 16: unknown config variable 0(2907) : <core> [cfg.y:3409]: parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 17-19: ERROR: bad config file (3 errors) kamailio.cfg::: lines 122-127 #!ifdef WITH_ASTERISK asterisk.bindip = "127.0.0.1" desc "Asterisk IP Address" asterisk.bindport = "5070" desc "Asterisk Port" kamailio.bindip = "209.15.226.21" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif # This follows as described here: http://by-miconda.blogspot.com/2010/04/kamailio-30x-dyk-07-custom-cfg.html Thanks in advance

2 2

Kamailio as proxy or b2bua?
by Eric Chamberlain 19 Jul '11

19 Jul '11

Hello, My experience has been with Asterisk, but for what I need to do, I think Kamailio may be a better solution. We currently log inbound caller ID information for voip calls (voip app posts call info to a URL). Our users have to fork calls to a custom SIPURI and we log the call information for them. We would like to be able to accommodate users that can't fork calls, and have to register with a provider (PBX or VoIP provider). We see two ways to accomplish this: 1) Act as a b2bua, the user would register with our server and our server would register with their upstream VoIP service (PBX or provider). or 2)Act as a proxy and modify the REGISTER request, such that incoming call signaling traffic goes through our proxy and we forward the signaling traffic on to the user's device. Can Kamailio easily do either of these options? Asterisk realtime doesn't support realtime changes of remote registration information, can Kamailio? Would proxying in such a manner be supported my most voip devices and service providers? -- Eric Chamberlain

2 1

siremis 2.0: generates CDRs
by Klaus Darilion 19 Jul '11

19 Jul '11

Hi! Seems like I am blind. There is a stored procedure which generates CDRs from acc table entries - but how/when does this stored procedure gets triggered? Thanks Klaus

2 5

MYSQL Compilation Kamailio 3.1.4
by Omar 16 Jul '11

16 Jul '11

bash-3.00# make clean bash-3.00# make quiet=0 /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c km_db_mysql.c -o km_db_mysql.o -MMD -MP /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_uri.c -o my_uri.o -MMD -MP /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_res.c -o my_res.o -MMD -MP /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/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 -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_cmd.c -o my_cmd.o -MMD -MP In file included from ../../globals.h:37, from ../../pt.h:42, from ../../counters.h:43, from mysql_mod.h:39, from my_cmd.c:40: ../../ip_addr.h:86: error: field 'sin6' has incomplete type ../../ip_addr.h: In function 'ip_addr_loopback': ../../ip_addr.h:283: warning: implicit declaration of function 'htonl' ../../ip_addr.h:286: warning: implicit declaration of function 'IN6_IS_ADDR_LOOPBACK' ../../ip_addr.h: In function 'sockaddr2ip_addr': ../../ip_addr.h:353: error: dereferencing pointer to incomplete type ../../ip_addr.h: In function 'su_getport': ../../ip_addr.h:399: warning: implicit declaration of function 'ntohs' ../../ip_addr.h: In function 'su_setport': ../../ip_addr.h:418: warning: implicit declaration of function 'htons' make: *** [my_cmd.o] Error 1 the onyl thing i did is uncomment the CROSSCOMPILE line regards O

4 10

Re: [SR-Users] sr-users Digest, Vol 74, Issue 41
by Henrik Aagaard Sørensen 15 Jul '11

15 Jul '11

On Fri, Jul 15, 2011 at 12:00 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. dbaliases for ring groups, possible? (Henrik Aagaard S?rensen) > 2. Re: dbaliases for ring groups, possible? (Alex Balashov) > 3. Re: dbaliases for ring groups, possible? (Morten Isaksen) > 4. Re: xcap server - http load?balancer: xcap authentication > problem (laura testi) > 5. Re: bypass rtp traffic. (MingHon) > 6. Re: AVPs missing in onreply_route (Sebastian Damm) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 15 Jul 2011 09:41:16 +0200 > From: Henrik Aagaard S?rensen <henrikaagaardsorensen(a)gmail.com> > Subject: [SR-Users] dbaliases for ring groups, possible? > To: sr-users(a)lists.sip-router.org > Message-ID: > <CAGH8SebPV=xz1n6Jp=M5fjnUhK0t=Za2aWd_jrpzbjdpdejatw(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > I've read somewhere about using dbaliases for ring groups. So I can call > one > number and several different numbers will then be called. The first to pick > up the call gets it. > > I've tried to an alias in my dbaliases table in MySQL, but I can only make > it possible to call different numbers for the same user. In a way the other > way around then what I thought. > > Is it somehow possible to call one number and then having Kamailio call > several different numbers /users? > > As I'm very newbee to Kamailio it would be nice with a more detailed > answer, > then perhaps just "Yes, use the module ...". >

2 1

Dispatcher OPTIONS ping not seen in onsend_route
by Olle E. Johansson 15 Jul '11

15 Jul '11

Hello! I noticed that the dispatcher's OPTIONs messages are not seen in onsend_route, even though I see responses in the onreply_route. Any reason why I can't see these in onsend_route? (Kamailio 3.1) /O

2 1

dbaliases for ring groups, possible?
by Henrik Aagaard Sørensen 15 Jul '11

15 Jul '11

I've read somewhere about using dbaliases for ring groups. So I can call one number and several different numbers will then be called. The first to pick up the call gets it. I've tried to an alias in my dbaliases table in MySQL, but I can only make it possible to call different numbers for the same user. In a way the other way around then what I thought. Is it somehow possible to call one number and then having Kamailio call several different numbers /users? As I'm very newbee to Kamailio it would be nice with a more detailed answer, then perhaps just "Yes, use the module ...".

3 2

Re: [SR-Users] sr-users Digest, Vol 74, Issue 38
by Henrik Aagaard Sørensen 14 Jul '11

14 Jul '11

On Thu, Jul 14, 2011 at 9:51 AM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: memory requirements for a registrar server > (Henning Westerholt) > 2. Re: kamailio - proxy/relay (Sunny) > 3. Syntax error - custom parameter (Skyler) > 4. Syntax error - custom parameter (Skyler) > 5. Re: kamailio - proxy/relay (I?aki Baz Castillo) > 6. Re: topoh module - query about operation (Pete Kelly) > 7. Re: Call several different subscribers, first to pickup gets > the call. (Klaus Darilion) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 14 Jul 2011 00:27:18 +0200 > From: Henning Westerholt <Henning.Westerholt(a)1und1.de> > Subject: Re: [SR-Users] memory requirements for a registrar server > To: Fabian Borot <fborot(a)hotmail.com> > Cc: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> > Message-ID: > <D3B36E097FE87E44A5CDE7C1AB61682C3CED2EC9(a)EXCHANGE03.webde.local> > Content-Type: text/plain; charset="us-ascii" > > Fabian wrote: > > ok, great...this is, I think, the relevant section on config.h > > Hi Fabian, > > > #define PKG_MEM_SIZE 4 > > #define PKG_MEM_POOL_SIZE PKG_MEM_SIZE*1024*1024 /*!< used only if > PKG_MALLOC is defined*/ > > #define SHM_MEM_SIZE 32 /*!< used if SH_MEM is > defined*/ > > > > > > SHM_MEM_SIZE 32 --> this looks like 32 MB > > and PKG_MEM_SIZE 4 looks like Bytes, that is why the PKG_MEM_POOL_SIZE is > multiplied by 1024*1024 to get MBs I assume. > > Correct, 4MB and 32MB. I'll have a look to to the file to document it a bit > better in the next days. > > You don't need to change config.h to change the shared pool. There exists a > cmd line parameter, > have a look to 'man kamailio' or run 'kamailio -h' for more informations. > The specified values are > just used in the case nothing else is configured. They are also some cfg > script parameters, i think. > > Regards, > > Henning > > > ------------------------------ > > Message: 2 > Date: Wed, 13 Jul 2011 23:39:57 +0100 > From: Sunny <no7find(a)gmail.com> > Subject: Re: [SR-Users] kamailio - proxy/relay > To: miconda(a)gmail.com > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAE_sWtEE4qv5u7bBWyhMSAjMWYB6hKpO6G=_DGyoCK7Jt0OKEA(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > Hi Daniel, > > I was using: > $ru = $(ru{re.subst,/[KAMAILIO]/[IP-PBX ]/}); > t_relay_to_udp("[IP-PBX ]", "5060"); > but your way looks better. > > This works fine in this direction: > [UA] --> [KAMAILIO] --> [IP-PBX ] > > but not in this: > [UA] <-- [KAMAILIO] <-- [IP-PBX ] > because Kamailio is not keeping the location of the UA. > > I want to register UA's on Kamailio, but the authentication is done by > [IP-PBX]... > So Kamailio as Outbound Proxy + Location Server ... any idea on how I can > achieve this configuration? > > Thanks, > Sunny > > > > On 8 July 2011 09:38, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > > Hello, > > > > > > On 7/5/11 8:46 PM, Sunny wrote: > > > >> Hi list, > >> > >> I'm trying to configure kamailio to work as a OBP (Outbound Proxy). > >> > >> Basically KAMAILIO receive requests from UAs, after that it acts like a > >> client and send those requests to the IP-PBX. > >> On the same way the responses are forwarded from IP-PBX to the UAs. > >> > >> > >> [UA] --> [KAMAILIO] --> [IP-PBX ] > >> [UA] <-- [KAMAILIO] <-- [IP-PBX ] > >> > >> Following possible call flow: > >> > >> [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909 > @[KAMAILIO], > >> with session description > >> [KAMAILIO] -> [UA ] SIP Status: 100 Trying > >> [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909 > @[IP-PBX > >> ], with session description > >> [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying > >> [IP-PBX ] -> [KAMAILIO] SIP Status: 407 Proxy Authentication Required > >> [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] > >> [KAMAILIO] -> [UA ] SIP Status: 407 Proxy Authentication Required > >> [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] > >> [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909 > @[KAMAILIO], > >> with session description > >> [KAMAILIO] -> [UA ] SIP Status: 100 Trying > >> [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909 > @[IP-PBX > >> ], with session description > >> [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying > >> [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 183 Session Progress, with > >> session description > >> [KAMAILIO] -> [UA ] SIP/SDP Status: 183 Session Progress, with > >> session description > >> [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 180 Ringing, with session > >> description > >> [KAMAILIO] -> [UA ] SIP/SDP Status: 180 Ringing, with session > >> description > >> (...) > >> [UA ] -> [KAMAILIO] SIP Request: CANCEL sip:12345678909@[KAMAILIO] > >> [KAMAILIO] -> [UA ] SIP Status: 200 OK > >> [KAMAILIO] -> [IP-PBX ] SIP Request: CANCEL sip:12345678909@[IP-PBX ] > >> [IP-PBX ] -> [KAMAILIO] SIP Status: 200 OK > >> [IP-PBX ] -> [KAMAILIO] SIP Status: 487 Request Terminated > >> [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] > >> [KAMAILIO] -> [UA ] SIP Status: 487 Request Terminated > >> [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] > >> > >> Wondering if anyone could help to make this configuration work? > >> > > > > what I understood from above is that you need to update the domain in > R-URI > > from [kamailio] to [ip-pbx]. This is easy to do: > > > > $rd = [ip-pbx]; > > > > You have to do this for initial requests (register, invite). > > > > If you need to update From header URI, look at uac module, > > uac_replace_from(). > > > > Cheers, > > Daniel > > > > -- > > Daniel-Constantin Mierla -- http://www.asipto.com > > Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat > > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >

1 0

Call several different subscribers, first to pickup gets the call.
by Henrik Aagaard Sørensen 14 Jul '11

14 Jul '11

I'm new to Kamailio. Is it possible to have a group of subscribers (with different extensions) that you can call? And the first to pickup the call gets it.

2 1

topoh module - query about operation
by Pete Kelly 14 Jul '11

14 Jul '11

Hi I have been researching the topoh module which I understand is stateless and can persist across a server restart. >From what I can gather it does this by adding some extra, encrypted headers to the INVITE as it passes through the proxy- is this correct? If so, can I ask how it handles situations where the UAS does not resend these headers in subsequent requests? Pete

2 2

Syntax error - custom parameter
by Skyler 13 Jul '11

13 Jul '11

Hi all, I'm getting this error when using custom parameters and I don't why. when I use #!define WITH_ASTERISK the script throws the below error and when I remove the #!define WITH_ASTERISK the script loads without errors. I'm running kamailio-3.1.4-2.1.i386.rpm on Centos 5.6 Does anyone know why this might be happening? parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 16: syntax error 0(2907) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 16: unknown config variable 0(2907) : <core> [cfg.y:3409]: parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 17-19: ERROR: bad config file (3 errors) kamailio.cfg::: lines 122-127 #!ifdef WITH_ASTERISK asterisk.bindip = "127.0.0.1" desc "Asterisk IP Address" asterisk.bindport = "5070" desc "Asterisk Port" kamailio.bindip = "209.15.226.21" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif # This follows as described here: http://by-miconda.blogspot.com/2010/04/kamailio-30x-dyk-07-custom-cfg.html Thanks in advance

1 0

memory requirements for a registrar server
by Fabian Borot 13 Jul '11

13 Jul '11

Hello We are going to use kamailio 3.1.4 for a registrar server and need to know if the default parameters [shared memory and the like] is fine or if we need to plan in advance and increase it at compile time. If so, how should we proceed about it? For subscribers and location type of information, which memory type should we be more concerned about? private memory, shared memory, memory per process etc because it looks like some modules use shared memory others private memory etc. Also , I see that the command "kamctl fifo get_statistics all" gives info about memory used etc. How should we interpret the output like this: shmem:fragments = 24 shmem:free_size = 9665456 shmem:max_used_size = 23915072 shmem:real_used_size = 23888976 shmem:total_size = 33554432 shmem:used_size = 23648072 is this fifo function exported by a specific module? thank you in advance fborot

2 5

kamailio - mssql
by Borin 13 Jul '11

13 Jul '11

Hi, pls do not ask me why I need this..but did anyone try to use Kamailio with mssql or is it at least possible? Best Regards, Kate

3 3

Header field manipulation
by Spencer Thomason 12 Jul '11

12 Jul '11

Hello all, Is it preferable to to a check before removing a header field or to simply call remove_hf()? Are there any performance considerations? i.e. if is_present_hf("X-Account-Code") remove_hf("X-Account-Code"); vs remove_hf("X-Account-Code"); Thanks, Spencer

2 2

siremis 2.0: disable captcha
by Klaus Darilion 12 Jul '11

12 Jul '11

Hi! Is there an easy way to disable the captcha? As siremis is an admin tool, not being public accessible (at least it shouldn't be) I want to get rid of it. Thanks Klaus

2 1

siremis 2.0 statistics problem
by Klaus Darilion 12 Jul '11

12 Jul '11

Hi! I just installed siremis 2.0. I did not let siremis update the SIP DB but did it manually (to sensitive) according to: http://siremis.git.sourceforge.net/git/gitweb.cgi?p=siremis/siremis;a=blob_… But when accessing some statistics, e.g. http://mydomain/siremis/ser/charts_tm/cg=tm I get an error: SELECT time_stamp,tm_active FROM statistics ORDER BY id DESC LIM... --> Unknown column 'tm_active' in 'field list' as statistics table does not have column "tm_active". Am I doing something wrong? Thanks Klaus

2 1

siremis 2.0 documentation
by Klaus Darilion 12 Jul '11

12 Jul '11

Hi! IMO the description "Import Default Data" (Import Default Data - this imports initial data into database 'siremis') is too ambiguous (at least it successfully confused me and I had to dig into the code to find out what it really does). Suggestion: Create Siremis Tables (Create Siremis Tables - this creates the tables in the siremis DB and fills it with inital data). regards Klaus

2 1

app_lua problem setting module parameter
by Bret McDanel 12 Jul '11

12 Jul '11

In 3.1.1 this worked just fine. In 3.1.3 it is giving me an error. I am unsure what to check at this point for why it would be failing. I am doing a kamailio -c to validate the config and not actually launching the program but I do not think that should cause this. modparam("app_lua", "register", "sqlops") sqlops.so is loaded. if I remove that modparam line the config validates, however the lua script would fail because it uses sqlops. modules.lst is the same for building. While I do not think that it is lua specific, both are using liblua5.1.0, one is Ubuntu 11.04 natty (where it fails) and the other is Debian wheezy (where its working). 0(18226) WARNING: <core> [sr_module.c:578]: /usr/local/kamailio/lib64/kamailio/modules/app_lua.so: exports dlflags interface is deprecated and it will notbe supported in newer versions; consider using mod_register() instead 0(18226) : <core> [cfg.y:3412]: parse error in config file kamailio.cfg, line 262, column 41: Can't set module parameter As a side note the "notbe supported ..." line is missing a space between not and be and a newline at the end. -- Trixter aka Bret McDanel website: http://www.0xdecafbad.com pgp key: http://bit.ly/9XYK4b

4 3

rtpproxy_stream2uac
by s.maertens＠telenet.be 12 Jul '11

12 Jul '11

Hi, I am trying to get kamailio (or more specific rtpproxy) to play a wav file when a call is setup In kamailio.cfg I have added the command in the onreply_route rtpproxy_startrecording(); rtpproxy_stream2uas("/var/log/rtpproxy/message","-1"); I am sure the command is executed and given from kamailio to rtpproxy because in syslog I see following output (using dbug output of rtpproxy) rtpproxy[2648]: DBUG:handle_command: received command "P-1 024a73d95bbe016014de647e700 /var/log/rtpproxy/behappy session 487095817;1 024a73d95bbe015f14de647e700;1" rtpproxy[2648]: DBUG:doreply: sending reply "E6#012" rtpproxy[2648]: DBUG:handle_command: received command "R 024a73d95bbe016014de647e700 487095817 024a73d95bbe015f14de647e700" rtpproxy[2648]: DBUG:doreply: sending reply "0#012" The "R" command is the startrecording command and that is indeed working perfect. (recording to /var/log/rtpproxy) The "P" command is the "Playing or stream2uas" command. Seems that the message rtpproxy gets differs a bit from the one I found in the rtpproxy manual from the manual : P[args] callid play_name codecs from_tag to_tag Direction of the playback is defined by the order of the from_tag and to_tag parameters. R callid from_tag to_tag I have used the command " makeann behappy.wav behappy" to convert a wavfile. It gave me a .0 and .a file which I have placed in the /var/log/rtpproxy directory (makeann comes with the rtpproxy source and is rather a bit shy with information to say the least ;) ) There is not too much information or examples found about this so I was hoping maybe somebody on this list could help me in finding out why i'm not getting more information in syslog and why i'm not hearing the audiofile. Best regards

2 3

transformation and PVAR
by Vitaliy Aleksandrov 11 Jul '11

11 Jul '11

>From the transformations documentation: >>{s.strip,len} >>Return string after removing starting 'len' characters. Parameter 'len' can be positive integer or pseudo-variable holding a positive integer. I tried to use pvar holding len but got an error: Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: pv [pv_trans.c:434]: select cannot get p1 Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: <core> [lvalue.c:351]: non existing right pvar This is part of my config file: $avp(varname) = '1'; <---- pseudo variable holding a positive integer $avp(result) = $(fU{s.strip,$avp(varname)}); <---- s.strip How can i use variables with transformation ?

2 1

transformation and pvar
by Vitaliy Aleksandrov 11 Jul '11

11 Jul '11

>From the transformations documentation: > {s.strip,len} > Return string after removing starting 'len' characters. Parameter 'len' can > be positive integer or pseudo-variable holding a positive integer. I tried to use pvar holding len but got an error. Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: pv [pv_trans.c:434]: select cannot get p1 Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: <core> [lvalue.c:351]: non existing right pvar This is part of my config file: $avp(varname) = '1'; <---- pseudo variable holding a positive integer $avp(result) = $(fU{s.strip,$avp(varname)}); <---- s.strip How can i use variables with transformation ?

1 0

Memcached Module
by Spencer Thomason 11 Jul '11

11 Jul '11

Hello all, I'm interested in using the memcached module however it requires libmemcache instead of libmemcached. As libmemcache is no longer supported I was curious if anyone had successfully patched the module to use libmemcached instead. Also as a side note, the provided rpm spec files in the git repo erroneously contains a reference to libmemcached. Thanks, Spencer

2 1

Employment opportunity
by miklos＠iptel.org 10 Jul '11

10 Jul '11

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Salute! I take up a position of HR manager in a large multinational company. Our enterprise is inseparably connected with a great number of various activities, like: - estate property - founding and abatement of enterprises - supporting in opening of banking accounts - etc. [...] Content analysis details: (23.1 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: totaljoblists.net] 1.6 URIBL_WS_SURBL Contains an URL listed in the WS SURBL blocklist [URIs: totaljoblists.net] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: totaljoblists.net] 3.5 BAYES_99 BODY: Bayes spam probability is 99 to 100% [score: 0.9981] 3.2 FH_HELO_EQ_D_D_D_D Helo is d-d-d-d 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [2.60.89.135 listed in zen.spamhaus.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?2.60.89.135>] 1.3 RCVD_IN_RP_RNBL RBL: Relay in RNBL, https://senderscore.org/blacklistlookup/ [2.60.89.135 listed in bl.score.senderscore.com] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [2.60.89.135 listed in bb.barracudacentral.org] 0.7 SPF_SOFTFAIL SPF: sender does not match SPF record (softfail) 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines 1.0 RDNS_DYNAMIC Delivered to internal network by host with dynamic-looking rDNS 0.0 T_URIBL_BLACK_OVERLAP T_URIBL_BLACK_OVERLAP 2.0 HELO_DYNAMIC_IPADDR Relay HELO'd using suspicious hostname (IP addr 1)

1 0

Job offer match, respond to apply
by miklos＠iptel.org 09 Jul '11

09 Jul '11

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Hello! I am a superintendent of Human Resources department of a large international company. Our company takes an active part in the life of its subsidiaries, for instance: \ real estate \ logistics \ private undertaking service \ etc. [...] Content analysis details: (12.1 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [212.164.140.23 listed in zen.spamhaus.org] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: totaljoblists.net] 1.6 URIBL_WS_SURBL Contains an URL listed in the WS SURBL blocklist [URIs: totaljoblists.net] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: totaljoblists.net] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [212.164.140.23 listed in bb.barracudacentral.org] 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines 1.5 BAYES_60 BODY: Bayes spam probability is 60 to 80% [score: 0.6047] 0.0 T_URIBL_BLACK_OVERLAP T_URIBL_BLACK_OVERLAP 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

on_reply and failure route names
by Fabian Borot 08 Jul '11

08 Jul '11

Hi Is it possible to use string names also for on_reply and failure_route blocks? I guess these errors when starting kamailio Jul 8 10:00:25 registrar kamailio: : <core> [cfg.y:3412]: parse error in config file /usr/local/kamailio-3.1//etc/kamailio//kamailio.cfg, line 432, column 41: syntax error Jul 8 10:00:25 registrar kamailio: : <core> [cfg.y:3412]: parse error in config file /usr/local/kamailio-3.1//etc/kamailio//kamailio.cfg, line 432, column 41: '('')' expected (function call) Jul 8 10:00:25 registrar kamailio: : <core> [cfg.y:3412]: parse error in config file /usr/local/kamailio-3.1//etc/kamailio//kamailio.cfg, line 432, column 41: function t_on_reply: parameter 1 is not constant Jul 8 10:00:25 registrar kamailio: : <core> [cfg.y:3412]: parse error in config file /usr/local/kamailio-3.1//etc/kamailio//kamailio.cfg, line 432, column 41: bad command thank you fborot

3 5

following the path a request take in the config file
by Fabian Borot 08 Jul '11

08 Jul '11

Hi Is there a way of tracing the path a request takes as it is being processed by the config file? Aside from logging using xlog(). I am not sure whether I read about this some where on the web , but I am under the impression that starting on version 3.x there was going to be a similar feature. please advise fborot

2 2

Loading multiple htables from db on startup? (Kamailio 1.5)
by Robin Rodriguez 08 Jul '11

08 Jul '11

Hi all, Currently running on 1.5.4-notls and using a single htable that gets loaded at startup ala modparam("htable", "htable", "local_routes=>size=4;autoexpire=0;dbtable=TABLE_A;") modparam("htable", "key_name_column", "NAME_1") modparam("htable", "key_value_column", "VALUE_1") modparam("htable", "fetch_rows", 512) This works great, the problem I've run into is attempting to load a second htable on startup whose data comes from a different table/scheme ala modparam("htable", "htable", "inbound_acl=>size=4;autoexpire=0;dbtable=TABLE_B;") modparam("htable", "key_name_column", "NAME_2") modparam("htable", "key_value_column", "VALUE_2") modparam("htable", "fetch_rows", 512) With both of them in the config file Kamailio fails to start with the error INFO:core:init_mod: initializing module htable ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'NAME_2' in 'field list' 'NAME_2' does not exist in 'TABLE_1' so that seems to suggest that only the first dbtable value is used, the SQL side supports this: select NAME_2,key_type,value_type,VALUE_2 from TABLE_A order by key_type So it uses the first defined database table, but then the last defined key_name and key_value. Is there any way that I've overlooked in the docs or elsewhere about about loading multiple htables from different database tables? I'd like to use this htable in an ACL and would like to avoid database hits though can make it work with avpops if I have to. Any suggestions are greatly appreciated. Thanks, Robin

3 3

Dialplan question
by alex pappas 08 Jul '11

08 Jul '11

Dear all, I have these 3 entries in the dialplan: dpid pr match_exp attr 2 50 ^38.+ porvider_A 2 50 ^3802.+ porvider_B 2 50 ^3806.+ porvider_C with the fifo command : *kamctl fifo dp_translate 2 3806328881* Output:: ATTRIBUTES::* porvider_A* I would expect that the dialplan module would choose the third choice which is the correct one for this call. >From the dialplan documentation i understand that the priority is needed when the prefixes are overlaping in terms of they are the same but with different attrs like : 2 50 ^38.+ porvider_A 2 50 ^3802.+ porvider_B *2 50 ^3806.+ porvider_C* *2 50 ^3806.+ porvider_D* * * here i understand that I should use priorities. Does anyone knnows how this works ?? Thank you in advance Alex * *

2 1

acc table versions
by Klaus Darilion 08 Jul '11

08 Jul '11

Hi! Just as a note: I find it a bit strange that acc (4) and missed_calls (3) have different version numbers although they are identical. Anyway, it seems that acc module does not make version checks at all. regards Klaus

2 1

Celebrating 10 years of SER
by Daniel-Constantin Mierla 07 Jul '11

07 Jul '11

Hello, the 3rd of September will mark the 10 years since the start of SIP Express Router (aka SER - also the origin for kamailio (openser) and now the two are back together). From last year: * http://sip-router.org/wiki/history/ser-9-years This is a good time to celebrate and I am thinking of at least a social event (e.g., dinner) in Berlin, Germany. Based of private discussion so far, other cities are interested in organizing something similar locally, e.g., Barcelona. If you are a group of people that can gather together for a drink in same place to celebrate the event, let me know and I will add in the list to coordinate - hopefully we will be even able connect between sites with some video/audio if the timezones are convenient. I'm looking as well to do a more formal event, something like a day long conference style, on Friday, 2nd of September. Berlin is the natural choice for location, being the place where the project started. I can take care of location when it is clear that there are people willing to travel. Berlin area alone should be able to get about 20 people, among them several long term developers of the project and companies offering SER/Kamailio-based services. Coming over gives you the opportunity to connect directly with them. At this event, along with presenting shortly the history and future plans, the event should be open to everyone that want to present their usage of SER/Kamailio. Attendance is going to be free, in the limit of available seats. If you think it would be interesting and want to attend, let me know. Also, if you have other ideas of how to celebrate, write us back. Best regards, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 2

Re: [SR-Users] Playing with dlg_bridge()
by Stefan Broddesson 07 Jul '11

07 Jul '11

Brodde(a)yahoo.com Sent from my HTC ----- Reply message ----- From: "Roberto Fichera" <kernel(a)tekno-soft.it> To: "Henning Westerholt" <henning.westerholt(a)1und1.de> Cc: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> Subject: [SR-Users] Playing with dlg_bridge() Date: Thu, Jul 7, 2011 16:01 On 07/07/2011 02:51 PM, Henning Westerholt wrote: > On Thursday 07 July 2011, Roberto Fichera wrote: >>>> Do you know what does means the warning below? >>>> [..] >>>> 68.2.50 Jul 6 11:54:17 roberto /usr/sbin/kamailio[20457]: WARNING: >>>> dialog [dlg_handlers.c:884]: unable to find dialog for NOTIFY with >>>> route param '7d2.14447fc4' [727:1291273281] >>> This means that the module can't find an existing (stored) dialog for the >>> routed message in question, in this case a NOTIFY. >> Does this depends by the fact of own the dlg_bridge() works? And in case, >> how can I solve that problem? > Hi Roberto, Hi Henning, > I'm not 100% sure, I'm not the expert for dlg_bridge.. To be more specific > (after discussion with Timo..) - the warning is emmited if the module can't > find the dialog with the ID added as a route parameter to the respective > message. One particiular reason is (if you don't store the dialogs in a > database) a simple kamailio restart, then all existing dialog information is > lost. actually I've the following setup: #!define DBURL "postgres://openser:openserrw@localhost/openser" modparam("dialog", "dlg_flag", 2) modparam("dialog", "bridge_controller", "sip:controller@192.168.2.92") modparam("dialog", "db_url", DBURL ) Actually, my thought is that problem is more strictly related to the dlg_bridge() which might not track the bridged dialog. Anyway, I'm playing with dlg_bridge() because I want to realize the logic so that two peers, and if possible more, can talk together only if authorized. The authorization might be a unique tag or transaction id, stored in a database, and valid only for the give conversation {tag, caller, callee} once started, than the tag become invalid and can't be used again. I really like to have some suggestions about how to realize such logic, so which might be the best way to approach such problem since I'm actually a novice to Kamailio or in general with OpenSer logic. I'm actually more confident with Asterisk logic. > Best regards, > > Henning > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
by MingHon 07 Jul '11

07 Jul '11

Hi List, im facing an issue that my kamailio proxy did not replace the ip address in the invite and 200OK sdp body. my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user my kamailio is listening on 192.168.1.3, also define: advertised_address="175.136.223.112"; & advertised_port=5060; and my asterisk is on 192.168.1.23. sip signalling and rtp port forwarded to kamailio. uacs from another nat register successfully. if i put 2 lines of force_rtp_proxy("fcow","175.136.223.112"); i will get double ip addr in c and o but kamailio ignore my ip addr. example i will get c=IN IP4 192.168.1.3192.168.1.3 here is part of my simple script. hope you can help. thank you very much. ---------------cfg------------------- route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy("fcow","175.136.223.112"); #force_rtp_proxy("fcow","175.136.223.112"); xlog("L_INFO","offer"); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } -------------------------------------- and here is the wireshark for uac INVITE and OK. -----------INVITE----------------- ve0 EE;p9INVITE sip:102@192.168.2.132:5062 SIP/2.0 Record-Route: <sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0 Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080 Max-Forwards: 69 From: "101" <sip:102@aextddns.dyndns.info>;tag=as032358a3 To: <sip:102@192.168.1.3:5060> Contact: <sip:102@192.168.1.23:5080> Call-ID: 416f6e09674ae9671bb7144a1cb11137(a)aextddns.dyndns.info CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.18 Date: Tue, 05 Jul 2011 07:20:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 327 v=0 o=root 1639709788 1639709788 IN IP4 192.168.1.3 s=Asterisk PBX 1.6.2.18 c=IN IP4 192.168.1.3 t=0 0 m=audio 10072 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes -----------200OK--------------- e90 ElE;pX4tSIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.200:5062 ;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416 Record-Route: <sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes> From: "101" <sip:101@aextddns.dyndns.info>;tag=1796959074 To: <sip:102@aextddns.dyndns.info>;tag=as2e4c0125 Call-ID: 1985782590(a)192.168.2.200 CSeq: 21 INVITE Server: Asterisk PBX 1.6.2.18 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:102@192.168.1.23:5080> Content-Type: application/sdp Content-Length: 286 v=0 o=root 403900934 403900934 IN IP4 192.168.1.23 s=Asterisk PBX 1.6.2.18 c=IN IP4 192.168.1.23 t=0 0 m=audio 14420 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------ My kamailio log. -----------LOG------------------ DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3 INFO: <script>: offer ------------------------------------- double force_rtp_proxy --------kamailio -> asterisk [INVITE]--------- Pyi-}E7V@:#pINVITE sip:102@aextddns.dyndns.info SIP/2.0 Record-Route: <sip:192.168.1.3;lr=on;ftag=640933430;nat=yes> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0 Via: SIP/2.0/UDP 192.168.2.200:5062 ;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648 From: "101" <sip:101@aextddns.dyndns.info>;tag=640933430 To: <sip:102@aextddns.dyndns.info> Call-ID: 1909950509(a)192.168.2.200 CSeq: 21 INVITE Contact: <sip:101@175.138.21.31:2788> Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 69 User-Agent: T20 9.41.0.80 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 334 v=0 o=20073 20073 IN IP4 192.168.1.3192.168.1.3 s=SDP data c=IN IP4 192.168.1.3192.168.1.3 t=0 0 m=audio 1006410064 RTP/AVP 0 8 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=nortpproxy:yes a=nortpproxy:yes -----------LOG------------------ DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3 DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3 INFO: <script>: offer -----------LOG------------------ -- Regards, MingHon

3 19

Playing with dlg_bridge()
by Roberto Fichera 07 Jul '11

07 Jul '11

Hi All in the list, I want to implement a click and dial feature using the xhttp interface but also keeping working the XCAP functionality. Than I added some code about the dialog module (it's my first time I try it) so that the URL look like http://localhost:5060/sip-bridge/<from>/<to> #!ifdef WITH_XCAPSRV #!define WITH_XHTTPAUTH event_route[xhttp:request] { xdbg("===== xhttp: request [$rv] $rm => $hu\n"); if($hu=~"^/xcap-root/") { #!ifdef WITH_XHTTPAUTH if (!www_authorize("xcap", "subscriber")) { www_challenge("xcap", "0"); exit; } #!endif set_reply_close(); set_reply_no_connect(); # xcap ops $xcapuri(u=>data) = $hu; if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:" + $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; xlog("===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); if($xcapuri(u=>auid)=="xcap-caps") { $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "ok", "application/xcap-caps+xml", "$var(xbody)"); exit; } #!ifdef WITH_XHTTPAUTH # be sure auth user access only its documents if ($au!=$(var(uri){uri.user})) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>$si:$sp</body></html>"); exit; } #!endif switch($rm) { case "PUT": xcaps_put("$var(uri)", "$hu", "$rb"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } exit; break; case "GET": xlog("===== xhttp: get $var(uri) => $hu\n"); xcaps_get("$var(uri)", "$hu"); exit; break; case "DELETE": xcaps_del("$var(uri)", "$hu"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp del: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } exit; break; } } else if($hu=~"^/sip-bridge/") { if ( ! ( src_ip == 127.0.0.1 ) ) { xhttp_reply( "403", "Forbidden", "text/html", "<html><body>Invalid Access</body></html>" ); exit; } $var(uri) = $hu; $var(from) = $(var(uri){s.select,2,/}); $var(to) = $(var(uri){s.select,3,/}); xdbg( "=== GOT sip-bridge between $var(caller) => $var(callee) ===" ); *dlg_bridge( $var(from), $var(to), "sip:192.168.2.92:5060" );* } # http ops xhttp_reply("200", "ok", "text/html", "<html><body>OK: $si:$sp</body></html>"); exit; } #!endif but I got the error below: Not starting : invalid configuration file! 0(22256) : <core> [cfg.y:3409]: parse error in config file //etc/kamailio/kamailio.cfg, line 986, column 17-26: function dlg_bridge: parameter 1 is not constant 0(22256) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 986, column 63: bad command ERROR: bad config file (2 errors) so my question is how can I call the dlg_bridge() passing some parameters coming from the http request? Thanks in advance. Roberto Fichera.

2 6

Problem with worst case scenario of accessing 15 gateways over the DISPATCHER module - MAX_BRANCHES is reached
by Klaus Feichtinger 07 Jul '11

07 Jul '11

Hi, I have a problem in my scenario where I use the dispatcher module for (round-robin based) distribution of incoming calls to 15 (!) different gateways. In idle case (= all gateways are up and running or just a couple of them does not accept calls) my solution is working fine. The solution is very primitive and based on the example of the dispatcher module's ReadMe. However, when e.g. 12 of 15 gateways are out of service (in detail: rejecting the call with "480 - Temporarily Unavailable") and only the 13th would be okay, I can not reach that gateway. This is because the TM module is interrupting the distribution with an error message indicating that the maximum number of branches (per transaction) is reached (ERROR:/tm/:/add_uac/: /maximum number of branches/ exceeded).... My first idea for solving this problem was/is in manipulating the definition of the "max_branches" constant in the config.h file. However, I prefer using the precompiled Debian packages. Therefore that solution is not optimal. I have not found a method for decreasing the destination_set on an easy way... Does anybody know an alternative solution / workaround? Thanks in advance, Klaus

3 6

new to ser
by Pradeep Sharma 07 Jul '11

07 Jul '11

Hi, I am installed ser 0.9.6 onto my mac machine and with following command i get it to listen to 5060 port. but when i use a sip client and send register message i get a 501 not implemented as a response from the server. i am attaching ser.cfg. your help is appreciated. -- Thanks Paul

2 1

updates to config preprocessor
by Daniel-Constantin Mierla 07 Jul '11

07 Jul '11

Hello, just to let you know about some recent additions regarding the config file preprocessor, hopefully they will help you keep the config cleaner and easy to maintain. - you can use either #! or !! as the start marker of preprocessor directive (to make happy those complaining about not able to distinguish quickly from comments) - subsdef was added some time ago to allow defining a token that will replace ID tokens as well as update inside string values. It has the same format as #!subst. For example:| #!substdef "!ABC!XYZ!g"| is pretty much same as: #!define ABC XYZ #!subs "/ABC/XYZ/" - substdefs, like substdef byt the defined value will be added enclosed in double quotes: |#!substdef "!ABC!XYZ!g"| is pretty much same as: #!define ABC "XYZ" #!subs "/ABC/XYZ/" - trydef - similar to define, but if the ID is defined already, ignore this time - redef - similar to define, but if the ID is defined, redefine its value (note that define itself throws error if the ID is already defined) Regarding the inclusion of files, a new option is available, 'import_file'. It is similar to 'include_file', but if the file is not found, it does not throw error. Bot include_file and import_file can be started as well with #! or !!. You can still use them without #! or !!, whatever you like more. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

RFC3311 troublefix
by Andrew O. Zhukov 06 Jul '11

06 Jul '11

Please add in all samples, default installed /etc/*.cfg files etc, etc, etc... RFC3311 processinging in a branch route: } else if (is_method("INVITE|UPDATE")){ force_rtp_proxy(); instead of } else if (is_method("INVITE")){ force_rtp_proxy(); Or in last cvs version sample if (nat_uac_test("3")) { .... if ( is_method("INVITE") ) { fix_nated_sdp("1"); # Add direction=active to SDP }; is_method("INVITE|UPDATE"); Thanks in advance !!!

4 6

strange behavior with pseuvars not being found
by Bret McDanel 06 Jul '11

06 Jul '11

I am in the process of migrating from opensips 1.4 to kamailio 3.2 (GIT current as of this morning). I think I have the config fully ported but I appear to be missing something. I get the following error unknown script pseudo variable $cl I also get the same error for $rU and some others. I wanted to see if this is a common problem that generates this error before the config has to be looked at, ie made public. I am making heavy use of #!define and #!ifdef statements, and from what I can tell I am not missing any curly braces or #!endifs. Thanks for any pointers, -- Trixter aka Bret McDanel website: http://www.0xdecafbad.com pgp key: http://bit.ly/9XYK4b

2 1

auth_db question
by Fabian Borot 06 Jul '11

06 Jul '11

Hello I am trying to store the password on the subscriber db as encrypted instead of plain text and when I do it then the registration fails, modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "ha1") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) If I change the 3rd line to this it works but it shows the password in clear text: modparam("auth_db", "password_column", "password") This is how my subscriber table is defined mysql> describe subscriber; +---------------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +---------------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | username | varchar(64) | NO | MUL | | | | domain | varchar(64) | NO | | | | | password | varchar(25) | NO | | | | | email_address | varchar(64) | NO | | | | | ha1 | varchar(64) | NO | | | | | ha1b | varchar(64) | NO | | | | | rpid | varchar(64) | YES | | NULL | | +---------------+------------------+------+-----+---------+----------------+ and this is how it looks like in the db: [root@registrar ~]# /usr/local/kamailio-3.1/sbin/kamctl db show subscriber +----+----------+---------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+---------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | 100 | 192.168.169.1 | abc | | 4a169916d58247f9e7eac55251fa7467 | 28787e5b5d3a2c6e2e0a80d3e9e3c6de | NULL | | 2 | 200 | 192.168.169.1 | abc | | 0b15f8f9e348ec64a2e7d5fca6b629c3 | 99eea8b456956d67e7915ed9c96e420b | NULL | +----+----------+---------------+----------+---------------+----------------------------------+----------------------------------+------+ please give me hand figuring this one out, I can't see what I am doing wrong fborot

2 4

kamailio fails to start with kamctl start
by Fabian Borot 06 Jul '11

06 Jul '11

Hello I just installed kamailio-3.1 and using the default config file "kamctl start" fails [root@registrar sbin]# ./kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed [root@registrar sbin]# but when I use this command it starts: <kamailio -f path_to_config_file [root@registrar sbin]# ./kamailio -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg loading modules under /usr/local/kamailio-3.1//lib64/kamailio/modules_k/:/usr/local/kamailio-3.1//lib64/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: 192.168.168.10:5060 tcp: 127.0.0.1:5060 tcp: 192.168.168.10:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: registrar:5060 tcp: registrar.mytestdomain.com:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: registrar:5060 udp: registrar.mytestdomain.com:5060 [root@registrar sbin]# Also, when I use "./kamctl start" the syslog file does not show anything at all that would help me why it is failing. But when I use "/usr/local/kamailio-3.1/sbin/kamailio -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg" it does update the syslog file. Also, after kamailio start with "./kamailio -f .." I can use ./kamctl fifo <any option>. Example: [root@registrar sbin]# ./kamctl fifo version Server:: kamailio (3.1.4 (x86_64/linux)) Build:: mi_core.c compiled on 15:09:40 Jun 24 2011 with gcc 4.1.2 Flags:: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES GIT:: e2d6f0 [root@registrar sbin]# [root@registrar sbin]# ./kamctl fifo ps Process:: ID=0 PID=27078 Type=attendant Process:: ID=1 PID=27079 Type=udp receiver child=0 sock=127.0.0.1:5060 Process:: ID=2 PID=27080 Type=udp receiver child=1 sock=127.0.0.1:5060 Process:: ID=3 PID=27081 Type=udp receiver child=2 sock=127.0.0.1:5060 Process:: ID=4 PID=27082 Type=udp receiver child=3 sock=127.0.0.1:5060 Process:: ID=5 PID=27083 Type=udp receiver child=0 sock=192.168.168.10:5060 Process:: ID=6 PID=27084 Type=udp receiver child=1 sock=192.168.168.10:5060 Process:: ID=7 PID=27085 Type=udp receiver child=2 sock=192.168.168.10:5060 Process:: ID=8 PID=27086 Type=udp receiver child=3 sock=192.168.168.10:5060 Process:: ID=9 PID=27087 Type=slow timer Process:: ID=10 PID=27088 Type=timer Process:: ID=11 PID=27089 Type=MI FIFO Process:: ID=12 PID=27090 Type=ctl handler Process:: ID=13 PID=27091 Type=tcp receiver child=0 Process:: ID=14 PID=27092 Type=tcp receiver child=1 Process:: ID=15 PID=27093 Type=tcp receiver child=2 Process:: ID=16 PID=27094 Type=tcp receiver child=3 Process:: ID=17 PID=27095 Type=tcp main process [root@registrar sbin]#

2 6

Muenster germany next week
by Olle E. Johansson 06 Jul '11

06 Jul '11

Friends, Going to be in Muenster next week. If anyone is nearby, I'll be happy to take part in a VoIP dinner :-) /O

1 0

Kamailio 3.1.4 Compilation Error ('TMCB_DONT_ACK' undeclared)
by Bruce McAlister 05 Jul '11

05 Jul '11

Hi All, Would someone be able to point me in the right direction here, I am trying to compile Kamailio 3.1.4 from git checkout and I am ending up with the following error: gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' -DVERSION='"3.1.4"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=3001004 -DCFG_DIR='"/usr/local/loadbalancer/lb1/etc/"' -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 -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 -DUSE_RAW_SOCKS -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -DOPENSER_MOD_INTERFACE -DMOD_NAME='"seas"' -c seas_action.c -o seas_action.o -MMD -MP seas_action.c: In function 'ac_uac_req': seas_action.c:1097: error: 'TMCB_DONT_ACK' undeclared (first use in this function) seas_action.c:1097: error: (Each undeclared identifier is reported only once seas_action.c:1097: error: for each function it appears in.) make[1]: *** [seas_action.o] Error 1 make: *** [modules_k] Error 1 I am fairly certain I have all required -devel packages installed, but I just cannot see where that identifier is meant to be defined. Any tips/suggestions would be greatly appreciated. If you require more info, please let me know what you need. thanks Bruce

4 5

job offer: System administrator VoIP
by Henning Westerholt 05 Jul '11

05 Jul '11

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional IT operation position open in our team in Karlsruhe, Germany at the moment. You would be responsible in a team of motivated system administrators for the operation and maintenance of the VoIP kamailio backend at 1&1, providing services for more than three million customers. You should have some experience in operating or developing Voice over IP services with kamailio, asterisk or others similar servers. Additionally you should be able to communicate professionally in the german language. For more informations please contact me directly by email. Best regards, Henning Westerholt -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany

1 0

more on asynchronous processing in config
by Daniel-Constantin Mierla 05 Jul '11

05 Jul '11

Hello, over the weekend I exported to config file the two main functions from tm that allow to suspend the execution of a SIP request and resume it in another place. They are exported through tmx module: - t_suspend() - will suspend the execution of current sip request and then you can do something else (like processing the next request) - t_continue(tindex, tlabel, rtname) - continue the processing of SIP request suspended in the transaction identified by tupple (tindex,tlable) with the actions in route[rtname] After t_suspend() you can get the transaction identifier (tindex, tlable) via pseudo-variables $T(id_index) and $T(id_label), e.g.: if(t_suspend()) { xlog("sip request suspended in transaction [$T(id_index)/$T(id_label)]\n"); exit; } One example of usage can be passing the $T(id_index) and $T(id_label) via mqueue to another process for further handling. So, besides the new additions, now following cfg tools can be used to do asynchronous operations directly via config: - the mqueue + rtimer module - pass a task via mqueue and get it in rtimer specific processes to run it further - async module - intended to collect asynchronous specific implementation, for now able to sleep and execute a specific route after a time interval Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 2

dbtext module
by Bruno Bresciani 05 Jul '11

05 Jul '11

Hi All, I'm using kamailio with database in text mode and i'd like to know if dbtext module has support the following parameters to search: ilike; limit; offset; Regards

3 3

"select count(*)" with avp_db_query always returns null
by Andrade Ricardo (CI/AFU1) 05 Jul '11

05 Jul '11

Hi list, Perhaps this fits better into a bug report, but I'd like to know if somebody out there experienced a similar issue. I am executing a "select count" query using the avp_db_query function, but it is not storing the results in any avp. Other queries are working fine. I use db_mysql connected with a mysql 5.1 server. I have tested this with kamailio-3.1.0 and kamailio-3.1.3, both didn't work. In an old box (version 1.3.x), the same query was returning the correct value. --------------------------------------------------- Here is the case which is not working: Code: avp_delete("$avp(s:count)"); $var(ret) = avp_db_query("SELECT count(*) FROM subscriber where username='foo'", "$avp(s:count)"); xlog("L_INFO", "var(ret)=$var(ret) avp(s:count)=$avp(s:count)"); Output: INFO: <script>: var(ret)=1 avp(s:count)=<null> (notice the return code 1, which means that the query was successfull and there should be some output value stored in the avp, but it is <null>) --------------------------------------------------- And here is one case which works as expected: $var(ret) = avp_db_query("SELECT id FROM subscriber where username='foo'", "$avp(s:count)"); Output: INFO: <script>: var(ret)=1 avp(s:count)=11 --------------------------------------------------- Thanks in advance for any feedback. Best regards, Ricardo Andrade.

4 8

two uac behind same nat one uac remain Cflag:: 0
by MingHon 05 Jul '11

05 Jul '11

Hi List, both uac register successfully but the issue is the first uac who register the Cflag remain 0. im using nat_uac_test("19"). thank you in advance. here is the ul show: AOR:: 102 Contact:: sip:102@175.136.221.60:5062 Q= Expires:: 103 Callid:: 678936531(a)175.136.221.60 Cseq:: 2 User-agent:: T22 7.3.0.50 Received:: sip:175.136.221.60:1056 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:60.53.82.184:5060 Methods:: 16383 AOR:: 103 Contact:: sip:103@175.136.221.60:5062 Q= Expires:: 62 Callid:: 1621958925(a)175.136.221.60 Cseq:: 3 User-agent:: Yealink SIP-T18 18.0.0.70 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:60.53.82.184:5060 Methods:: 16383 and here is the wireshark on kamailio for ua 103 and 102. E@< \<<5Rw REGISTER sip:aextddns.dyndns.info SIP/2.0 Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK2080454766 From: "103" <sip:103@175.136.221.60>;tag=681558690 To: "103" <sip:103@aextddns.dyndns.info> Call-ID: 1621958925(a)175.136.221.60 CSeq: 2 REGISTER Contact: <sip:103@175.136.221.60:5062> Authorization: Digest username="103", realm="aextddns.dyndns.info", nonce="Tgp7mU4Kem3Qqyd66hwhWgmU8bNpL463", uri="si$ Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T18 18.0.0.70 Expires: 3600 Content-Length: 0 E@<5R<PSIP/2.0 200 OK Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK2080454766;rport=5062 From: "103" <sip:103@175.136.221.60>;tag=681558690 To: "103" <sip:103@aextddns.dyndns.info >;tag=cbf8071f5b02e3e2fd329ee5d72c3f65.d24e Call-ID: 1621958925(a)175.136.221.60 CSeq: 2 REGISTER Contact: <sip:103@175.136.221.60:5062>;expires=120 Server: kamailio (3.1.4 (i386/linux)) Content-Length: 0 -------------------------------------------------- E@< j<<5R 7REGISTER sip:aextddns.dyndns.info SIP/2.0 Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK1080036976 From: "102" <sip:102@175.136.221.60>;tag=544575245 To: "102" <sip:102@aextddns.dyndns.info> Call-ID: 678936531(a)175.136.221.60 CSeq: 2 REGISTER Contact: <sip:102@175.136.221.60:5062> Authorization: Digest username="102", realm="aextddns.dyndns.info", nonce="Tgp8AU4KetWGb79MEU5MUHq+xA4xy+OW", uri="si$ Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: T22 7.3.0.50 Expires: 3600 Content-Length: 0 E@<5R< {SIP/2.0 200 OK Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK1080036976;rport=1056 From: "102" <sip:102@175.136.221.60>;tag=544575245 To: "102" <sip:102@aextddns.dyndns.info >;tag=cbf8071f5b02e3e2fd329ee5d72c3f65.5cb8 Call-ID: 678936531(a)175.136.221.60 CSeq: 2 REGISTER Contact: <sip:102@175.136.221.60:5062>;expires=120;received="sip: 175.136.221.60:1056" Server: kamailio (3.1.4 (i386/linux)) Content-Length: 0 -- Regards, MingHon

2 2

Hello,can you send me the useage of siremis?Thank you
by 吴明 05 Jul '11

05 Jul '11

HI: I'm a user of siremis from China.Now I install the siremis successfully.But I would not use it. So I want the useage of it. I'll wait for your letter.Thank you! Best regards. Calvin wu 2011-6-24

2 1

Kamailio 64bit Compile Errors
by Nathaniel L Keeling 05 Jul '11

05 Jul '11

I am trying to install Kamailio 3.1.0 on Solaris 10 that is a sparc machine (v120) and is running a 64bit OS. When I originally performed the compile utilizing the instructions from the website, I was getting warnings about using 32 bit headers. I was able to compile and install successfully, but when I started Kamailio, loading of modules were failing due to "not found" error. The warning messages: "atomic/atomic_sparc.h:41:2: warning: #warning "sparc32 atomic operations support not tested"" I thought was because the target architecture specified sparc and not sparc64, so I tried specified the target architecture as listed below which got rid of the warning messages but then I got ELFCLASS32 error messages. What is the proper way to create a 64bit version of Kamailio on a sparc Solaris 10. It seems like the target architecure is not being configured properly and that may be causing other problems. I have included output from different commands about the environment and the commands that I entered to compile Kamailio. # make FLAVOUR=kamailio cfg target architecture <sparc>, host architecture <sparc64> # isainfo -n sparcv9 # uname -p sparc # uname -m sun4u # isainfo -v 64-bit sparcv9 applications vis 32-bit sparc applications vis v8plus div32 mul32 # isainfo -b 64 Here is what I entered to try and force the building of a 64 bit version and a sample error that I was getting. make FLAVOUR=kamailio prefix=/usr/local/kamailio-3.1 CPU=ultrasparc ARCH=sparc64 cfg LD (gcc) [M lcr.so] lcr.so ld: fatal: file /usr/local/lib/libpcre.so: wrong ELF class: ELFCLASS32 ld: fatal: File processing errors. No output written to lcr.so collect2: ld returned 1 exit status make[1]: *** [lcr.so] Error 1 make: *** [modules] Error 1 LD_LIBRARY_PATH=/lib/sparcv9:/usr/lib/sparcv9:/usr/sfw/lib/sparcv9:/usr/local/postgres/9.0-pgdg/lib/64 LD_RUN_PATH=/lib/sparcv9:/usr/lib/sparcv9:/usr/sfw/lib/sparcv9:/usr/local/postgres/9.0-pgdg/lib/64 PATH=/opt/csw/bin:/usr/sfw/bin:/usr/local/bin:/usr/local/postgres/9.0-pgdg/bin:/usr/sbin:/usr/bin:/usr/ccs/bin: # find . -name libpcre.so -print ./usr/local/lib/libpcre.so Thanks

2 1

Playing with XCAP_SERVER and Postgresql
by Roberto Fichera 05 Jul '11

05 Jul '11

Hi all in the List, I'm playing with the xcap_server using the jitsi client. Kamailio is v3.1.4 and postgresql is v8.4.8 under Fedora 14 64bit. Basically I'm getting the error below. Jul 5 10:36:36 roberto /usr/sbin/kamailio[8529]: ERROR: <script>: ===== xhttp: xcap-caps : <null> Jul 5 10:36:36 roberto /usr/sbin/kamailio[8530]: ERROR: <script>: ===== xhttp: resource-lists : sip:roberto@192.168.2.92 Jul 5 10:36:36 roberto /usr/sbin/kamailio[8530]: ERROR: <script>: ===== xhttp: get sip:roberto@192.168.2.92 => /xcap-root/resource-lists/users/sip:roberto@192.168.2.92/index Jul 5 10:36:36 roberto /usr/sbin/kamailio[8531]: ERROR: <script>: ===== xhttp: pres-rules : sip:roberto@192.168.2.92 Jul 5 10:36:36 roberto /usr/sbin/kamailio[8531]: ERROR: <script>: ===== xhttp: get sip:roberto@192.168.2.92 => /xcap-root/pres-rules/users/sip:roberto@192.168.2.92/presrules Jul 5 10:36:36 roberto /usr/sbin/kamailio[8521]: INFO: presence [notify.c:1588]: NOTIFY sip:roberto@192.168.2.92 via sip:roberto@192.168.2.92:5090;transport=udp;registering_acc=192_168_2_92 on behalf of sip:mattia@192.168.2.92 for event presence Jul 5 10:36:36 roberto /usr/sbin/kamailio[8520]: ERROR: presence [subscribe.c:711]: Missing or unsupported event header field value Jul 5 10:36:36 roberto /usr/sbin/kamailio[8520]: ERROR: presence [subscribe.c:714]: #011event= message-summary Jul 5 10:36:36 roberto /usr/sbin/kamailio[8522]: INFO: presence [notify.c:1588]: NOTIFY sip:roberto@192.168.2.92 via sip:roberto@192.168.2.92:5090;transport=udp;registering_acc=192_168_2_92 on behalf of sip:roberto@192.168.2.92 for event presence.winfo *Jul 5 10:36:36 roberto /usr/sbin/kamailio[8520]: ERROR: db_postgres [km_dbase.c:427]: invalid query, execution aborted Jul 5 10:36:36 roberto /usr/sbin/kamailio[8520]: ERROR: db_postgres [km_dbase.c:428]: driver error: PGRES_FATAL_ERROR, ERROR: null value in column "sender" violates not-null constraint#012 Jul 5 10:36:36 roberto /usr/sbin/kamailio[8520]: WARNING: db_postgres [km_dbase.c:473]: unexpected result returned* Any idea? Best regards, Roberto Fichera.

2 2

Presence and Presence_mwi
by Marvin59 05 Jul '11

05 Jul '11

Hi all, I am trying to configure kamailio-3.1.4 in order to send me MWI Notify when there is change in messages. I have already configure it, in order to use a FreeRadius to authenticate my users but I have no idea to use Presence and Presence_mwi module. I need this behavior : - The phone send me a subscribe, then if it authorize i send it the Notify with the messages-waiting. - I must write myself the notify request because an external script will tell me if there are messages waiting. I have looked here (http://www.kamailio.org/docs/modules/3.1.x/modules_k/presence.html#id2614273 and http://www.kamailio.org/docs/modules/3.1.x/modules_k/presence_mwi.html). If someone has further informations .... Thanks for your help. Best regards, Marvin59 -- View this message in context: http://old.nabble.com/Presence-and-Presence_mwi-tp31946526p31946526.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

Session authorization just before initiate it
by Roberto Fichera 05 Jul '11

05 Jul '11

Hi All in the list, I'm new to Kamailio but I'm actually able to quickly setup a server, including also the SIP presence, looking at the example I found at http://kb.asipto.com/kamailio:presence:k31-made-simple which explain very well how to do it. I also made some changes in the final setup and everything seems working well, I can have audio/video/chat session including the presence status of any user. Now I would like to improve my testing setup so that for each call request Kamailio receive, I would like to perform some SQL lookup for deciding if the conversation might be established or not. About the SQL lookup I read about the SQLOPS module, it seems really easy to use, but I don't know where to put such check before the INVITE. Could someone give me some hints? P.S: Indeed, I need to read some good documentation about how to code a Kamailio config file. Any URL to suggest? Thanks in advance, Roberto Fichera.

2 4

route_graph.py
by s.maertens＠telenet.be 04 Jul '11

04 Jul '11

Hi, I have a kamailio 3.1.4 installation installed from the APT repository kamailio.org. I have read about a python script route_graph which I would like to test. The only thing is, i cannot seem to find out what deb package I should install to have this. Could someone point me in the right direction please? :) Best regards Stefaan

3 2

force_rtp_proxy[flag, ipaddress] not rewriting c and o ?
by MingHon 04 Jul '11

04 Jul '11

im trying to replace the ip address in c= and o= with force_rtp_proxy. i tried different flag co,rc,ro,fo,fc,rfco but still doesnt work. hope some one can help me with this.. below is my noob simple script. hope someone can help.. thanks you :) route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ xlog("L_INFO","force_rtp_proxy!!!! $rm from $fu (IP:$si:$sp)\n"); force_rtp_proxy("rfco","publicip"); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } -- Regards, MingHon

2 4

MWI Notify Request
by Marvin59 01 Jul '11

01 Jul '11

Hi all, I would like to know how kamailio sends Notify request by using the handle_subscribe of presence module. My purpose is to modify the request to add information for "message waiting indication" like : Messages-Waiting: yes, Voice-Message: 2/0 (0/0), ... I need your help ! Thanks in advance, Best regards Marvin -- View this message in context: http://old.nabble.com/MWI-Notify-Request-tp31954500p31954500.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

Internet RTP filters
by Dominguez Jover, Ricardo 30 Jun '11

30 Jun '11

Hi everybody, First to say sorry if this is not the right list to send this post, anyway we hope somebody can help us. We have Kamalio 3.1.3 working since many time ago. We use an RTP Proxy for NAT issues in the server side and STUN for NAT issues in the internet client side. Calls in Spain are working fine. This month one of our users has gone to Rwanda (our University has created a Hospital and a School in a region called Nemba). They are using an internet connection by satellite, with a low bandwith (128kb down, 16 up). In some countries we have trouble with internet calls. SIP signaling works fine, the call rings and is stablished, but there is a problem with RTP streams: - RTP streams sent by RTP proxy to client in internet never arrives to the client in the internet (although they RTP is sent to the right IP and port) - RTP streams sent by client in the internet never arrives to RTP proxy (although RTP is sent to the right IP and port) As I said this happens sometimes, and it's happening in Rwanda. We usually solve this problem making a VPN connection to the University, then the RTP stream works fine. However when our Rwanda's users connect to VPN, the Bandwith goes down (45 kbps down and 8kbps up), so they can hear as perfectly, but we can't hear them. Does anyone know if T companies are filtering RTP traffic? I've tried to change ports in RPT Proxy with no result. How could we solve this? Thanks in advance and best regards, Ricardo Dominguez Universidad Miguel Hernández de Elche

2 2

Re: [SR-Users] MWI handling
by Ovidiu Sas 30 Jun '11

30 Jun '11

If you are using asterisk as a voicemail server, let asterisk to handle the SUBSCRIBE and don't involve the kamailio presence server in this process. If you really want to use kamailio as a mwi presence server, then you will need to build some external scripts for pooling the asterisk mwi db and send PUBLISH out. I would stick with asterisk handling the mwi subscription and configure kamailio as a relay only for those events. Regards, Ovidiu Sas On Thu, Jun 30, 2011 at 9:30 AM, Spinov Evgeniy <spinov_evgeniy(a)intalisan.com> wrote: > This makes the main problem as in documents I've found on asterisk is > that it doesn't support PUBLISH, i.e. there is no way to force sending > them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme. > > Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio > about MWI? If yes, how? Cause right now, for me, it looks like a theory, > cause no mechanics to force asterisk send PUBLISHes. > > Thank you. > > Regards, > Evgeniy Spinov > > >> If you are using asterisk for voicemail, you should let asterisk >> notify the subscribers about mwi. >> If you want to use kamailio for notifications, then you need to >> configure asterisk to send PUBLISH requests to kamailio and kamailio >> will notify the subscribers. > > >> Regards, >> Ovidiu Sas > >

1 0

Re: [SR-Users] MWI handling
by Spinov Evgeniy 30 Jun '11

30 Jun '11

I've done that in this fashion: Kamailio is only proxing SUBSCRIBEs to Asterisk and it's replying with NOTIFYs to Kamailio, Kamailio is just proxying reply to UAC. So Kamailio is transparent for MWI. But it looks like spike. I'd better send PUBLISHes from Asterisk to Kamailio for all the clients and Kamailio reply with NOTIFY to subscribed ones. This is like Kamailio MWI module was designed. Anyway, thanks for the help. Scripts for making PUBLISHES is spike even more. On Thu, 2011-06-30 at 09:53 -0400, Ovidiu Sas wrote: > If you are using asterisk as a voicemail server, let asterisk to > handle the SUBSCRIBE and don't involve the kamailio presence server in > this process. > If you really want to use kamailio as a mwi presence server, then you > will need to build some external scripts for pooling the asterisk mwi > db and send PUBLISH out. > > I would stick with asterisk handling the mwi subscription and > configure kamailio as a relay only for those events. > > > Regards, > Ovidiu Sas > > On Thu, Jun 30, 2011 at 9:30 AM, Spinov Evgeniy > <spinov_evgeniy(a)intalisan.com> wrote: > > This makes the main problem as in documents I've found on asterisk is > > that it doesn't support PUBLISH, i.e. there is no way to force sending > > them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme. > > > > Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio > > about MWI? If yes, how? Cause right now, for me, it looks like a theory, > > cause no mechanics to force asterisk send PUBLISHes. > > > > Thank you. > > > > Regards, > > Evgeniy Spinov > > > > > >> If you are using asterisk for voicemail, you should let asterisk > >> notify the subscribers about mwi. > >> If you want to use kamailio for notifications, then you need to > >> configure asterisk to send PUBLISH requests to kamailio and kamailio > >> will notify the subscribers. > > > > > >> Regards, > >> Ovidiu Sas > > > >

1 0

Re: [SR-Users] MWI handling
by Spinov Evgeniy 30 Jun '11

30 Jun '11

This makes the main problem as in documents I've found on asterisk is that it doesn't support PUBLISH, i.e. there is no way to force sending them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme. Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio about MWI? If yes, how? Cause right now, for me, it looks like a theory, cause no mechanics to force asterisk send PUBLISHes. Thank you. Regards, Evgeniy Spinov > If you are using asterisk for voicemail, you should let asterisk > notify the subscribers about mwi. > If you want to use kamailio for notifications, then you need to > configure asterisk to send PUBLISH requests to kamailio and kamailio > will notify the subscribers. > Regards, > Ovidiu Sas

1 0

mq_fetch and while loops
by Watkins, Bradley 30 Jun '11

30 Jun '11

I'm having an issue trying to use the construct show in the module documentation. The docs show: while(mq_fetch("myq")) { xlog("$mqk(myq) - $mqv(myq)\n"); } And I'm doing essentially the same thing (differently named mqueue, but otherwise identical), but invariably I get this in the logs: ERROR: <core> [action.c:1388]: ERROR: runaway while (559, 28): more then 100 loops Am I doing something wrong, or is mq_fetch not returning appropriate values for successful versus failed fetches? This is on Kamailio 3.1.4, FWIW. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

4 5

Table location, methods field
by Andrew O. Zhukov 30 Jun '11

30 Jun '11

Can someone send me bitmasks for the Subj. Or just the reference to include file with comments in a source code. I can't find it in all set of availible document's since the SER router. :( I just have to know which UA except Grandstreams an CSipsample use RFC3311 :) Thanks in advance. Andrew O. Zhukov

2 5

kamailio + rtpproxy behind nat possible?
by MingHon 30 Jun '11

30 Jun '11

will kamailio as proxy behind nat and UACs behind another nat work? port forward sip and rtp done in the router. UACs register successfully but no audio. advertised_address = public_ip advertised_port = sip both define after the line of listen=public_ip please advice. thanks. -- Regards, MingHon

3 6

Proxy setup in a non-recursive mode
by Abdul Hakeem 29 Jun '11

29 Jun '11

Hello, Could anyone help with suggestions on how to setup a Proxy ( R-Proxy) that only does a route lookup and pass the parameter back to origination Proxy/Gateway A to forward the call based on the new parameter. On receipt of an invite from GW/Proxy A, R- Proxy will do a route lookup for the called number 447988870290, If it finds a route then it should append the network prefix to the new INVITE or rewrite the INVITE 23430442088870290 in the 302 Moved Temporarily message sent back to GW/Proxy A in the Contact field and there are two CC-Diversion header fields included; at GW/Proxy (IP address) who originated the call and already has a route or dial plan for the new prefix and also another one for same at GW/Proxy (IP address). Is there any way to do this without creating a loop in the network ? Previously this setup can be achieved with the originating network having 2 GW/Proxies i.e. GW/Proxy A and GW/Proxy B. In the 302 Temporarily moved message the Contact will show GW/Proxy B, while the 2 CC-Diversion header fields will add GW/Proxy B and GW Proxy A. Regards, Abdul Hakeem

2 3

MWI handling
by Spinov Evgeniy 29 Jun '11

29 Jun '11

Hello. Trying to understand how to work with the presence_mwi module. It seems to me, that it's just for my purpose - inform subscribed peers about waiting messages. But it do not export any functions, noting. The scheme is simple: Peer->Kamailio->Asterisk. Asterisk can accept voicemail and when the message is saved I want to receive NOTIFY for subscribed peer. As I understand, peer should register on Kamailio, kamailio should accept this subscription and send subscription to asterisk. After this, need to proxy all NOTIFY to the peer. Anyway, cannot find any way to do that. How it's usually done, please, advice. Thanks in advance.

2 1

Re: [SR-Users] Questions about LCR module
by Fabian Borot 29 Jun '11

29 Jun '11

Thanks Juha I populated the lcr tables by hand. This is how they look now, I have 2 lcr_ids [1 and 2], each instance has its own gw. And each instance has its own prefix to match for [4444 and 5555]. The idea is that a call comes in with R-URI 444413051234567, it should match the lcr_id= 200, then it should go out using gw 192.168.169.200 etc mysql> select * from lcr_gw; +----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | tag | flags | defunct | +----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+ | 1 | 200 | switchvox | 192.168.169.200 | pbx1.mydomain.com | 5060 | none | 1 | 1 | 0 | | 0 | 0 | | 2 | 201 | switchvox2 | 192.168.169.201 | pbx1.mydomain.net | 5060 | none | 1 | 1 | 0 | | 0 | 0 | +----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+ mysql> select * from lcr_rule; +----+--------+--------+----------+---------+---------+ | id | lcr_id | prefix | from_uri | stopper | enabled | +----+--------+--------+----------+---------+---------+ | 1 | 200 | 4444 | | 0 | 1 | | 2 | 201 | 5555 | | 0 | 1 | +----+--------+--------+----------+---------+---------+ mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 200 | 1 | 1 | 5 | 1 | | 2 | 201 | 2 | 2 | 5 | 1 | +----+--------+---------+-------+----------+--------+ Then I tried the sercmd commands to reload and dump and I do not see the contents o the screen. Is this how is supposed to work now?. with debug level set to 3 I do this "/usr/local/kamailio-3.1/sbin/sercmd lcr.dump_rules" and see these lines on the syslog: Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:390]: DBG: io_watch_add(0x2b772e494660, 12, 3, 0x9679e8), fd_no=1 Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:452]: handle_stream read: new connection (1) on /tmp/ser_ctl Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:505]: handle_stream read: eof on /tmp/ser_ctl Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:617]: DBG: io_watch_del (0x2b772e494660, 12, -1, 0x10) fd_no=2 called if I do this: "/usr/local/kamailio-3.1/sbin/sercmd lcr.reload" I see these: Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:390]: DBG: io_watch_add(0x2b772e494660, 12, 3, 0x9679e8), fd_no=1 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:452]: handle_stream read: new connection (1) on /tmp/ser_ctl Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db.c:294]: connection 0x951b78 found in pool Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 5 columns returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 40 bytes for result names at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 20 bytes for result types at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[0]=[id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952928)[1]=[prefix] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951a98)[2]=[from_uri] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951b08)[3]=[stopper] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[4] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952998)[4]=[enabled] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_dbase.c:326]: no rows returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: lcr [lcr_mod.c:985]: loading, cycle 0 with <0> rows Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 12 columns returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 96 bytes for result names at 0x951ef8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 48 bytes for result types at 0x951c38 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951b08)[0]=[gw_name] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951a98)[1]=[ip_addr] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952928)[2]=[port] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[3]=[uri_scheme] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[4] at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952610)[4]=[transport] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[5] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952998)[5]=[params] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[6] at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9527b8)[6]=[hostname] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[7] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951fb8)[7]=[strip] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[8] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952028)[8]=[tag] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[9] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952098)[9]=[flags] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[10] at 0x952108 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952108)[10]=[defunct] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[11] at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951288)[11]=[id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:169]: no rows returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 12 columns Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[5] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[6] at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[7] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[8] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[9] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[10] at 0x952108 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[11] at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x951ef8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x951c38 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 4 columns returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 32 bytes for result names at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 16 bytes for result types at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952098)[0]=[rule_id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952028)[1]=[gw_id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951fb8)[2]=[priority] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[3]=[weight] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_dbase.c:326]: no rows returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: lcr [lcr_mod.c:1344]: loading, cycle 0 with <0> rows Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 4 columns Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_pool.c:97]: connection still kept in the pool Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:505]: handle_stream read: eof on /tmp/ser_ctl Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:617]: DBG: io_watch_del (0x2b772e494660, 12, -1, 0x10) fd_no=2 called I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see > huge differences on the LCR module. there is a script somewhere (read wiki) that upgrades the tables. > I have these main questions: > > 1- does lcr_id replaces the grp_id? not really. lcr_ids denote independent lcr instances. > 2- what happened to the fifo commands to add gws etc: fifo commands to manipulate lcr tables are not supported by me. -- juha From: fborot(a)hotmail.com To: sr-users(a)lists.sip-router.org Subject: Questions about LCR module Date: Mon, 27 Jun 2011 17:23:05 -0400 I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see huge differences on the LCR module. I have these main questions: 1- does lcr_id replaces the grp_id? 2- what happened to the fifo commands to add gws etc: lcr addgw <gw_name> <ip> <port> <scheme> <transport> <grp_id> <flags> <tag> <strip> ............... add a gateway with flags, tag and strip ............ ................(flags, tag, and strip are optional arguments) ..... lcr rmgw <gw_name> .......................................................... ............... delete a gateway lcr addroute <prefix> <from> <grp_id> <prio> ................................. .............. add a route ( use '' to match anything in <from> ) lcr rmroute <prefix> <from> <grp_id> <prio> ................................. .............. delete a route 3- it looks like the kamctl fifo lcr reload can not be used anymore, instead I found that there is a sercmd command that can be used to "show" and "reload" only the lcr tables What would be the best way to use this module on this version? Any help/hint would be greatly appreciated. thank you fborot

2 3

Kamailio+ Asterisk 491 error code ????
by Sébastien Cramatte 28 Jun '11

28 Jun '11

Hello, I've just setup Kamailio+ Asterisk following Daniel Constantin Mierla howto http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb Everything seemts to works and I'm able to place a call to local extensions. I'm able to establish a call but strangely I get and 491 error code on my SIP Phones In my Asterisk console I get this error : [Jun 28 18:14:42] WARNING[24432]: chan_sip.c:18087 handle_response_invite: just did sched_add waitid(23) for sip_reinvite_retry for dialog LIFuQLu1l3ZtKrNy0XDX7lEk.Y5JdFko in handle_response_invite == Spawn extension (from-local, 102, 1) exited non-zero on 'SIP/103-00000000' Anybody can give me an hand to solve this issue ? Best regards

2 1

Dialog module question
by Javier Gallart 28 Jun '11

28 Jun '11

Hello I have little experience with the dialog module -so sorry if the quesiton is silly-, and I was wondering if I could use it for doing accounting (acc works just fine, but I was looking for a unique stop record that includes all the call information). The basic idea would be to assign the initial INVITE to a profile; and then, when BYE is received, build a cdr based of the dialog attributes...but afaik , upon reception of the BYE the dialog is destroyed. is there a way to access the dlg attributes inside the script when the BYE transaction is processed? Thanks in advance Javi

2 2

Questions about LCR module
by Fabian Borot 28 Jun '11

28 Jun '11

I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see huge differences on the LCR module. I have these main questions: 1- does lcr_id replaces the grp_id? 2- what happened to the fifo commands to add gws etc: lcr addgw <gw_name> <ip> <port> <scheme> <transport> <grp_id> <flags> <tag> <strip> ............... add a gateway with flags, tag and strip ............ ................(flags, tag, and strip are optional arguments) ..... lcr rmgw <gw_name> .......................................................... ............... delete a gateway lcr addroute <prefix> <from> <grp_id> <prio> ................................. .............. add a route ( use '' to match anything in <from> ) lcr rmroute <prefix> <from> <grp_id> <prio> ................................. .............. delete a route 3- it looks like the kamctl fifo lcr reload can not be used anymore, instead I found that there is a sercmd command that can be used to "show" and "reload" only the lcr tables What would be the best way to use this module on this version? Any help/hint would be greatly appreciated. thank you fborot

2 1

Re: [SR-Users] [sr-dev] new module: async - asynchronous SIP request processing in cfg
by Dan-Cristian Bogos 28 Jun '11

28 Jun '11

Hey Daniel, As said in the past, this module is good news, thanks for making it available. As a big fan of delayed billing, sleep() does a great job of delaying 200 OK from the upstream for me (useful for example with analogue/CDMA gateways sending 200 OK on ringback). Cheers, DanB

1 0

Kamailio and DTMF
by Phillman25 Kyriacou 28 Jun '11

28 Jun '11

Hi I was wondering what DTMF modes Kamailio supports? If i wanted to force a DTMF mode on Kamailio how could i go about doing it? thanking you in advance Phillip

3 3

rtpproxy and kamailio doesnt work out from the box.
by MingHon 28 Jun '11

28 Jun '11

hey list, i currently test on my asterisk 1.6 box on on pc with private ip and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc with public ip and private ip. everything installed successfully. i follow this tutorial realtime intergration with asterisk http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb my config. rtpproxy -l publicip -s udp:127.0.0.1:7722 -u user #!define WITH_NAT modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "sipping_from", "sip:pinger@publicip") nat_uac_test("19") # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); is my config correct? may i know the best version work out from the box? please advice.. thanks in adv. :) -- Regards, MingHon

4 9

new module: async - asynchronous SIP request processing in cfg
by Daniel-Constantin Mierla 27 Jun '11

27 Jun '11

Hello, a new module is now in GIT master branch, named async. Its purpose is to provide asynchronous SIP request processing using t_suspend()/t_continue() mechanism from tm module. While the tm module had it for quite some time, it was not available for usage in configuration file. There are two functions implemented by now: - async_route(routename, interval) - execute asynchronously a route block after a time interval - aync_sleep(interval) - sleep asynchronously for a time interval and then resume SIP request processing See the readme for more details: http://kamailio.org/docs/modules/devel/modules/async.html Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 4

LCR module
by Fabian Borot 27 Jun '11

27 Jun '11

After installing the latest stable version 3.1.4 I noticed that I can not run any 'lcr' commands from the 'kamctl fifo' interface. I noticed that the lcr.so is not on the lib64/modules_k or lib64/modules_s folders so I guess is was not included at installation time: [root@registrar kamailio]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_k/ [root@registrar modules_k]# ls acc.so db_text.so drouting.so mi_datagram.so permissions.so rtimer.so statistics.so usrloc.so alias_db.so db_unixodbc.so exec.so mi_fifo.so pike.so rtpproxy.so textops.so xlog.so auth_db.so dialog.so group.so msilo.so pua_mi.so siptrace.so tmx.so auth_diameter.so dispatcher.so htable.so nathelper.so pv.so siputils.so uac_redirect.so benchmark.so diversion.so imc.so nat_traversal.so qos.so speeddial.so uac.so call_control.so domainpolicy.so kex.so path.so registrar.so sqlops.so uri_db.so cfgutils.so domain.so maxfwd.so pdt.so rr.so sst.so userblacklist.so [root@registrar modules_k]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_s/ [root@registrar modules_s]# ls acc_db.so avp.so diversion.so fifo.so nathelper.so pike.so speeddial.so uri_db.so acc_syslog.so db_ops.so domain.so gflags.so options.so print.so textops.so uri.so auth_db.so dialog.so eval.so maxfwd.so pdt.so registrar.so timer.so usrloc.so avp_db.so dispatcher.so exec.so msilo.so permissions.so rr.so uac.so xlog.so [root@registrar modules_s]# I checked the modules.lst file and it is not on the exclude list, because I did removed it along with db_mysql and db_odbc, so I expected it to be included but it is not. Am I missing something? This is from the syslog Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: ERROR: mi_fifo [fifo_fnc.c:470]: fifo command lcr is not available Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: entered consume Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: **** done consume and this is from the fifo interface: [root@registrar kamailio]# /usr/local/kamailio-3.1/sbin/kamctl fifo lcr reload 500 command 'lcr' not available [root@registrar kamailio]#

1 1

async processing in a module
by Matthew Williams 27 Jun '11

27 Jun '11

Hi, I'm working on a jsonrpc (client) module that we would like to make async, using t_suspend and t_continue from the tm module. I have a couple questions related to this. The way I understand these functions, t_suspend will freeze the transaction in shared memory and immediately return and t_continue will unfreeze the transaction. What I'm not clear on here is whether t_continue will continue processing of the transaction within the process that calls t_continue, or will it delegate to a worker process? I assume I should spawn an io process in child_init which will be responsible for sending/receiving on the socket(s), but what is the best way to communicate between processes? Is there a way to get a fd from fork_process (that can be written to by any child and read from using select() in the io thread)? This is my first attempt at a real module, so any pointers are greatly appreciated. Regards, Matthew WIlliams

3 4

Kamailio start failed
by Roman Dmytriv 27 Jun '11

27 Jun '11

Hello friends! Having trouble to start newly installed Kamailio This is what I got: root@debian:/home/user# /etc/init.d/kamailio status Status of kamailio: kamailio is running. root@debian:/home/user# kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed root@debian:/home/user# Any ideas will appreciated Regards Roman

3 3

changing script (perl, lua ...) without kamailio restart
by Klaus Darilion 27 Jun '11

27 Jun '11

Hi! Does any of the script interpreters (perl, lua, python ....) allow to use a modified shell script without restarting Kamailio? Thanks Klaus

2 1

The 15000th development commit
by Daniel-Constantin Mierla 27 Jun '11

27 Jun '11

Hello, I just wanted to mark the 15000th development commit in our project (counted only on GIT master branch). It happened several days ago and it was done by the same person that did the first commit, Andrei: git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -1 512dcd9 Andrei Pelinescu-Onciul Mon Sep 3 21:27:11 2001 +0000 Initial revision The 15000th one is: git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -15000 | tail -1 8a90dd3 Andrei Pelinescu-Onciul Sat Jun 11 11:24:05 2011 +0200 core: remove unused variables + coding style I published the news about it on project sites as well: * http://asipto.com/u/37 * http://asipto.com/u/38 Now it's the time to look forward to the 20000th commit! Enjoy the summer, autumn comes with v3.2.0, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 12

an idea - the SIP browser - elsipo
by Daniel-Constantin Mierla 24 Jun '11

24 Jun '11

Hello, I want to present an idea and see if it is something people find benefits with such approach. Some time ago, due to existence of only SIP client applications (aka softphones) that have static GUI, which were designed for generic purposes, do not present what servces the currently used provider offers nor are easy to customize, I played with the idea of a SIP application with the GUI in HTML/CSS/Javascript. Being busy with Kamailio, the whole thing was postponed, but during last days I took the time and updated the code to latest developing tools (QT/Webkit and pjsip) and made it public on github, see: * http://asipto.com/u/3c A screenshot and some details in the wiki at: * http://asipto.com/u/3b The UI can be pulled from a remote site via http/https for example. The approach was to have a framework to render the UI, used QT/Webkit for that (cross platform, Linux, Mac OS X, Windows) and exported SIP API to Javascript. I worked with pjsip, but the system is plugin based, so pjsip plugin can be replaced, plugins for other things (not only sip/voip) can be written easily. The code is just for demo/prototyping, with the goal of showing feasibility. At this moment can be used to make/receive calls and send/receive instant messaging. In addition, content of UI can be carried by SIP messages - the examples added so far is to request some server statistics when clicking on a link of UI and the SIP server pushing periodically some info to a section of UI. Personally I don't have any plan to go into client side development, still the idea was too appealing to try and I thought would give lot of value and enable new features on SIP services to make user experience more pleasant. Therefore the target is to see if any of you think is something that worth pursuing and if yes, get people on board to develop it further from prototype phase. In the wiki I listed a part of my wish list with current VoIP/SIP services that would be trivial to implement with this approach. The naming as SIP browser instead of softphone hopefully suggest also the purpose of the application - to be able to browse the services offered by provider at that time and the provider to be able to push content back to application (on demand or periodically). Development was done on Mac OS X, a tarball with binaries for this OS is available for download on github - compiling for other OSes should go pretty straightforward. At the moment the idea poped up there was no such approach that I could find (at least not in open source space), did I miss something, are you aware of any similar approach? What is your opinion overall? Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 2

Dispatcher module for INVITE
by Giuseppe Carella 24 Jun '11

24 Jun '11

Hi all, I'm using kamailio dispatcher module between the OpenIMSCore and sailfin. Normally, without the dispatcher, all works fine. From the IMS Bench I can see that there are many successful call, but when I use the dispatcher module I'm having a lot of retransmission in the first part of the INVITE protocol. I'm using the default configuration file: # ----------------- setting module-specific parameters --------------- # -- dispatcher params -- modparam("dispatcher", "list_file", "../etc/dispatcher.list") # modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); forward(); # t_relay(); } I tried also the t_relay() function instead of the forward command, but the problem is the same. Thanks in advance. Bests, Giuseppe

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Don't publish Is not supposed the e-mail should be Hiden
by Omar 24 Jun '11

24 Jun '11

I really don't want to share my e-mail publicly, i remember there was an option before. O

3 2

Remote Party ID Header
by dotnetdub 23 Jun '11

23 Jun '11

Hi All, If an endpoint is sending us the remote party ID header, I know how to extract $(re{uri.user}) - how would I check if the Privacy Flag was true in Kamailio script? Version 1.41 Thanks, Brian

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Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint
by Brett Woollum 23 Jun '11

23 Jun '11

Hi Carsten, Thanks for the tip. All audio is going through RTPProxy on the Kamailio server, not directly to Asterisk. I will look into that patch. Thanks! Brett ----- Original Message ----- From: "Carsten Bock" <carsten(a)ng-voice.com> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Sent: Thursday, June 23, 2011 12:46:11 AM GMT -08:00 US/Canada Pacific Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint Hi, another solution might be, to either configure an RTP-Timeout on the Asterisk (if you send your calls through the asterisk anyway). You might also consider using the RTPProxy with the patch in the sip-router-repository. With the patch, the RTPProxy will trigger a teardown of calls (via XML-RPC) if the RTP-Session has a timeout. Carsten 2011/6/23 Brett Woollum <brett(a)woollum.com>: > Hi Alex, > > Thanks for this information. I've started researching the session-timer > capabilities in Asterisk, and I think that's my solution. I've already > implemented it on a test system and it works well, except that it's using > reINVITES to update as opposed to UPDATE messages, resulting in chops in the > audio every so often. I'll research this further though. > > Thanks again! > Brett > > ----- Original Message ----- > From: "Alex Balashov" <abalashov(a)evaristesys.com> > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users > Mailing List" <sr-users(a)lists.sip-router.org> > Cc: sr-users(a)lists.sip-router.org > Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific > Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to > SIP endpoint > > This is a complex topic. There is no way for a proxy like Kamailio to > detect this scenario per se. Kamailio reacts to and forwards signaling > events. If an endpoint disappears, it won't send any of those to indicate > that it has gone away. How would Kamailio know? Media stream timeout? > Kamailio doesn't relay media. > Your Kamailio-side solution is a dialog timeout, requiring use of > dialog-stateful tracking using the dialog module. But that will time out > calls indiscriminately, so you need to make it long enough to not anger your > users but short enough to be useful. > Your endpoint solution is SIP Session Timers. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > On Jun 23, 2011, at 1:10 AM, Brett Woollum <brett(a)woollum.com> wrote: > > Hello, > > We are running Kamailio as a registration point for our SIP phones, which > then interacts with Asterisk. SIP registrations are processed by Kamailio, > but everything else is passed to Asterisk. The Kamailio configuration is > close to the article at: > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. > Everything seems to be working well, until today. > > I found several calls today that were still connected to our provider, even > though our SIP phones were not active. There were three calls with timers at > 9 hours and counting. We had some IP connectivity issues earlier today, and > I'm wonder if it's related. > > If a SIP phone was connected and on a call (through kamailio), and the > kamailio/asterisk servers became unreachable, the SIP phones will drop the > call. But, it appears that kamailio/asterisk never drop the call in this > case, and the call stays live with the carrier. I had to manually kill the > calls by command prompt. > > What's the best way to handle this? Is there a way to have kamailio or > asterisk poll the phone to see if it's still on the call or something? How > can I give visibility to asterisk or kamailio so the calls are always > dropped properly? I don't want to run up a large bill because of calls that > didn't terminate when they should have. > > Thanks! > Brett > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ SIP Express Router (SER) and > Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint
by Brett Woollum 23 Jun '11

23 Jun '11

Hi Alex, Thanks for this information. I've started researching the session-timer capabilities in Asterisk, and I think that's my solution. I've already implemented it on a test system and it works well, except that it's using reINVITES to update as opposed to UPDATE messages, resulting in chops in the audio every so often. I'll research this further though. Thanks again! Brett ----- Original Message ----- From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: sr-users(a)lists.sip-router.org Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint This is a complex topic. There is no way for a proxy like Kamailio to detect this scenario per se. Kamailio reacts to and forwards signaling events. If an endpoint disappears, it won't send any of those to indicate that it has gone away. How would Kamailio know? Media stream timeout? Kamailio doesn't relay media. Your Kamailio-side solution is a dialog timeout, requiring use of dialog-stateful tracking using the dialog module. But that will time out calls indiscriminately, so you need to make it long enough to not anger your users but short enough to be useful. Your endpoint solution is SIP Session Timers. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jun 23, 2011, at 1:10 AM, Brett Woollum < brett(a)woollum.com > wrote: Hello, We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb . Everything seems to be working well, until today. I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related. If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt. What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have. Thanks! Brett _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Kamailio doesn't hang up upon IP connectity loss to SIP endpoint
by Brett Woollum 23 Jun '11

23 Jun '11

Hello, We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. Everything seems to be working well, until today. I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related. If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt. What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have. Thanks! Brett

2 1

apt-get install doesn't work
by Roman Dmytriv 23 Jun '11

23 Jun '11

Hello, I was trying to install kamailio using your instruction from http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I run Debian Squeeze. Imported apt repository keys, created this file </etc/apt/sources.list.d/kamailio.list> with this information in it <deb http://deb.kamailio.org/kamailio31-nightly squeeze main> Now when I run apt-get install kamailio kamailio-mysql-modules kamailio-tls-modules I got this response: root@s2:/etc/apt/sources.list.d# apt-get install kamailio kamailio-mysql-modules kamailio-tls-modules Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package kamailio E: Unable to locate package kamailio-mysql-modules E: Unable to locate package kamailio-tls-modules Can you give me a hint what could be wrong? Regards

2 5

OT: are there still asymmetric SIP clients?
by Klaus Darilion 22 Jun '11

22 Jun '11

Hi! Just a small survey: Are you still using NAT detection checks? I prefer to apply NAT traversal always (never trust the contact header) which of course only works if all the clients are symmetric. Thus - do you still see asymmetric SIP clients? Thanks Klaus

3 4

Kamailio/Asterisk/FreeSWitch dinner in Madrid - Thursday next week
by Olle E Johansson 22 Jun '11

22 Jun '11

Next week I'll be in the hot city of Madrid doing Asterisk/Kamailio training - The SIP master class. Maybe we can organize a voip nerd dinner on Thursday evening? If you're interested, please e-mail me off list and I'll send out more details later. Greetings /Olle PS. E-mail off list :-)

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Ukulele Kamailio Jingle
by Daniel-Constantin Mierla 22 Jun '11

22 Jun '11

Hello, a good friend, Javi Beltran, recorded a composition playing Ukulele (aka, ukelele - Hawaiian instrument - http://en.wikipedia.org/wiki/Ukulele). The jingle is available at: * http://asipto.com/u/39 If you like it, feel free to use it in relation with Kamailio project. Javi's compositions are open sourced, available at: * http://www.javibeltran.com/?lang=en I posted a news about it on project's site: * http://www.kamailio.org/w/2011/06/ukulele-kamailio-jingle/ If you or your friends have done any artistic work (music, graphics, a.s.o.) related to Kamailio, feel free to send it over and I will make them available on project's site. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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audio issue in same nat.
by †ஐû●чƃцίш† Goh 22 Jun '11

22 Jun '11

Hi List, below is my setup.. rtpproxy and kamailio in one PC with 2 nic. (ppp0 with public IP[60.49.119.XX] and eth1 with private IP[192.168.2.3])and asterisk is on another PC with private IP[192.168.2.23] i use realtime integration for kamailio and asterisk.http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6… i have two yealink hardphone ext 101(ip 192.168.1.200) and 102(ip 192.168.1.132) and a softphone ext 103 registered successful.both hardphone are behind same nat (175.136.221.XX)and softphone ext 103(ip 10.129.138.225) behind nat also (113.210.97.XX) ul show:database engine 'MYSQL' loadedControl engine 'FIFO' loadedentering fifo_cmd ul_dumpDomain:: location table=512 records=3 max_slot=1 AOR:: 102 Contact:: sip:102@175.136.221.XX:5062 Q= Expires:: 3109 Callid:: 2043273564(a)175.136.221.241 Cseq:: 4 User-agent:: T22 7.3.0.50 Received:: sip:175.136.221.241:1039 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:60.49.119.69:5060 Methods:: 16383 AOR:: 103 Contact:: sip:103@113.210.97.XX:58776;transport=UDP;ob Q= Expires:: 294 Callid:: oa8Pqx3mR.SVnzAVEYHTwVKZE8CbpY9l Cseq:: 27626 User-agent:: v1.0.0/iPhone State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:60.49.119.69:5060 Methods:: 8143 AOR:: 101 Contact:: sip:101@175.136.221.XX:5062 Q= Expires:: 1738 Callid:: 451417581(a)175.136.221.241 Cseq:: 2 User-agent:: T20 9.41.0.80 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:60.49.119.69:5060 Methods:: 16383FIFO command was::ul_dump:openser_receiver_17783 103 try to call 102 and 101 work fine. 101 and 102 try call 103 also fine.when 101 call 102 it work fine but when 102 call 101 there is no audio for both side.102 call 101 wireshark capture on 102 sidekeep send rtp but no receive.192.168.1.132 -> 60.49.119.XX RTP when capture on 101 side.keep send rtp but no receive.192.168.1.200 -> 60.49.119.XX RTP and also when 101 try to call into voicemail there is no audioit keep send rtp packet but to192.168.1.200 -> 192.168.2.23 RTP in kamailio.cfg#!WITH_NATlisten=60.49.119.XXlisten=192.168.2.3 # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); nat_uac_test("19")rtpproxy -l 60.49.119.XX -s udp:127.0.0.1 is running anyone can help me? how can i fix this?thanks in adv. Regards, minghon

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Re: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service when load the ldap.so module
by Daniel-Constantin Mierla 22 Jun '11

22 Jun '11

Thanks for reporting back, taking the chance to forward this email from sr-dev to sr-users since the thread was cross-posted to some point, for the benefit of future reads of the archive to show the solution of this case. Cheers, Daniel On 6/21/11 7:19 PM, laura testi wrote: > Found the problem after some investigation based on your suggestions. > In the openldap configuration file /etc/openldap/ldap.conf, the entry > TLS_CACERT is pointed to a certificate file in a directory which is > accessible only by root user. After I move the file to a directory > which all user can access it, now it works. I can start the kamailio > as a service with "service kamailio start" with the running user and > group is kamailio. > > Thank you all for your helps and suggestions! > > Best Regards, > Laura > > On Tue, Jun 21, 2011 at 5:44 PM, laura testi <lau.testi(a)gmail.com > <mailto:lau.testi@gmail.com>> wrote: > > The SELinux is disabled. > > > > > On Tue, Jun 21, 2011 at 5:26 PM, Andrew Pogrebennyk > <andrew.pogrebennyk(a)portaone.com > <mailto:andrew.pogrebennyk@portaone.com>> wrote: > > On 21.06.2011 17 <tel:21.06.2011%2017>:30, laura testi wrote: > > Should I need to set a some special environment variable > to load ldap.so > in the /etc/init.d/kamailio? > > > The env output for root does not confirm my theory :) I'm not > sure what is preventing kamailio from communication with LDAP, > could it be that SELinux is enabled and denying it in some way? > > -- > Sincerely, > Andrew Pogrebennyk > > > > > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

kamailio 3.1.3 fail to start as a service when load the ldap.so module
by laura testi 21 Jun '11

21 Jun '11

Hi, we try to use LDAP module in kamailio, first we try only load the ldap.so only to see if it can be loaded. When we start from the command line with either "kamctl start" or directly "kamilio", the kamailio is started normally with different children. When I try to start as service with "service kamailio start" or "/etc/init.d/kamailio start". It fails with the following error in the /var/log/messages: ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- Jun 21 10:28:39 kamailio3p1 kamailio: INFO: <core> [tcp_main.c:4716]: init_tcp: using epoll_lt as the io watch method (auto detected) Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: usrloc [hslot.c:53]: locks array size 512 Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: auth [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support disabled Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [parse_config.c:251]: file not found: //etc/kamailio/permissions.allow Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [permissions.c:606]: default allow file (//etc/kamailio/permissions.allow) not found => empty rule set Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [parse_config.c:251]: file not found: //etc/kamailio/permissions.deny Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [permissions.c:615]: default deny file (//etc/kamailio/permissions.deny) not found => empty rule set Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: ldap [ldap_mod.c:246]: OpenLDAP - 20343 Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_init.c:511]: tls: _init_tls_h: compiled with openssl version "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on, compression: on Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_init.c:519]: tls: init_tls_h: installed openssl library version "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on, zlib compression: on compiler: gcc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -I/usr/kerberos/include -DL_ENDIAN -DTERMIO -Wall -DMD32_REG_T=int -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wa,--noexecstack -DOPENSSL_USE_NEW_FUNCTIONS -fno-strict-aliasing -DOPENSSL_BN_ASM_MONT -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_init.c:373]: tls: init_tls: disabling compression... Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls [tls_init.c:564]: tls: init_tls_h: openssl kerberos malloc bug detected, kerberos support will be disabled... Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:176]: TLSs<default>: tls_method=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:188]: TLSs<default>: certificate='//etc/kamailio/FBCIDS01V01.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:195]: TLSs<default>: ca_list='//etc/kamailio/tiglobalca.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:202]: TLSs<default>: crl='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:206]: TLSs<default>: require_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:213]: TLSs<default>: cipher_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:220]: TLSs<default>: private_key='//etc/kamailio/FBCIDS01V01KEY.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:224]: TLSs<default>: verify_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:544]: TLSs<default>: No client certificate required and no checks performed Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f) has the SSL_OP_TLS_BLOCK_PADDING bug, but compression is disabled so no workaround is needed Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:176]: TLSc<default>: tls_method=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:188]: TLSc<default>: certificate='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:195]: TLSc<default>: ca_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:202]: TLSc<default>: crl='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:206]: TLSc<default>: require_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:213]: TLSc<default>: cipher_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:220]: TLSc<default>: private_key='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:224]: TLSc<default>: verify_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:227]: TLSc<default>: verify_depth=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:547]: TLSc<default>: Server MAY present invalid certificate Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f) has the SSL_OP_TLS_BLOCK_PADDING bug, but compression is disabled so no workaround is needed Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:176]: TLSs<10.10.5.214:5061>: tls_method=12 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:188]: TLSs<10.10.5.214:5061>: certificate='//etc/kamailio/FBCIDS01V01.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:195]: TLSs<10.10.5.214:5061>: ca_list='//etc/kamailio/tiglobalca.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:202]: TLSs<10.10.5.214:5061>: crl='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:206]: TLSs<10.10.5.214:5061>: require_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:213]: TLSs<10.10.5.214:5061>: cipher_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:220]: TLSs<10.10.5.214:5061>: private_key='//etc/kamailio/FBCIDS01V01KEY.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:224]: TLSs<10.10.5.214:5061>: verify_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:227]: TLSs<10.10.5.214:5061>: verify_depth=9 Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:544]: TLSs<10.10.5.214:5061>: No client certificate required and no checks performed Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f) has the SSL_OP_TLS_BLOCK_PADDING bug, but compression is disabled so no workaround is needed Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP server Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module ldap (/usr/lib64/kamailio/modules_k/ldap.so) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core> [pt.c:337]: ERROR: fork_process(): init_child failed for process 1, pid 8786, "udp receiver child=0 sock=127.0.0.1:5060" Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: : <core> [main.c:1546]: main_loop: Cannot fork Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP server Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module ldap (/usr/lib64/kamailio/modules_k/ldap.so) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core> [pt.c:337]: ERROR: fork_process(): init_child failed for process 2, pid 8788, "udp receiver child=1 sock=127.0.0.1:5060" Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: : <core> [main.c:1546]: main_loop: Cannot fork Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core> [main.c:738]: child process 8786 exited normally, status=255 Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core> [main.c:738]: child process 8788 exited normally, status=255 Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8801]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8799]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8797]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8795]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8793]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8790]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8812]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8808]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8810]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8814]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8816]: INFO: <core> [main.c:807]: INFO: signal 15 received ... -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- Without ldap.so loaded, we can start the kamailio as a service normally with "service kamailio start". The platform running kamailio is: RHEL5.6_X64 Can you help us please? Because we want to start the kamailio as a service from the boot Thanks a lot in advanced! Best Regards, Laura

4 10

an IDEA
by afshin afzali 21 Jun '11

21 Jun '11

Hi Guys, I need to integrate a SIP processing environment to H.248 and SS7/ ISUP links. For those communication links, OpenSS7 and Erlang/OTP MEGACO are good solutions. My idea is to write applications over OTP and SIP-Router to be able to bridge non-SIP protocols to SIP! On OTP the way is clear, but on SIP-Router I need to be able to generate / terminate SIP transactions. I'll appreciate every comment. BEST, -- afshin

3 3

Security in proxy
by Anto 20 Jun '11

20 Jun '11

Hi After reading the modules pike, pipelimit, etc. I wanted to know what measures can be used in the proxy, because like me, there will be more people interested ;-). We see that the module pike is a good security measure, but for users with many channels, we used the configuration of users with few channels (could we discriminate and make different rules for each?). If we have the proxy settings for normal users and have trunk configuration as well, I guess we also mark the traffic of the trunk to see it too aggressive about the user. As you might know if discriminate different types of users according to their traffic, one solution I can think of is to have different proxies for each user type (trunk, normal, etc). Is there any way to implement something that detects unusual behavior on the user (referred to countries that do not ever called, excessive traffic, etc)?. I ask not to try to implement something that exists or perhaps someone can advise me how to do so because it is very likely not get it or do it wrong :-P . Continue to seek information and if I find something to contribute, send him to the list. Can you think of some way more secure proxy (mainly user accounts)? Do some basic safety tips to take into consideration? Any guidance? My advice to avoid trying to schedule something with 100 lines of code when I can do with 5 lines? Does this indicate that I intend to do not I make it :-P ?. What I said, I will try paper on the subject and if I can find information on something (though it may seem a bad code :-( ), which publishes. Thanks ;-) Regards Anto

3 4

Keeping transactions in standby till future events
by Gustavo Garcia Bernardo 20 Jun '11

20 Jun '11

Is it possible to configure sip-router to keep a transaction for an incoming message (f.e. an INVITE) without sending a response and without forwarding it, and then reference and complete it in the future? Using pseudocode this is the logic I would like to implement: //When receiving an INVITE leave it pending if (request is INVITE) { store transactionID // This transactionID will be send somehow to the client return } //When receiving a PUBLISH referencing a pending transaction complete the forwarding if (request is PUBLISH) { transactionId = header("X-TransactionId") destination = header("X-Destination") transactions [transactionID]. forward(destination) } Thank you very much, G. Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx

4 7

Help find out where to modify request uri
by Simone Felici 20 Jun '11

20 Jun '11

Hi to all, I need to edit our ser installation (OLD ser-2.0.0-rc1) implementing a check to rewrite the INVITE URI address in case a flag on a DB is set or not. I'm not proof on this and as always the solution is to find out on the production system without the possibility (for now) to upgrade to most recent version. In fact I would load the flag with the db_ops module. The scenario should be the follow: 1. a call is coming from sip or from pstn to an internal sip account 2. the ser should check if a given flag for the called account is set (db_ops) 3. if it's set it should take the destination URI (taken from location db table) and rewrite the number to another known number. i.e. from sip:123456789@IP;uniqu=... to sip:new987654321@IP;uniq=... My first issue is to identify WHERE ser execute the INVITE to the sip account and how rewrite the URI. Could someone point me to the right place and give me some suggestion? The ser.cfg used could be read from a previous post of me: http://www.mail-archive.com/sr-users@lists.sip-router.org/msg04900.html Thank's a lot! Simon

2 4

Invite replies
by Lucas Alvarez 19 Jun '11

19 Jun '11

Hi, how may I catch the reply of an invite? I did this but it seems is not working: if (method=="INVITE") { $avp(sdp_ret) = fix_nated_sdp("11"); xlog("L_ALERT","#####********###### SDP return: $avp(sdp_ret)\n"); append_hf("P-hint: sdp rewritten\r\n"); t_on_reply("REPLY_TWO"); } onreply_route("REPLY_TWO") { xlog("L_ALERT","ALERT: ################ $rU - $rm from $fu (IP:$si:$sp--------$di)\n"); return; } Any help will be appreciate it, thanks in advance. Lucas

2 1

100rel without direct reachability
by Alex Balashov 19 Jun '11

19 Jun '11

(Sorry for cross-posting from Sip-Implementors; however, upon some reflection this is really more of a Kamailio mechanics question.) Greetings, I have a scenario where endpoint A is calling endpoint B through proxy P. Endpoints A and B are on different networks that are not directly reachable--A is on the public Internet, to which proxy P is connected, but NAT'd, and B is on a private RFC1918 network on which the proxy P has an interface as well. The purpose of proxy P is to make calling A <-> B possible, while providing the appropriate far-end NAT traversal functionality for A and so on and so on. It does this by inserting the correct Record-Route headers into the INVITE from A, and otherwise by staying in the signaling path; both A and B can reach the proxy, so all requests and replies go through it. The problem I'm having is prior to full dialog establishment. Upon receipt of INVITE from A by way of proxy P, B replies with 183 + SDP and also: Require: 100rel In the 183 it provides a Contact on its private network that A obviously cannot reach: Contact: <sip:xxxxx@192.168.0.10:5060> As a result, A attempts to send PRACKs to sip:xxxxx@192.168.0.10, which are obviously not going to go anywhere. After not receiving a few of these expected PRACKs, B replies to the call with a final negative failure. The question is, is there anything I can twiddle on the proxy to make this work? The problem with PRACKs is that although they are sequential requests, they arise (in this case) prior to the point at which RR can be used to guide them through the proxy. The most obvious answer is, "Make B not Require: 100rel", of course. However, that is not an option here. Thanks! -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 6

HTTP POST
by Matthew Williams 16 Jun '11

16 Jun '11

Hi, I'm working on an application that needs to create an HTTP POST from within Kamailio. The utils module has the http_query function, but that is only for GET. I think I'm looking for something more like the uac module's uac_req_send -- but for http. Any suggestions? Thanks in advance!

2 1

mi_datagram usage example
by Asgaroth 16 Jun '11

16 Jun '11

Hi All, I am trying to send a mi command over the mi_datagram udp socket. I dont seem to be getting a response back from the server. I am certain that it is something I am doing wrong. I have mi_datagram loaded and listening on udp:1.1.1.1:8080. I then try to send the following dispatcher mi command over to the mi_dg interface: echo -ne ":ds_list:\n\n" | nc -w 1 -u 1.1.1.1 8080 I dont see any error in the log file, so it appears that the mi_datagram is processing the request, but I am not getting any response back. The ds_list command works properly if I run it using "kamctl fifo ds_list", or if I go via "sercmd mi_dg ds_list" that works too. Would anyone have an example bash/ksh/perl snippet that would help me on my way to communicating with this mi interface. Any thoughts comments would be greatly appreciated. Thanks

3 3

Kamailio dispatcher
by Evgeniy Spinov 16 Jun '11

16 Jun '11

Hello. I had a Kamailio version of 3.0.3 and during this time configured a failover with simple routine. In short like this: if (ds_next_domain()) { xlog(.....); if (!t_relay()) { xlog(.....); return; } return; } else { t_reply("503"); } Then I've updated to 3.1.3 and was happy enough until I've discovered that my failover is not working anymore. What is observed: - It changes $rd variable, but not changing $du variable, is it ok? - Not any of the nodes receive packet on t_relay after ds_next_domain(). - In route decision section, where t_on_reply() and t_on_failure() are located I have the same t_relay() and it works fine, cause there is no any ds_next_domains(). Is it a bug or I'm doing something wrong? Thanks in advance.

3 11

How to check a bit (1/0) of a pseudo-variable?
by Iñaki Baz Castillo 16 Jun '11

16 Jun '11

Hi, summarizing I have two pseudo-variables: $var(allowed_zones) = 5; $var(id_zone) = 2; The binary representation of 5 (3+2), assuming 8 bits lenght, is: 00000101 so bit 0 = 1, bit 1 = 0, bit 2 = 1 ... Now I just need to know if bit number 2 ($var(id_zone)) of $var(allowed_zones) is 0 or 1. Is there any function like isflagset() but for pseudo-variables? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 6

Re: [SR-Users] Use of mtree module having multiple trees in the same table
by Javier Gallart 16 Jun '11

16 Jun '11

Hello not sure what you want to accomplish, but in case it might help this is how we use the mtree module. Our 3 columns are: partition, prefix, route Partition is just a name that defines a group, this is done inside the script before calling mt_match (via a table lookup for instance). Let's say if a call comes from ip x we assign it to partition 400; if it comes from ip y we assign it to partition 401. Each partition is a different tree. Consider these entries in the tree table: 400,123456,A 401,123456,A 401,1234567,B Let's say both x and y send a call with $rU 1234567; mt_match will return A for ip x, and B for ip y; we use that information for taking further routing decisions. A common situation might be: x has a unique route for a whole country while y has that entry but has more defined routes for more specific area codes inside that country. The column route could also be for instance the setid of a group of gw's for the dispatcher module, etc... Regards Javi > > Message: 3 > > Date: Tue, 14 Jun 2011 14:27:53 +0200 > > From: I?aki Baz Castillo <ibc(a)aliax.net> > > Subject: Re: [SR-Users] Use of mtree module having multiple trees in > > the same table > > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > > Users Mailing List" <sr-users(a)lists.sip-router.org> > > Message-ID: <BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww(a)mail.gmail.com> > > Content-Type: text/plain; charset=UTF-8 > > > > 2011/6/13 jose luis millan <doctore96(a)gmail.com>: > > > I wonder if anyone could help me understanding how can I use this > > > module in a scenario where multiple mtrees are defined behind the same > > > database table. > > > > > > I can't guess how to speficy to 'mt_match' the mtree I want to match > > > against, since the 'mtree' module parameter defines a specific memory > > > tree based on a specific table, but not based on a specific 'tname' > > > value. > > > > Indeed we have a table with these colums: > > > > - prefix: string to identify number destination. This is the regex > > column. > > - id_zone: integer, points to other table. This is the value column. > > - scope: integer, can be 0 or 1. > > > > So wee need two instances of mtree: > > 1) those rows with scope 0. > > 2) those rows with scope 1. > > > > Is it possible with current mtree module? or maybe we'd do better by > > creating two views of the table (one for entries with scope 0 and > > another for entries with scope 1)? > > > > Thanks a lot. > > > > -- > > I?aki Baz Castillo > > <ibc(a)aliax.net> > > > > > > > > >

2 1

Re: [SR-Users] sr-users Digest, Vol 73, Issue 40
by Javier Gallart 16 Jun '11

16 Jun '11

Hello not sure what you want to accomplish, but in case it might help this is how we use the mtree module. Our 3 columns are: partition, prefix, route Partition is just a name that defines a group, this is done inside the script before calling mt_match (via a table lookup for instance). Let's say if a call comes from ip x we assign it to partition 400; if it comes from ip y we assign it to partition 401. Each partition is a different tree. Consider these entries in the tree table: 400,123456,A 401,123456,A 401,1234567,B Let's say both x and y send a call with $rU 1234567; mt_match will return A for ip x, and B for ip y; we use that information for taking further routing decisions. A common situation might be: x has a unique route for a whole country while y has that entry but has more defined routes for more specific area codes inside that country. The column route could also be for instance the setid of a group of gw's for the dispatcher module, etc... Regards Javi Message: 3 > Date: Tue, 14 Jun 2011 14:27:53 +0200 > From: I?aki Baz Castillo <ibc(a)aliax.net> > Subject: Re: [SR-Users] Use of mtree module having multiple trees in > the same table > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww(a)mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > 2011/6/13 jose luis millan <doctore96(a)gmail.com>: > > I wonder if anyone could help me understanding how can I use this > > module in a scenario where multiple mtrees are defined behind the same > > database table. > > > > I can't guess how to speficy to 'mt_match' the mtree I want to match > > against, since the 'mtree' module parameter defines a specific memory > > tree based on a specific table, but not based on a specific 'tname' > > value. > > Indeed we have a table with these colums: > > - prefix: string to identify number destination. This is the regex > column. > - id_zone: integer, points to other table. This is the value column. > - scope: integer, can be 0 or 1. > > So wee need two instances of mtree: > 1) those rows with scope 0. > 2) those rows with scope 1. > > Is it possible with current mtree module? or maybe we'd do better by > creating two views of the table (one for entries with scope 0 and > another for entries with scope 1)? > > Thanks a lot. > > -- > I?aki Baz Castillo > <ibc(a)aliax.net> > > > >

1 0

t_relay_to("0x02") alternative with kamailio 3.1
by Herve Couplet 16 Jun '11

16 Jun '11

Hi, I had a setup with an old version where I used the function t_relay_to("0x02"). Following the 3.1 documentation, flag : * 0x02* - do not generate reply on internal error (NOTE: has no effect anymore) I used that function as a workaround with a pbx that stop to register when it receive an error response like 478 Unresolvable destination or 408 request timeout. I did upgrade to 3.1 to have dns srv caching. Do you know how this would be possible with kamailio 3.1. Thanks for your suggestions, hp

4 4

Operations With Variables $var- does it need to load a module
by Omar 15 Jun '11

15 Jun '11

In Kamailio 1.4.3 I have the following scenario: $var(a) = 0 $var(b) = 0 $var(a) = $(tU{s.len}); $var(b) = $var(a) - 5; xlog(" $var(b)"); the result $var(b) is always the same as $var(a) should i check if some specific module is loaded? the debugs does not show any error, and $var(a) and $var(b) are unique. Variables are working, but not the operations with the variables. regards O

3 4

200 OK Retransmission Error
by Jijo 15 Jun '11

15 Jun '11

Hi All, In an INVITE transaction, no ACK from A party and B party continue retransmission of 200 OK. I have observed that kamailio is not passing the retransmitted 200 OK message from B party to the Application(script) after 3 seconds. Instead kamailio directly sending to A party without passing to the script. I see in the logs that kamailio is cleaning up the transaction after 3 seconds..look for the string "DEBUG: t_lookup_request: no transaction found" before this log it is cleaning up the timer in the log "DEBUG: cleanup_uac_timers: RETR/FR timers reset" What does it mean by that?? How can we make the kamailio to pass the response to the application? Please let me know your comments.. Here is the flow. I cannot send the complete logs which is quite big. so i grep'd using the stirng "tcp_read_req: content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" from the complete log.. A kamailo B ----INVITE-->- --------------------INVITE 180 Ringing--<---------------180 Ringing 200 OK-----------<---------------200 OK 0 200 OK-----------<---------------200 OK 500ms 200 OK-----------<---------------200 OK 1s 200 OK-----------<---------------200 OK 2s < From here it is not passing to the application- script> 200 OK-----------<---------------200 OK 4s 200 OK-----------<---------------200 OK 4s 28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; uri=[ sip:5616990102@192.168.243.151:5060] 36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 0 282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:180 Reason=Ringing CT=5616990102 <sip: 5616990102@192.168.243.161:5060;transport=tcp> 331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction

2 2

NAPTR priorities doesn't seem to work properly
by Iñaki Baz Castillo 15 Jun '11

15 Jun '11

Hi, I'm testing Kamailio's NAPTR resolution: version: kamailio 3.2.0-dev4 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB DNS related configuration: dns_servers_no=2 dns_srv_lb=yes dns_try_naptr=yes dns_use_search_list=no use_dns_cache=yes use_dns_failover=yes dns_tls_preference=1 dns_tcp_preference=1 dns_udp_preference=1 As the doc says it should respect the priorities of the NAPTR records, see http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#dns_sctp_pref_dns…: "To use the remote site preferences set all dns_*_pref to the same positive value (e.g. dns_udp_pref=1, dns_tcp_pref=1, dns_tls_pref=1, dns_sctp_pref=1)" So my kamailio receives an INVITE for an external domain oversip.net (no transport param neither port in the RURI) and does a t_relay. Request RURI is sip:qwe@oversip.net. According to NAPTR: ~$ host -t naptr oversip.net oversip.net has NAPTR record 5 50 "S" "SIPS+D2T" "" _sips._tcp.oversip.net. oversip.net has NAPTR record 10 50 "S" "SIP+D2T" "" _sip._tcp.oversip.net. oversip.net has NAPTR record 20 50 "S" "SIP+D2U" "" _sip._udp.oversip.net. oversip.net has NAPTR record 40 50 "S" "SIP+D2S" "" _sip._sctp.oversip.net. oversip.net has NAPTR record 50 50 "S" "SIPS+D2S" "" _sips._sctp.oversip.net. So it should try TLS over TCP first, if it fails try TCP and if it fails try UDP. However it just uses UDP, why?? Even if I set a minor value to dns_tls_preference (so higher priority I expect) it still uses UDP. Am I doing something wrong? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

6 29

kamailio conference calls support
by Mino Haluz 15 Jun '11

15 Jun '11

Hi, does kamailio support conference calls or it has to be implemented in some B2BUA ? If it is supported, what are the requirements (modules etc.). Thanks, Mino

2 1

302 rewrite ruri based on contact
by Omar 15 Jun '11

15 Jun '11

I need to rewrite the destination after I receive 302 I have this logic Onreply Loads in avp the information needed I try sending to failure route and to a route in both cases the $ru become <null> In the route and failureroute I did try rewritehost, rewriteuri, $ru= number+ host And append the route But $ru is always null. Inside the route or failure route If there are no 302 nor the onreply involved everything works fine Any suggestion to rewrite the ruri after a 302 is welcome I am still using Kamailo1.4.3 Omar

3 6

Modifying sip packet
by Lucas Alvarez 14 Jun '11

14 Jun '11

Hi, I want to modify the following sip packet, I want to change the SDP IP address by $di, which rule could filter this packet? Something like: [ if method("OK") and $di is RFC1918 ] ??? And with which function could I change the Ip address in the SDP content? In this case I would like to change 172.16.99.2 by 190.244.125.41. Thanks in advance, Lucas T 172.16.99.2:5060 -> 190.244.125.41:9055 [AP] SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.99.2;branch=z9hG4bK7c5.bfc43456.0;i=1;received=172.16.99.2;rport=5060. Via: SIP/2.0/TCP 192.168.15.231:1952 ;received=190.244.125.41;branch=z9hG4bK-d8754z-be53247ec649be5b-1---d8754z-;rport=9055. Record-Route: <sip:172.16.99.2;r2=on;lr=on;ftag=b11c6a4e>. Record-Route: <sip:172.16.99.2;transport=tcp;r2=on;lr=on;ftag=b11c6a4e>. From: "Lucas"<sip:1104@67.152.18.231>;tag=b11c6a4e. To: <sip:1101@67.152.18.231>;tag=as5356a251. Call-ID: NzcwZWIxNGE1MDg5ZjVhZWRhYWNlM2RiNmMwM2M4MWM.. CSeq: 2 INVITE. Server: Asterisk PBX 1.8.4.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: <sip:1101@172.16.99.2:5080>. Remote-Party-ID: "Gus Office One" <sip:1101@67.152.18.231 >;party=called;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 305. . v=0. o=root 1124623741 1124623741 IN IP4 172.16.99.2. s=Asterisk PBX 1.8.4.1. c=IN IP4 172.16.99.2. t=0 0. m=audio 13640 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.

2 1

Use of mtree module having multiple trees in the same table
by jose luis millan 14 Jun '11

14 Jun '11

hello, I wonder if anyone could help me understanding how can I use this module in a scenario where multiple mtrees are defined behind the same database table. I can't guess how to speficy to 'mt_match' the mtree I want to match against, since the 'mtree' module parameter defines a specific memory tree based on a specific table, but not based on a specific 'tname' value. Any help is apreciated, Thanks in advance.

2 2

Fwd: 200OK Retransmission Error
by Jijo 14 Jun '11

14 Jun '11

Hi All, In an INVITE transaction, no ACK from A party and B party continue retransmission of 200 OK. I have observed that kamailio is not passing the retransmitted 200 OK message from B party to the Application(script) after 3 seconds. Instead kamailio directly sending to A party without passing to the script. I see in the logs that kamailio is cleaning up the transaction after 3 seconds..look for the string "DEBUG: t_lookup_request: no transaction found" before this log it is cleaning up the timer in the log "DEBUG: cleanup_uac_timers: RETR/FR timers reset" What does it mean by that?? How can we make the kamailio to pass the response to the application? Please let me know your comments.. Here is the flow. I cannot send the complete logs which is quite big. so i grep'd using the stirng "tcp_read_req: content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" from the complete log.. A kamailo B ----INVITE-->- --------------------INVITE 180 Ringing--<---------------180 Ringing 200 OK-----------<---------------200 OK 0 200 OK-----------<---------------200 OK 500ms 200 OK-----------<---------------200 OK 1s 200 OK-----------<---------------200 OK 2s < From here it is not passing to the application- script> 200 OK-----------<---------------200 OK 4s 200 OK-----------<---------------200 OK 4s 28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; uri=[ sip:5616990102@192.168.243.151:5060] 36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 0 282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:180 Reason=Ringing CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction

1 1

RTP proxy
by michel freiha 14 Jun '11

14 Jun '11

Dear All, I'm using the below config plan for routing my calls...The issue that forcing rtp proxy is not working well and the rtp proxy is never forced....Can you please provide me a guidance here? if($rU=~"^00.*" ) { if(!cr_route("default", "domain.com", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { setbflag(6); route(1); } exit; ## this was missing here -bogdan!!! } else { route(LOCATION); route(RELAY); } } ############# THIS IS THE GENERIC RELAY ROUTE THAT MUST BE USED ALL THE TIME FOR RELAY route[1] { # apply any NAT changes? if(isflagset(5)||isbflagset(6)) { route(4); } exit; } route[4] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ if (nat_uac_test("8")) { * force_rtp_proxy();* } else { force_rtp_proxy(); } xlog("L_ERR","66666666666666666666666666666666666666666"); t_on_failure("1"); t_on_reply("1"); }; The call is going through Route[1] then through Route[4], but rtp proxy is never forced...Any tips please? Regards

2 1

Meaning of empty body in NOTIFY
by Eugen Dedu 13 Jun '11

13 Jun '11

Hi, ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for users who are not registered an empty NOTIFY body is returned when asked by a SUBSCRIBE. What does this mean from SIP standard point of view, and from kamailio point of view (are they identical?) I see in RFC3265/3.1.6.2: .... If the resource has no meaningful state at the time that the SUBSCRIBE message is processed, this NOTIFY message MAY contain an empty or neutral body but is difficult for me to interpret what it means. Example: I ask the presence for a user xyz who registered and quit application long time ago: SUBSCRIBE sip:xyz@ekiga.net SIP/2.0 CSeq: 1 SUBSCRIBE Via: SIP/2.0/UDP 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport User-Agent: Ekiga/3.3.1 From: <sip:eugen.dedu@ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8 Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8@snoopy Supported: eventlist To: <sip:xyz@ekiga.net> Accept: application/pidf+xml Accept: multipart/related Accept: application/rlmi+xml Contact: <sip:eugen.dedu@82.238.108.175:5060> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK Expires: 300 Event: presence Content-Length: 0 Max-Forwards: 70 I receive the following answer: NOTIFY sip:eugen.dedu@82.238.108.175:5060 SIP/2.0 CSeq: 1 NOTIFY Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0 User-Agent: Kamailio (1.5.3-notls (i386/linux)) From: sip:xyz@ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0 Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8@snoopy To: sip:eugen.dedu@ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8 Contact: <sip:86.64.162.35:5060> Subscription-State: active;expires=370 Event: presence Content-Length: 0 Max-Forwards: 70 To resume: What does SIP standard say about this NOTIFY with empty body? Does this mean that the user xyz is offline? Or does this mean that user's status has not changed? In fact, the NOTIFY with empty body (as shown above) is the first one sent by kamailio, so there is no "previous" state of that user, hence "unchanged" status has no meaning. Thank you, -- Eugen

6 21

Proxy Address
by michel freiha 13 Jun '11

13 Jun '11

Dear All, I'm using kamailio as Proxy server for registration with RTP proxy as relay server for voice packets...I have an issue that when sending an Invite, the audio address in SDP (o header)is not RTP Proxy address while if I send another Invite to kamailio (second Invite), the RTP Proxy address will be present in SDP sent back by Kamailio... Anyone has a clue about what could be that matter? Regards

2 2

Double tuple in notify
by Eugen Dedu 12 Jun '11

12 Jun '11

Hi again, > Then it is the time for Eugen to add it in his sip client and send > the patch for enhancing kamailio's presence server with rfc4481, as > well :o) I start my application, and make my own status be shown. I change my status back and forth a few times between Away and DoNotDisturb (each time a publish is sent), the status returned (in notify) is good. When I change my status to Online/Available (and only to this), kamailio returns a double tuple. Is it normal to have a double tuple? If yes, which one to choose (the one with latest timestamp)? Here it is: NOTIFY sip:eugen.dedu@82.238.108.175:5060 SIP/2.0 CSeq: 7 NOTIFY Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK7634.4d863c14.0 User-Agent: Kamailio (1.5.3-notls (i386/linux)) From: sip:eugen.dedu@ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-29fc Call-ID: 8a0d723b-0f8d-e011-84d7-0024d693d8e8@snoopy To: sip:eugen.dedu@ekiga.net;tag=7c75723b-0f8d-e011-84d7-0024d693d8e8 Contact: <sip:86.64.162.35:5060> Subscription-State: active;expires=270 Event: presence Content-Length: 938 Content-Type: application/pidf+xml Max-Forwards: 70 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns: rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:eugen.dedu@ekiga.net"> <tuple id="T00000002"> <status> <basic>open</basic> </status> <contact priority="1">sip:eugen.dedu@ekiga.net</contact> <note>online - </note> <timestamp>2011-06-04T13:55:56+02:00+02:00</timestamp> </tuple> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="T00000001"> <status> <basic>open</basic> </status> <contact priority="1">sip:eugen.dedu@ekiga.net</contact> <note>away - jjk</note> <timestamp>2011-06-04T13:54:37+02:00+02:00</timestamp> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="pid1"> <rpid:activities> <rpid:away/> </rpid:activities> </dm:person> </presence> -- Eugen

3 2

Re: [SR-Users] Meaning of empty body in NOTIFY
by Daniel-Constantin Mierla 10 Jun '11

10 Jun '11

On 6/10/11 10:07 PM, Craig Southeren wrote: > On 10/06/2011 1:01 PM, Daniel-Constantin Mierla wrote:..deleted >>> >>> So, if a client uploads a presence document that contains a >>> <timed-status> element, that element could be sent by the PA when >>> the client goes offline, rather than the entire document being >>> discarded (which is what I suspect is happening now). >> >> we don't have this rfc implemented in kamailio at this time. Is any >> UA you know doing this kind of thing? > > No Then it is the time for Eugen to add it in his sip client and send the patch for enhancing kamailio's presence server with rfc4481, as well ;-) Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

Functions for DNS
by Nathan Angelacos 10 Jun '11

10 Jun '11

I'm looking to build a set of contacts with q values based on SRV records for serial/parallel forking. I want "enum_query" (loading the contact set with a q value based on the order,preference of the NAPTR record) but the ruri is not an e.164 number, and SRV records are used instead. Think in terms of "find all peer proxies based on SRV lookup, and sort them for use with t_load_contacts()/t_next_contacts()." So far I haven't found any api that exposes the DNS lookups in sip-router. Before rolling something custom with Lua, am I missing something that's already documented? Or are there plans to make such a module? Sure don't want to re-invent the wheel. tia

2 2

New developer: Peter Dunkley
by Daniel-Constantin Mierla 09 Jun '11

09 Jun '11

Hello, just to let everyone to know that starting with today Peter got developer access to GIT repository. He sent many patches in the past months to fix issues or add new features to presence modules (e.g., OMA extensions to embedded XCAP server) -- some are pending to be committed and expect more to come, as he is continuing to work with this part of SIP server. The user id is 'pd'. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Best way to handle OPTIONS message with Max-Forwards of 0
by Sean O'Donnell 09 Jun '11

09 Jun '11

Looks more comprehensive than mine. Thanks! Sean O'Donnell ---- On Wed, 8 Jun 2011, Klaus Darilion (klaus.mailinglists(a)pernau.at) wrote: Anyway - not standard conform but efficient (put somewhere in the beginning) force_rport(); if (is_method("OPTIONS")) { sl_send_reply("200","OK"); exit; } # sipura/linksys keepalive if (is_method("NOTIFY")) { if ( search("^Event: keep-alive") ) { sl_send_reply("200","Keep Alive"); exit; } } # ignore requests generated by sipvicious # User-Agent: friendly-scanner if ($ua == "friendly-scanner") { exit; } regards Klaus Am 08.06.2011 18:28, schrieb Sean O'Donnell: > Hi: > > We're running kamailio 1.5.5 as part of our VoIP infrastructure. One of our > service providers > has an SBC that sends an OPTIONS messages with a Max-Forwards of 0. > > Looking at RFC 3261 section 11, this seems to be a valid method of forcing the > proxy to respond to the OPTIONS. > However, the standard initial sanity checks in most proxy configs will reject > this message. I fixed our script > by changing to this: > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > if( method != "OPTIONS" ) { > sl_send_reply("483","Too Many Hops"); > return; > }; > }; > if ( msg:len > max_len ) { > sl_send_reply("513", "Message too big"); > return; > }; > > but does anyone have a better solution? > > Thanks, > > Sean O'Donnell > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] Best way to handle OPTIONS message with Max-Forwards of 0
by Sean O'Donnell 08 Jun '11

08 Jun '11

Thanks for the reply. In our case, the SBC treated the 483 as an error of some sort and wouldn't send traffic. It wanted a 200/OK. Anyway, I thought that when a proxy responds to an OPTIONS, its supposed to answer with a 200/OK (3261 11.2). Is that correct? Sean O'Donnell ---- On Wed, 8 Jun 2011, Klaus Darilion (klaus.mailinglists(a)pernau.at) wrote: What is the problem with the default config? The proxy will respond with 483 as the standard defines. I guess the SBC will doe this to check if the proxy is alive and will accept any response as "the proxy is alive". regards Klaus Am 08.06.2011 18:28, schrieb Sean O'Donnell: > Hi: > > We're running kamailio 1.5.5 as part of our VoIP infrastructure. One of our > service providers > has an SBC that sends an OPTIONS messages with a Max-Forwards of 0. > > Looking at RFC 3261 section 11, this seems to be a valid method of forcing the > proxy to respond to the OPTIONS. > However, the standard initial sanity checks in most proxy configs will reject > this message. I fixed our script > by changing to this: > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > if( method != "OPTIONS" ) { > sl_send_reply("483","Too Many Hops"); > return; > }; > }; > if ( msg:len > max_len ) { > sl_send_reply("513", "Message too big"); > return; > }; > > but does anyone have a better solution? > > Thanks, > > Sean O'Donnell > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Best way to handle OPTIONS message with Max-Forwards of 0
by Sean O'Donnell 08 Jun '11

08 Jun '11

Hi: We're running kamailio 1.5.5 as part of our VoIP infrastructure. One of our service providers has an SBC that sends an OPTIONS messages with a Max-Forwards of 0. Looking at RFC 3261 section 11, this seems to be a valid method of forcing the proxy to respond to the OPTIONS. However, the standard initial sanity checks in most proxy configs will reject this message. I fixed our script by changing to this: # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { if( method != "OPTIONS" ) { sl_send_reply("483","Too Many Hops"); return; }; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); return; }; but does anyone have a better solution? Thanks, Sean O'Donnell

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Incorrect registrar Expires handling
by Andreas Granig 08 Jun '11

08 Jun '11

Hi, Found an interesting call-flow today where a client (pjsip) is behind NAT and behaves like this during registration (kamailio-3.1.3): REGISTER sip:somedomain.com Contact: <sip:me@10.0.0.7:2001;ob> Expires: 900 SIP/2.0 200 OK Contact: <sip:me@10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001" REGISTER sip:somedomain.com Contact: <sip:me@1.2.3.4:3001;transport=UDP;ob> Contact: <sip:me@10.0.0.7:2001;ob>;expires=0 Expires: 900 SIP/2.0 200 OK Contact: <sip:me@10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001", <sip:me@1.2.3.4:3001;transport=UDP;ob>;expires=900 So obviously the client is trying to unregister its private contact (for whatever reason - probably to play nice with registrars not supporting far-end NAT traversal), and tries to register the Contact with the "received" address instead. Kamailio however ignores the ";expires=0" for the private Contact and just saves both of them with the expiry value given in the Expires header. Taking the client behavior aside for now: is this the correct behavior of the registrar module? In RFC3261 chapter 10.2.1.1 it says: #+ There are two ways in which a client can suggest an expiration interval for a binding: through an Expires header field or an "expires" Contact header parameter. The latter allows expiration intervals to be suggested on a per-binding basis when more than one binding is given in a single REGISTER request, whereas the former suggests an expiration interval for all Contact header field values that do not contain the "expires" parameter. #- So I'd say that in my specific case "expires" Contact header param (which is 0 for the binding having 10.0.0.7) should de-register this particular binding, whereas the Expires header should be applied to the second binding (having the 1.2.3.4) because it doesn't carry an "expires" param. Opinions on that one? Andreas

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incorrect port 0 in reply from rtp proxy
by Mokhtar Bengana 08 Jun '11

08 Jun '11

I have Kamailio and rtpproxy running behind NAT on the same server. Inbound calls to Kamailio from the outside work fine and I have two way audio but when I try an outbound echo test call the call connects but I have no audio. rtpproxy seems to be running and all rtp ports are poiting to Kamailio internal ip. The syslog is showing this error. Not sure what's missing in the config. Appreciate any inputs. Thanks. root@Kamailio:~# tail /var/log/syslog Jun 3 01:24:02 Kamailio kamailio[810]: INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:22222> found, support for it enabled Jun 3 01:25:05 Kamailio kamailio[791]: INFO: <core> [forward.c:786]: broken reply to forward - no 2nd via Jun 3 01:25:29 Kamailio kamailio[791]: ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy Jun 3 01:25:30 Kamailio kamailio[793]: ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy

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t.38 fax
by Sanjeev Singh 07 Jun '11

07 Jun '11

does SER support t.38 fax? what other features related to fax does SER support. please expain in detail. thanks, Sanjeev Singh

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xhttp and dollar sign
by Andreas Granig 06 Jun '11

06 Jun '11

Hi, Is it possible to somehow quote the $ sign in xhttp_send_reply() in order not to be interpreted by kamailio-3.1.3? Tried with "\" and putting the whole body in single quotes, but I always get this error message during startup: ERROR: <core> [pvapi.c:720]: wrong char [$/36] in [${foo}] ERROR: <core> [sr_module.c:1156]: bad PVE format: "${foo}" ERROR: <core> [mod_fix.c:501]: Cannot convert function parameter 1 to spve So I actually just want to return the string "${foo}". Andreas

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Error bad syntax - DEBUG_FLAGS
by Skyler 06 Jun '11

06 Jun '11

Hi all, I'm new to Kamailio (3.1), learning the configuration language. I'm trying to take the advice from one of the docs to create a DEBUG block and place calls everywhere in my script. Right out of the gate I get these errors on loading. Can someone tell me where I went wrong and how I can fix this? It says flag not declared though it's there and I'm a bit lost. 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 6: syntax error 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 6: missing '(' or ')' ? 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 6: bad command: missing ';'? 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 18: '('')' expected (function call) 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 18: bad command: missing ';'? 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 20: bad command 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 802, column 24: flag not declared #------------------------------ # Flags #------------------------------ #!define FLAG_DEBUG 1 ... route { # Call debug flags route(ROUTE_DEBUG_FLAGS); ..... route[ROUTE_DEBUG_FLAGS] { resetflag(FLAG_DEBUG); #------------------------------------------------------------ # Methods that will have debugging turned on #----------------------------------------------------------- if (is_method("ACK")) { resetflag(FLAG_DEBUG); } else if (is_method("BYE")) { setflag(FLAG_DEBUG); } else if (is_method("CANCEL")) { setflag(FLAG_DEBUG); } else if (is_method("INVITE")) { setflag(FLAG_DEBUG); } else if (is_method("OPTIONS")) { setflag(FLAG_DEBUG); } else if (is_method("PUBLISH")) { setflag(FLAG_DEBUG); } else if (is_method("REGISTER")) { resetflag(FLAG_DEBUG); } else if (is_method("MESSAGE")) { setflag(FLAG_DEBUG); } else if (is_method("NOTIFY")) { setflag(FLAG_DEBUG); } else if (is_method("SUBSCRIBE")) { setflag(FLAG_DEBUG); } else if (is_method("INFO")) { setflag(FLAG_DEBUG); } else if (is_method("PRACK")) { setflag(FLAG_DEBUG); } else if (is_method("REFER")) { setflag(FLAG_DEBUG); } else { setflag(FLAG_DEBUG); } } }

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Using =~ to get the group identifier with longest match
by Iñaki Baz Castillo 06 Jun '11

06 Jun '11

Hi, I wonder how feasible is the folowing: I have a regular expresion like: REGEXP = "^((1|20)|(13|3|4)|(5|6))" There are 3 matching groups: 1) (1|20) 2) (13|3|4) 3) (5|6) I want to match the RURI user, let's suppose $rU = "138787686223". In this case the above regexp matches groups 1 and 2, but since group 2 matching is longest ("13" is longest than "1") I want to get value 2. In case $rU = "999999" I need to get "null", in case $rU = "1000022323" I need to get "1". AFAIK there is no way to do it with current regular expression utils in Kamailio, but maybe I miss something. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Questino about dialplan module
by Gary Chen 06 Jun '11

06 Jun '11

Kamailio 3.1.2 I am testing out dialplan module. Every time when I issue 'kamctl dialplan show', it shows what is in the database even if I have not yet issue "kamctl dialplan reload' after I changed data in database. I thought that dialplan is stored in the memory and you have to issue the reload before you can see the change. Why the ' kamctl dialplan show' display the data directly from mysql database? Does that mean that dialplan data is not stored in the memory? Gary

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callback function of tm module
by Bruno Bresciani 06 Jun '11

06 Jun '11

Hi All, I'm having doubts in the implementation of the callback function module tm. As she calls the function to record register_tmcb ()? Can someome help me? Regards

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Re: [SR-Users] Questino about dialplan module
by Gary Chen 03 Jun '11

03 Jun '11

I am talking about kamctl dialplan command not mi. When you type in kamctl, it lists a lot commands. Are these commands still valid? See below: kamctl ../../sbin/kamctl 3.1.0 Existing commands: -- command 'start|stop|restart' restart ............................ restart Kamailio start .............................. start Kamailio stop ............................... stop Kamailio -- command 'acl' - manage access control lists (acl) acl show [<username>] .............. show user membership acl grant <username> <group> ....... grant user membership (*) acl revoke <username> [<group>] .... grant user membership(s) (*) -- command 'lcr' - manage least cost routes (lcr) lcr show_gws....... show database gateways lcr show_routes.... show database routes lcr dump_gws....... show in memory gateways lcr dump_routes.... show in memory routes lcr reload ........ reload lcr gateways and routes lcr eval_weights .. evaluates probability for given GW's weights -- command 'cr' - manage carrierroute tables cr show ....................................................... show tables cr reload ..................................................... reload tables cr dump ....................................................... show in memory tables cr addcn <carrier id> <carrier name> .......................... add a carrier name cr rmcn <carrier id> ......................................... rm a carrier name cr adddn <domain id> <domain name> ............................ add a domain name cr rmdn <domain id> .......................................... rm a domain name cr addcarrier <carrier> <scan_prefix> <domain> <rewrite_host> ................ <prob> <strip> <rewrite_prefix> <rewrite_suffix> ............... <flags> <mask> <comment> .........................add a carrier (prob, strip, rewrite_prefix, rewrite_suffix,................... flags, mask and comment are optional arguments) ............... cr rmcarrier <carrier> <scan_prefix> <domain> ................ rm a carrier -- command 'rpid' - manage Remote-Party-ID (RPID) rpid add <username> <rpid> ......... add rpid for a user (*) rpid rm <username> ................. set rpid to NULL for a user (*) rpid show <username> ............... show rpid of a user -- command 'add|passwd|rm' - manage subscribers add <username> <password> .......... add a new subscriber (*) passwd <username> <passwd> ......... change user's password (*) rm <username> ...................... delete a user (*) -- command 'add|dump|reload|rm|show' - manage trusted trusted show ...................... show db content trusted dump ...................... show cache content trusted reload .................... reload db table into cache trusted add <src_ip> <proto> <from_pattern> <tag> ....................... add a new entry ....................... (from_pattern and tag are optional arguments) trusted rm <src_ip> ............... remove all entres for the given src_ip -- command 'dispatcher' - manage dispatcher * Examples: dispatcher addgw 1 sip:1.2.3.1:5050 1 'outbound gateway' * dispatcher addgw 2 sip:1.2.3.4:5050 3 '' * dispatcher rmgw 4 dispatcher show ..................... show dispatcher gateways dispatcher reload ................... reload dispatcher gateways dispatcher dump ..................... show in memory dispatcher gateways dispatcher addgw <setid> <destination> <flags> <description> .......................... add gateway dispatcher rmgw <id> ................ delete gateway -- command 'db' - database operations db exec <query> ..................... execute SQL query db roexec <roquery> ................. execute read-only SQL query db run <id> ......................... execute SQL query from $id variable db rorun <id> ....................... execute read-only SQL query from $id variable db show <table> ..................... display table content -- command 'speeddial' - manage speed dials (short numbers) speeddial show <speeddial-id> ....... show speeddial details speeddial list <sip-id> ............. list speeddial for uri speeddial add <sip-id> <sd-id> <new-uri> [<desc>] ... ........................... add a speedial (*) speeddial rm <sip-id> <sd-id> ....... remove a speeddial (*) speeddial help ...................... help message - <speeddial-id>, <sd-id> must be an AoR (username@domain) - <sip-id> must be an AoR (username@domain) - <new-uri> must be a SIP AoR (sip:username@domain) - <desc> a description for speeddial -- command 'avp' - manage AVPs avp list [-T table] [-u <sip-id|uuid>] [-a attribute] [-v value] [-t type] ... list AVPs avp add [-T table] <sip-id|uuid> <attribute> <type> <value> ............ add AVP (*) avp rm [-T table] [-u <sip-id|uuid>] [-a attribute] [-v value] [-t type] ... remove AVP (*) avp help .................................. help message - -T - table name - -u - SIP id or unique id - -a - AVP name - -v - AVP value - -t - AVP name and type (0 (str:str), 1 (str:int), 2 (int:str), 3 (int:int)) - <sip-id> must be an AoR (username@domain) - <uuid> must be a string but not AoR -- command 'alias_db' - manage database aliases alias_db show <alias> .............. show alias details alias_db list <sip-id> ............. list aliases for uri alias_db add <alias> <sip-id> ...... add an alias (*) alias_db rm <alias> ................ remove an alias (*) alias_db help ...................... help message - <alias> must be an AoR (username@domain)" - <sip-id> must be an AoR (username@domain)" -- command 'domain' - manage local domains domain reload ....................... reload domains from disk domain show ......................... show current domains in memory domain showdb ....................... show domains in the database domain add <domain> ................. add the domain to the database domain rm <domain> .................. delete the domain from the database -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart <uri> ................ restart phone configured for <uri> -- command 'online' - dump online users from memory online ............................. display online users -- command 'monitor' - show internal status monitor ............................ show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping <uri> ......................... ping <uri> with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [<username>]................ show in-RAM online users ul show --brief..................... show in-RAM online users in short format ul rm <username> [<contact URI>].... delete user's usrloc entries ul add <username> <uri> ............ introduce a permanent usrloc entry ul add <username> <uri> <expires> .. introduce a temporary usrloc entry -- command 'fifo' fifo ............................... send raw FIFO command -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart <uri> ................ restart phone configured for <uri> -- command 'online' - dump online users from memory online ............................. display online users -- command 'monitor' - show internal status monitor ............................ show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping <uri> ......................... ping <uri> with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [<username>]................ show in-RAM online users ul show --brief..................... show in-RAM online users in short format ul rm <username> [<contact URI>].... delete user's usrloc entries ul add <username> <uri> ............ introduce a permanent usrloc entry ul add <username> <uri> <expires> .. introduce a temporary usrloc entry -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Juha Heinanen Sent: Friday, June 03, 2011 3:50 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List Subject: [SR-Users] Questino about dialplan module Gary Chen writes: > I am testing out dialplan module. Every time when I issue 'kamctl > dialplan show', it shows what is in the database even if I have not > yet issue "kamctl dialplan reload' after I changed data in database. I > thought that dialplan is stored in the memory and you have to issue > the reload before you can see the change. Why the ' kamctl dialplan > show' display the data directly from mysql database? Does that mean > that dialplan data is not stored in the memory? there is no mi function to show dialplan rules in memory. i have no idea, what 'show' does. please read readme of dialplan module before asking questions about the module. -- juha _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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IRC meeting today, 15:00GMT
by Daniel-Constantin Mierla 03 Jun '11

03 Jun '11

Hello, everyone that wants to discuss about the next major release (v3.2.0), please join us on IRC: Channel: #sip-router Server: irc.freenode.net Time: 15:00GMT/London (16:00 Berlin/Paris/Madrid, 10:00am New York) Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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IMC module - #join to private room crashes kamailio
by Ozren Lapcevic 03 Jun '11

03 Jun '11

Hi, there is a bug in IMC module that crashes Kamailio. Here is the scenario: One user (pero(a)abc.hr) creates private chat room ( chat-999(a)abc.hr). Other user (oz(a)abc.hr) tries to join the chat-999(a)abc.hrby sending #join. Kamailio crashes at that point. Here is the log: Best regards, Ozren Jun 1 15:10:37 ser-test kernel: [5844312.456709] kamailio[20118]: segfault at 10 ip 7f28f89dae8e sp 7fffffffb750 error 4 in imc.so[7f28f89cf000+17000] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: INFO: <script>: going to imc: F [sip:oz@abc.hr] T [sip:chat-999@abc.hr] B [#join] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [receive.c:186]: preparing to run routing scripts... Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc [imc.c:525]: found command Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc [imc_cmd.c:146]: command: [join] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=223 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc [imc_cmd.c:292]: found room [chat-999] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: ERROR: imc [imc_cmd.c:317]: attept to join private room [chat-999] from user [oz] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5e3288a6 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT: <core> [main.c:741]: child process 20118 exited by a signal 11 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG: <core> [parser/msg_parser.c:630]: SIP Request: Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT: <core> [main.c:744]: core was not generated Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [msg_translator.c:204]: check_via_address(161.53.255.82, 161.53.255.82, 0) Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG: <core> [parser/msg_parser.c:632]: method: <MESSAGE> Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20114]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20115]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20116]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20117]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20122]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20124]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20125]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20129]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20128]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20127]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20123]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20126]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20120]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20130]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20131]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc [imc.c:709]: room 3 chat-777 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc [imc.c:709]: room 11 chat-888 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc [imc.c:709]: room 13 chat-999

2 6

perlvdb error
by Roman Yeryomin 03 Jun '11

03 Jun '11

On 18 April 2011 15:47, Henning Westerholt <henning.westerholt(a)1und1.de> wrote: > On Monday 18 April 2011, Roman Yeryomin wrote: >> coredump? easy :) >> you can pick core file itself here (it's pretty heavy to send to the list): >> http://83.99.159.113/core.kamailio.20110418-1331 >> >> What do you mean by backtrace? log output or strace output? > > Hi Roman, > > he refers to a gdb backtrace, "gdb bt". More informations e.g. here: > http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:corefiles > ok, here is the backtrace: # gdb kamailio core GNU gdb 6.8-debian Copyright (C) 2008 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "i486-linux-gnu"... warning: Can't read pathname for load map: Input/output error. Reading symbols from /lib/tls/i686/cmov/libdl.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libdl.so.2 Reading symbols from /lib/tls/i686/cmov/libresolv.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libresolv.so.2 Reading symbols from /lib/tls/i686/cmov/libc.so.6...done. Loaded symbols for /lib/tls/i686/cmov/libc.so.6 Reading symbols from /lib/ld-linux.so.2...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/local/lib/kamailio/modules_k/mi_fifo.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/mi_fifo.so Reading symbols from /usr/local/lib/kamailio/libkmi.so.1...done. Loaded symbols for /usr/local/lib/kamailio/libkmi.so.1 Reading symbols from /usr/local/lib/kamailio/modules_k/kex.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/kex.so Reading symbols from /usr/local/lib/kamailio/libkcore.so.1...done. Loaded symbols for /usr/local/lib/kamailio/libkcore.so.1 Reading symbols from /usr/local/lib/kamailio/modules/tm.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/tm.so Reading symbols from /usr/local/lib/kamailio/modules_k/tmx.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/tmx.so Reading symbols from /usr/local/lib/kamailio/modules/sl.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/sl.so Reading symbols from /usr/local/lib/kamailio/modules_k/rr.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/rr.so Reading symbols from /usr/local/lib/kamailio/modules_k/pv.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/pv.so Reading symbols from /usr/local/lib/kamailio/modules_k/maxfwd.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/maxfwd.so Reading symbols from /usr/local/lib/kamailio/modules_k/usrloc.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/usrloc.so Reading symbols from /usr/local/lib/kamailio/libsrdb1.so.1...done. Loaded symbols for /usr/local/lib/kamailio/libsrdb1.so.1 Reading symbols from /usr/local/lib/kamailio/modules_k/registrar.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/registrar.so Reading symbols from /usr/local/lib/kamailio/modules_k/textops.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/textops.so Reading symbols from /usr/local/lib/kamailio/modules_k/siputils.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/siputils.so Reading symbols from /usr/local/lib/kamailio/modules_k/xlog.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/xlog.so Reading symbols from /usr/local/lib/kamailio/modules/sanity.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/sanity.so Reading symbols from /usr/local/lib/kamailio/modules/ctl.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/ctl.so Reading symbols from /usr/local/lib/kamailio/modules/mi_rpc.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/mi_rpc.so Reading symbols from /usr/local/lib/kamailio/modules_k/acc.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/acc.so Reading symbols from /lib/tls/i686/cmov/libpthread.so.0...done. Loaded symbols for /lib/tls/i686/cmov/libpthread.so.0 Reading symbols from /usr/local/lib/kamailio/modules_k/perl.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/perl.so Reading symbols from /usr/lib/libperl.so.5.8...done. Loaded symbols for /usr/lib/libperl.so.5.8 Reading symbols from /lib/tls/i686/cmov/libm.so.6...done. Loaded symbols for /lib/tls/i686/cmov/libm.so.6 Reading symbols from /lib/tls/i686/cmov/libcrypt.so.1...done. Loaded symbols for /lib/tls/i686/cmov/libcrypt.so.1 Reading symbols from /usr/local/lib/kamailio/modules_k/perlvdb.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/perlvdb.so Reading symbols from /usr/local/lib/kamailio/modules/auth.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/auth.so Reading symbols from /usr/local/lib/kamailio/modules_k/auth_db.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/auth_db.so Reading symbols from /usr/local/lib/kamailio/modules/app_lua.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/app_lua.so Reading symbols from /usr/lib/liblua5.1.so.0...done. Loaded symbols for /usr/lib/liblua5.1.so.0 Reading symbols from /lib/tls/i686/cmov/libnss_files.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libnss_files.so.2 Reading symbols from /lib/tls/i686/cmov/libnss_dns.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libnss_dns.so.2 Core was generated by `kamailio -dddd'. Program terminated with signal 11, Segmentation fault. [New process 13773] #0 0xb7228e3a in parseurl (url=0xb7213230 "T�0\b#") at perlvdbfunc.c:60 60 if (strlen(cn) > 0) If I understand this correctly error refers to row 60 in perlvdbfunc.c strlen expects const char* instead of char* ? Regards, Roman

2 17

Re: [SR-Users] Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

That works. Thanks. From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Thursday, June 02, 2011 4:28 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gary Chen Subject: Re: [SR-Users] Problem using dispatcher. On 6/2/11 9:41 PM, Gary Chen wrote: Thanks for the quick reply. How can I fix this? When you set t_on_failure for an INVITE, the respective failure route will be executed for any reply >=0. What you have to do is to not re-route to a new destination from failure route, simply exit for those reply codes, like: if(t_check_status("401|407")) exit; You don't handle as well the canceled transactions, when re-routing makes no sense -- you should do it (see default config file for example. Cheers, Daniel We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. Any suggestion? -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [ mailto:sr-users-bounces@lists.sip-router.org ] On Behalf Of Alex Balashov Sent: Thursday, June 02, 2011 3:30 PM To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Problem using dispatcher. On 06/02/2011 03:26 PM, Gary Chen wrote: > the dispatcher always get into FAILURE route although the log shows > that kamailio did send INVITE to 10.10.1.222 and received '407 - > Proxy Authentication Required' from Freexwitch1 and then dispatcher > send INVITE to 10.10.1.223 and also received '407 - Proxy > Authentication Required' from Freexwitch2 and eventually failed. That's because any reply >= 300 is considered a failure. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

Thanks for the quick reply. How can I fix this? We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. Any suggestion? -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Alex Balashov Sent: Thursday, June 02, 2011 3:30 PM To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Problem using dispatcher. On 06/02/2011 03:26 PM, Gary Chen wrote: > the dispatcher always get into FAILURE route although the log shows > that kamailio did send INVITE to 10.10.1.222 and received '407 - > Proxy Authentication Required' from Freexwitch1 and then dispatcher > send INVITE to 10.10.1.223 and also received '407 - Proxy > Authentication Required' from Freexwitch2 and eventually failed. That's because any reply >= 300 is considered a failure. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

Kamailio 3.1.2 Here is my setup: Phone1 (10.10.1.114) | Kamailio (10.10.1.230) / \ / \ Freeswitch1 Freeswitch2 ----------------------> Other phones (10.10.1.222) (10.10.1.223) Both Freeswitch's share the same REGISTRATION information. Phone1 is already registered with Freeswitch. When I try to dial 'Other phones' using phone1, the dispatcher always get into FAILURE route although the log shows that kamailio did send INVITE to 10.10.1.222 and received '407 - Proxy Authentication Required' from Freexwitch1 and then dispatcher send INVITE to 10.10.1.223 and also received '407 - Proxy Authentication Required' from Freexwitch2 and eventually failed. Any idea? Here is my configuration: modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 10000) modparam("dispatcher", "ds_probing_threshhold", 3) #modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:proxy@SIP-router1.testnet.net") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "flags", 3) modparam("dispatcher", "force_dst", 1) route { .... if ( ds_select_dst("$var(group)", "9") ){ route(RELAY_DS); return; ... } route[[RELAY_DS] if (is_method("INVITE")) { t_on_branch("BRANCH_ONE"); t_on_reply("NAT_REPLY"); t_on_failure("FAIL_DS"); } if (!t_relay()) { sl_reply_error(); } } failure_route[FAIL_DS] { if (t_check_status("408")) { ds_mark_dst("i"); } if (!ds_next_dst()) { t_reply("503", "Service unavailable: no more dst"); exit; } t_on_failure("FAIL_DS"); route(RELAY_DS); } onreply_route[NAT_REPLY] { xlog("L_INFO", "REPLY_ROUTE: S=$rs - $rr F=$fu T=$tu IP=$avp(s:ip):$avp(s:port) ($si:$sp) ID=$ci\n"); } And here is the SIP messages from ngrep: U 10.10.1.144:51029 -> 10.10.1.230:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..F rom: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet .net;user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 70..Date: Thu, 02 Jun 2011 19:14:14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires: 180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces..C ontent-Length: 261..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.144..s=SIP Call..t=0 0..m=audio 20372 RTP/AVP 0 8 18 101..c=IN IP4 10.10.1.144..a=rtpmap: 0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv .. # U 10.10.1.230:5060 -> 10.10.1.144:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net; user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..Server: kamailio (3.1.2 (x8 6_64/linux))..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.222:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Record-Route: <sip:10.10.1.230;lr=on;nat=yes>..Via: SIP/2.0/ UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "L ine1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;us er=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..Date: Thu, 02 Jun 2011 19:14: 14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires: 180..A ccept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces..Content- Length: 277..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.231..s=SIP Call..t=0 0..m=audio 38426 RTP/AVP 0 8 18 101..c=IN IP4 10.10.1.231..a=rtpmap:0 PCMU/80 00..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..a=nortp proxy:yes.. # U 10.10.1.222:5060 -> 10.10.1.230:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/UDP 10.10.1.144:5060 ;branch=z9hG4bK116d554d..Record-Route: <sip:10.10.1.230;lr=on;nat=yes>..From: "Line1" <sip:5022441011@fs2000.lightyea r.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>..Call-ID: 00082166-efcb0013 -692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 0 3-05-32 -0400..Content-Length: 0.... # U 10.10.1.222:5060 -> 10.10.1.230:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/U DP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb0021 53bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>;tag=6mjQm2eayZ49j..Call-ID: 00082166-efcb0013-692f44d9 -6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 03-05-32 - 0400..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-se ize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest rea lm="fs2000.testnet.net", nonce="8da1ddb1-e393-442c-8be0-17e439457363", algorithm=MD5, qop="auth"..Content-Length: 0... . # U 10.10.1.230:5060 -> 10.10.1.222:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Fro m: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.n et;user=phone>;tag=6mjQm2eayZ49j..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..CSeq: 101 ACK..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.223:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Record-Route: <sip:10.10.1.230;lr=on;nat=yes;nat=yes>..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d.. From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.lightyea r.net;user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..Date: Thu, 02 Jun 201 1 19:14:14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires : 180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces.. Content-Length: 328..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.23110.10.1.231..s=SIP Call..t=0 0..m=audio 3842638426 RTP/AVP 0 8 18 101..c=IN IP4 216.49. 129.23110.10.1.231..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8 000..a=fmtp:101 0-15..a=sendrecv..a=nortpproxy:yes..a=nortpproxy:yes.. # U 10.10.1.223:5060 -> 10.10.1.230:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/UDP 10.10.1.144:5060 ;branch=z9hG4bK116d554d..Record-Route: <sip:10.10.1.230;lr=on;nat=yes;nat=yes>..From: "Line1" <sip:5022441011@fs2000. testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>..Call-ID: 00082166- efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011 -04-19 03-05-32 -0400..Content-Length: 0.... # U 10.10.1.223:5060 -> 10.10.1.230:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/U DP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb0021 53bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>;tag=Ky0XB4gDeDy6F..Call-ID: 00082166-efcb0013-692f44d9 -6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 03-05-32 - 0400..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-se ize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest rea lm="fs2000.testnet.net", nonce="8769298b-c52d-43b8-b69c-6ebf26f3184d", algorithm=MD5, qop="auth"..Content-Length: 0... . # U 10.10.1.230:5060 -> 10.10.1.223:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Fro m: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.n et;user=phone>;tag=Ky0XB4gDeDy6F..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..CSeq: 101 ACK..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.144:5060 SIP/2.0 503 Service unavailable: no more dst..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1 " <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=p hone>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..Server: kamailio (3.1.2 (x86_64/linux))..Content-Length: 0.... # U 10.10.1.144:51030 -> 10.10.1.230:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From : "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.ne t;user=phone>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..D ate: Thu, 02 Jun 2011 19:14:14 GMT..CSeq: 101 ACK..Content-Length: 0.... Gary Chen

2 1

discussing about next major release - v3.2.0
by Daniel-Constantin Mierla 02 Jun '11

02 Jun '11

Hello, we should decide the roadmap to next major release v3.2.0, so probably an IRC meeting for some real time chatting would be good. I am thinking of June 3 or 4, 15:00GMT. Anyone interested in the meeting and available in one of those dates? The date with most of devs available will be selected. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 7

RTP traffic
by Lucas Alvarez 01 Jun '11

01 Jun '11

Hi, I'm having a problem with RTP traffic, I have softphone connecting from the outside, I can register but Kamailio is sending all RTP traffic to the private IP address that the softphone has in it own local network. How can I fix this? Any help will be appreciated. Lucas

2 2

serweb & xSER 3.1
by Skyler 01 Jun '11

01 Jun '11

Hi all, Wondering if anyone has serweb working/functioning with Kamailio or SER 3.1 branch? If so which version of serweb are you using 2.0.0 or 2.1.0-dev1? Any issues or tips for first time install/setup of serweb? S.

1 0

has_tran_tmcbs function
by Bruno Bresciani 01 Jun '11

01 Jun '11

Hi, I can't understand the has_tran_tmcbs function... Can someone tell me how it works? Regards,

2 1

Kamailio Tables with index
by Klaus Darilion 01 Jun '11

01 Jun '11

Hi! With Openser 1.2 all the tables got an auto-increment-index column. Is this index column just for layout/performance purposes or is it also used somewhere insides the Kamailio code? If not (what I expect), then Kamailio 3.1 should still work with tables without those columns - correct? Thanks Klaus

2 1

ClueCon 2011 News and Notes
by marketing＠cluecon.com 31 May '11

31 May '11

1 0

Issue using pua_bla module
by Watkins, Bradley 31 May '11

31 May '11

I'm trying to get the pua_bla module to do its thing, and I'm running into an issue. I'm getting these errors when the first NOTIFY comes from the phone: ERROR: presence [publish.c:515]: Missing or unsupported event header field value ERROR: presence [publish.c:518]: event=[dialog;sla] I see the phone register and the subsequent SUBSCRIBE (both successful) from Kamailio, so this bit seems to be correct. I'm loading what I think are all the necessary modules: loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "pua.so" loadmodule "pua_bla.so" And I'm handling the NOTIFY as per the documentation (xlogs to see that in fact the script executes): if(is_method("NOTIFY") && to_uri=~"bla_aor(a)domain.com") { xlog("Matched bla_aor to_uri\n"); if(bla_handle_notify()) { xlog("Handled NOTIFY\n"); t_reply("200", "OK"); } } I see the log entries from the above xlogs, so it certainly is executing this part of the script. Any ideas why I'm getting the errors? All ideas and/or assistance are greatly appreciated. Thanks in advance, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

3 6

Siremis: changes to Kamailio DB
by Klaus Darilion 31 May '11

31 May '11

Hi! I just read the Siremis installation howto. One of the task of the installer is: "Update SIP Database - this updates the structure of Kamailio database for accounting and charts modules" I wonder what siremis exactly does to the DB. Is greped over the siremis source file but couldn't find the changes. Are they somewhere documented? (e.g. .sql file with the statemens). Anyway, I wonder changes are needed at all. IMO, if possible, the Kamailio DB should not be changed by an external too - otherwise this may cause problems with updates to the Kamailio DB structure. Thanks Klaus

3 3

kamailio debian package bug report : 3.1.3+lenny2
by Philippe HENSEL 31 May '11

31 May '11

Hi developpers and packagers, The file "/usr/lib64/kamailio/libsrdb2.so.1.0" seems to be included in both "kamailio_3.1.3+lenny2_amd64.deb" and "kamailio-mysql-modules_3.1.3 +lenny2_amd64.deb". So it is not possible to install "kamailio-mysql-modules". Thanks for help

5 8

Multidomains - quick question
by Lucas Alvarez 31 May '11

31 May '11

Hi, if I want to register clients from outside of my local network, do I have to enable the domain module and add the external IP as a new domain? Thanks in advance, Lucas Alvarez

2 2

acc Locks
by Borin 31 May '11

31 May '11

Hi all, Recently I met the following problem with kamailio acc table. We are using kamailio for calls dispatching to a couple of rtp gateways and for call statistics acc is used (with mysql). I tried to delete some old entries from acc table and for this time mysql "show processes" showed a lot of entries like " kamailio | Query | 0 | System lock | insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,start_time,src_addr,dst_addr,src_ |". I think that meant that kamailio could not write to the acc table for the time the data were being deleted. And it also did not forward INVITE requests further. Does that mean that if kamailio does not manage to write acc record for any reason it does not forward that INVITE/200ok? Best Regards, Kate

2 4

Green VoIP - energy efficiency and performaces of v3.0
by Daniel-Constantin Mierla 31 May '11

31 May '11

Hello, Jan Janak conducted a very interesting research project regarding energy efficiency of VoIP systems during 2010, a collaboration between iptel.org and Columbia University. The team used the source code from sip-router.org GIT repository from January 2010, which corresponds to Kamailio (former OpenSER) and SER v3.0. The latest stable series v3.1 shares the same internal architecture with v3.0. As part of the research work, Jan could also gather some figures about capacity and performances of v3.0 with a quite complex configuration file: etc/sip-router-oob.cfg (involving authentication and NAT traversal as well). You can read the paper about energy efficiency at: - Green VoIP Article: http://asipto.com/u/2j The draft notes about capacity and performances of v3.0 are available at: - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k Some interesting results: - one instance of SIP server with 500 000 online users (mixed users – behind and not NAT routers) – consumed energy 210W - one instance of SIP server with 1 000 000 online users (no NAT involved) – consumed energy 190W - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for SIP server, the server could support 43 000 simultaneous TLS connections – consumed energy 203W - one SIP server instance with 80 000 permanent TCP connections, the SIP server could still handle at least 1000 requests per second and a connection arrival rate of 1000 new connections per second, done for 20 000 new connections. CPU load generated by the SIP server was from 6% to 8%. I added a new section to the draft notes to list the enhancements done for the latest stable release (v3.1.x) that contribute to performance improvements, like asynchronous TLS, fine tuning of memory for TLS connections and raw UDP sockets. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

5 9

Late parallel branch (without canceling previous pending branches)
by Iñaki Baz Castillo 31 May '11

31 May '11

Hi, just wondering whether the following scenario (which it's not possible in kamailio 1.5) is possible in 3.X: 1) Alice calls a PSTN number. 2) Proxy receives the call and routes it to a media server which plays an early-announcement (30 seconds long). 3) After N seconds (maybe 10 seconds) proxy starts a new parallel branch to the PSTN gateway *without* cancelling the previous branch with the media server. 4) Let's suppose the PSTN phone rings and proxy gets 183 with SDP (ringing in real audio). Proxy converts such 183 into a 180 with no SDP (already possible in Kamailio). 5) After 30 seconds media server rejects the call with 480. PSTN branch remains ringing as usual. I just would like to know if Kamailio 3.X allows point 3. I think it's hard but maybe I miss some feature. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

Dispatcher variables
by Lee Archer 31 May '11

31 May '11

Hi all, I am using dispatcher and on a failure I would like to log the server that has failed. I am using $dd but this only ever returns the address of the first server that was tried and not any subsequent servers that were tried and also failed. What should I be using to display the IP address of the server dispatcher is about to use? Thanks Lee

2 2

Fw: Load capacity of SER??
by Piyush.Bansal＠relianceada.com 31 May '11

31 May '11

Dear All, I'm using SER as a SIP proxy server for quite a long time. I'm not been able to find its load bearing capacity. If you can help me get a figure in terms of BHCA, that'll be quite helpful. Thanks and Regards, --Piyush Bansal The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

1 0

IMC module doesn't save data to mysql database
by Ozren Lapcevic 30 May '11

30 May '11

Hi all, I'm trying to configure IMC module and get it to work properly but I've run into problems with saving IMC module data to mysql database. I'm using following code provided in IMC module documentation: modparam("imc", "db_url", DBURL) ## route block ### if(is_method("MESSAGE") && (uri=~ "sip:chat-[0-9]+@") || (uri=~ "sip:chat-manager@")) { xlog("L_INFO", "going to imc: F [$fu] T [$tu] B [$rb]\n"); if(imc_manager()) sl_send_reply("200", "ok - imc_manager()"); else sl_send_reply("500", "command error"); exit; } I am creating chat room with following message sent to chat-manager(a)abc.hr: "#create chat-111" DEBUG log returns following: May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: INFO: <script>: going to imc: F [sip:oz@abc.hr] T [sip:chat-manager@abc.hr] B [#create chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc.c:525]: found command May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:146]: command: [create] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:151]: parameter 0=[chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:171]: new room [chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:186]: added room uri= sip:chat-111@abc.hr May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_mng.c:322]: [uri]= sip:oz@abc.hr May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_mng.c:326]: [user]= oz May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:196]: added the owner as the first member [ sip:oz@abc.hr] "./kamctl fifo imc_list_rooms" returns following: ROOM:: URI=sip:chat-111@abc.hr MEMBERS=0 OWNER=sip:oz@abc.hr "./kamctl fifo imc_list_members sip:chat-111@abc.hr" returns following: ROOM:: sip:chat-111@abc.hr NR_OF_MEMBERS=1 MEMBER:: sip:oz@abc.hr However, that data is not persisted/saved in the imc_rooms and imc_members tables in the database. Any clues what might be the problem here? Any help is greatly appreciated. Best regards, Ozren Lapcevic PS I'm runing Kamailio 3.1.4 with default cfg file with enabled mysql, multidomain, usrlocdb, auth and NAT. I can register clients, establish calls, send IMs, etc. imc_rooms and imc_members tables were created when kamailio was installed. I've also checked imc-create.sql script and tables are correctly created. The same DBURL used for imc is also used for auth_db, domain and usrloc modules and I haven't experienced any problems with those modules and database connection/persistence. domain, location, version and subscriber tables are correctly populated. I'm not getting any errors from imc_manager(), it returns 200 OK. I can't trace any errors in DEBUG related to problems with mysql database.

1 0

Kamailio service not starting on reboot when using mysql database
by Rick Aldred 30 May '11

30 May '11

I have set up kamailio 3.1.2 on an install of Debian Squeeze, using the default config but with the following defines enabled: WITH_MYSQLWITH_AUTHWITH_USRLOCDBWITH_NAT Everything works as expected when starting Kamailio from a terminal, e.g.service kamailio start But after a reboot, the service fails to start and all SIP requests are dropped with a ICMP port unreachable error. The following is an extract of /var/log/syslog when this problem occurs. It looks like Kamailio is failing to access the mysql socket, as the mysql service has not yet started? Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2)Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [db.c:289]: could not add connection to the poolApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: usrloc [dlist.c:486]: failed to open database connectionApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: registrar [reg_mod.c:396]: failed to register domainApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:569Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:572Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:574Apr 12 10:08:34 SIPServer mysqld_safe: Starting mysqld daemon with databases from /var/lib/mysqlApr 12 10:08:34 SIPServer mysqld: 110412 10:08:34 [Note] Plugin 'FEDERATED' is disabled.Apr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 InnoDB: Started; log sequence number 0 44233Apr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 [Note] Event Scheduler: Loaded 0 eventsApr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 [Note] /usr/sbin/mysqld: ready for connections.Apr 12 10:08:36 SIPServer mysqld: Version: '5.1.49-3' socket: '/var/run/mysqld/mysqld.sock' port: 3306 (Debian) Both the Kamailio and mysql services start with priority 18 on run levels 2345. I tried adjusting the Kamailio service to priority 19, but the problem still occurs. I am relatively new to Kamailio and mysql as well as Linux / Debian administration, so I'm not really sure how to solve this problem. Any ideas would be greatly appreciated. Regards,Rick Aldred

5 10

kamailio and radius
by Asun 30 May '11

30 May '11

Hello I've installed kamailio 3.1 but It's doesn't Work with radius authentication. I've followed this manual <http://www.kamailio.org/docs/openser-radius-1.0.x.html> http://www.kamailio.org/docs/openser-radius-1.0.x.html but it does'nt work, because is for openser not kamailio. Is there another manual to do this? Anyone have radius authentication with kamailio? Can anyone help me? Thank you Asun

3 8

Presence changes not seen for first time login
by Swetha Raj 30 May '11

30 May '11

Hi All, I have Kamailio3.1 set up by following the steps mentioned here - http://kb.asipto.com/kamailio:presence:k31-made-simple. When i login two users via Jitsi and add each other,one user is not able to see the other user's presence changes.Below are the steps to reproduce - 1.Login user1@sipdomain via jitsi 2.Login user2@sipdomain vis jitsi 3.user1 adds user2,user2 receives the request and accepts it. 4.user2 adds user1. 5.Now any presence changes from user1 is visible to user2 But any changes from user2 is not visible to user1. This issue is resolved if the users re-login.Any ideas what could be the issue ? Also how do i see the complete SIP headers in kamailio log,is their a way to enable it ? Thanks, Swetha

3 4

Invitation to connect on LinkedIn
by Anand Kumar via LinkedIn 28 May '11

28 May '11

LinkedIn ------------ Anand Kumar requested to add you as a connection on LinkedIn: ------------------------------------------ Luis, I'd like to add you to my professional network on LinkedIn. - Anand Accept invitation from Anand Kumar http://www.linkedin.com/e/-y8k3ym-go88enz1-1n/XkppUi8mK5TI9VJsZIpnfBovln4Su… View invitation from Anand Kumar http://www.linkedin.com/e/-y8k3ym-go88enz1-1n/XkppUi8mK5TI9VJsZIpnfBovln4Su… ------------------------------------------ DID YOU KNOW you can conduct a more credible and powerful reference check using LinkedIn? Enter the company name and years of employment or the prospective employee to find their colleagues that are also in your network. This provides you with a more balanced set of feedback to evaluate that new hire. http://www.linkedin.com/e/-y8k3ym-go88enz1-1n/rsr/inv-27/ -- (c) 2011, LinkedIn Corporation

1 0

core kamailio 3.1.2
by Bruno Bresciani 27 May '11

27 May '11

Hi ALL Why kamailio 3.1.2 generate core when function recvmsg return 0? Below is the trace of the core kamailio: May 27 10:29:14 sswpst00 /home2/local/kamailio/sbin/kamailio[31288]: INFO: <script>: CFG: Cseq INVITE ... May 27 10:29:14 sswpst00 /home2/local/kamailio/sbin/kamailio[31288]: DEBUG: tm [t_reply.c:1129]: ->>>>>>>>> T_code=100, new_code=488 May 27 10:29:14 sswpst00 /home2/local/kamailio/sbin/kamailio[31288]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb61304a0, callback type 64, id 0 entered *May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 11* May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: DEBUG: <core> [tcp_main.c:3551]: DBG: handle_ser_child: dead child 1, pid 31288 (shutting down?) May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x82561e0, 11, -1, 0x0) fd_no=18 called *May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31285]: ALERT: <core> [main.c:741]: child process 31288 exited by a signal 11* May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31285]: ALERT: <core> [main.c:744]: core was generated May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31285]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD *May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: INFO: <core> [main.c:807]: INFO: signal 15 received* May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: DEBUG: <core> [main.c:818]: Memory status (pkg): Regards,

1 0

ser_ctl / serweb
by Claudio Furrer 27 May '11

27 May '11

Hi all, Just want to ask if ser_ctl application will be merged into sr3.1 or future releases. By the moment I've found it here [1], [2] or [3]. Is it full compatible with sip-router v3.x? I know kamctl and kamdbctl would be suggested but these are for kamailio flavour. I don't find too much work based on SER flavour regarding to database/users administration and web provisioning. BTW, siremis only works with kamailio db structures, but for sip-router v3 ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if is it working well with the current new releases of the project. I would appreciate if someone can advice me if kamailio is the way to go, because have find that sip-router or ser flavours miss some docs and apps that kamailio already have. I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ [2] http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… [3] http://cvs.berlios.de/viewvc/ser/serctl/ [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ [5] http://developer.berlios.de/projects/serweb/ Thank you, Claudio

3 2

Howto restore a FIFO file?
by Spinov Evgeniy 27 May '11

27 May '11

Hello, everyone. I've accidentally started 2nd copy of Kamailio and killed it at once. However, fifo file was deleted by killed Kamailio, while the first one copy is running fine. Is there any way to restore the fifo file to the first kamailio, unless restarting it? Thank you.

3 2

[OT] IETF SIMPLE WG will destroy MSRP with the new draft-ietf-simple-msrp-sessmatch-11
by Iñaki Baz Castillo 27 May '11

27 May '11

Hi, for those interested in MSRP protocol (instant message sessions and file transfer for SIP) there are bad news: Even if there are already clients and servers (MSRP relays o IM conference servers) implementing the MSRP protocol (RFC 4975 and 4976) the SIMPLE WG will publish a new draft [*] that breaks these RFC's just to satisfy big vendors interested in deploying MSRP capable SBC/ALG boxes (so instead of solving NAT issues with MSRP relays as RFC 4976 states, they want it to be fixed in the router by doing ugly ALG, or in a SBC). This is terrible because all the MSRP devices should implement this new draft in order to interoperate (no backward compatibility at all). The draft also breaks the security defined in RFC 4975 (for example, TLS name based authentication cannot work anymore). For further information I recommend reading these two posts: http://blog.tekelec.com/blog/bid/29816/More-on-MSRP-Session-Match-Extension http://blog.tekelec.com/blog/bid/33138/MSRP-Session-Match-Backwards-Compati… and also these very *hot* mail threads in the SIMPLE maillist: http://www.ietf.org/mail-archive/web/simple/current/msg09227.html http://www.ietf.org/mail-archive/web/simple/current/msg09229.html [*] http://tools.ietf.org/html/draft-ietf-simple-msrp-sessmatch-11 -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 23

Re: [SR-Users] sr-users Digest, Vol 72, Issue 67
by Boris 27 May '11

27 May '11

sr-users-request(a)lists.sip-router.org wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Aliases Using the Default Script; (David J.) > 2. Re: Green VoIP - energy efficiency and performances of v3.0 > (Jan Janak) > 3. Kamailio v3.1.4 Released (Daniel-Constantin Mierla) > 4. Re: ser_ctl / serweb (caio) > 5. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (I?aki Baz Castillo) > 6. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 26 May 2011 14:27:55 -0400 >From: "David J." <david(a)styleflare.com> >Subject: [SR-Users] Aliases Using the Default Script; >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <4DDE9BAB.4060401(a)styleflare.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >I am using the kamailio default script on 3.1.3. > >I was wondering what happens when I added an Alias in dbaliases? > >For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com > >when an invite comes in; it works perfect I got a 200 back. (1001 Device >rings.) > >If I add another alias to a remote server; ie. > >18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com > >I get a 404 back. > >I wonder what happens internally? > >What should I modify to enable this case; > >Thanks. > > > >------------------------------ > >Message: 2 >Date: Thu, 26 May 2011 20:49:39 +0200 >From: Jan Janak <jan(a)ryngle.com> >Subject: Re: [SR-Users] Green VoIP - energy efficiency and > performances of v3.0 >To: miconda(a)gmail.com >Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org> >Message-ID: <BANLkTikUb_iwBDEuqk977-ECmmHo4dEXvQ(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >No, I turned it off. > >-Jan > >On Thu, May 26, 2011 at 11:50, Daniel-Constantin Mierla ><miconda(a)gmail.com> wrote: >> Hello, >> >> out of curiosity, since you used the sources from GIT - was memory debugging >> on? It is usually enabled in master branch and that could have some impact >> in memory usage and performances... >> >> Thanks, >> Daniel >> >> On 5/25/11 3:00 PM, Jan Janak wrote: >>> >>> On Wed, May 25, 2011 at 06:54, Jeremya<jeremy(a)electrosilk.net> ?wrote: >>>> >>>> These figures pale into insignificance compared to the power required >>>> for standard SIP devices - typically 5-8 watts per device multiplied by >>>> the number of devices. >>>> >>>> When you factor in Gigabit Ethernet the power ups significantly. >>>> >>>> Optimisation at the server level is not significant on any scale. >>>> Optimisation on communications power: i.e. end-devices, DSL& ?switches >>>> is where the power savings are important. >>> >>> Sure, the total power consumption of the whole system is dominated by >>> the power consumption of end-point devices, there's no doubt about >>> that and the paper says that. >>> >>> Nevertheless, as an ITSP you are typically paying for the energy >>> consumed by your servers and in that case knowing what you can expect >>> and how many servers you need is useful. Modern data-center servers >>> have significant base-line power consumption and a portion of that >>> needs to be attributed to the SIP service running on those servers. >>> >>> -Jan >>> >>>> Daniel-Constantin Mierla wrote: >>>>> >>>>> Hello, >>>>> >>>>> Jan Janak conducted a very interesting research project regarding >>>>> energy efficiency of VoIP systems during 2010, a collaboration between >>>>> iptel.org and Columbia University. >>>>> >>>>> The team used the source code from sip-router.org GIT repository from >>>>> January 2010, which corresponds to Kamailio (former OpenSER) and SER >>>>> v3.0. The latest stable series v3.1 shares the same internal >>>>> architecture with v3.0. >>>>> >>>>> As part of the research work, Jan could also gather some figures about >>>>> capacity and performances of v3.0 with a quite complex configuration >>>>> file: etc/sip-router-oob.cfg (involving authentication and NAT >>>>> traversal as well). >>>>> >>>>> You can read the paper about energy efficiency at: >>>>> >>>>> - Green VoIP Article: http://asipto.com/u/2j >>>>> >>>>> The draft notes about capacity and performances of v3.0 are available >>>>> at: >>>>> >>>>> - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k >>>>> >>>>> Some interesting results: >>>>> >>>>> - one instance of SIP server with 500 000 online users (mixed users ? >>>>> behind and not NAT routers) ? consumed energy 210W >>>>> - one instance of SIP server with 1 000 000 online users (no NAT >>>>> involved) ? consumed energy 190W >>>>> - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for >>>>> SIP server, the server could support 43 000 simultaneous TLS >>>>> connections ? consumed energy 203W >>>>> - one SIP server instance with 80 000 permanent TCP connections, the >>>>> SIP server could still handle at least 1000 requests per second and a >>>>> connection arrival rate of 1000 new connections per second, done for >>>>> 20 000 new connections. CPU load generated by the SIP server was from >>>>> 6% to 8%. >>>>> >>>>> I added a new section to the draft notes to list the enhancements done >>>>> for the latest stable release (v3.1.x) that contribute to performance >>>>> improvements, like asynchronous TLS, fine tuning of memory for TLS >>>>> connections and raw UDP sockets. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > > > >------------------------------ > >Message: 3 >Date: Thu, 26 May 2011 21:00:42 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >Subject: [SR-Users] Kamailio v3.1.4 Released >To: kamailio <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org >Message-ID: <4DDEA35A.7070606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >Kamailio SIP Server v3.1.4 stable release is out. > >This is a maintenance release of latest stable branch, 3.1, that >includes fixes since release of v3.1.3. There is no change to database >schema or configuration language structure. Deployments running previous >v3.1.x versions are strongly recommended to be upgraded to v3.1.4. > >For more details about version 3.1.4, visit: > >http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ > >Cheers, >Daniel > >-- >Daniel-Constantin Mierla -- http://www.asipto.com >http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >------------------------------ > >Message: 4 >Date: Thu, 26 May 2011 17:59:42 -0300 >From: caio <elcaio(a)gmail.com> >Subject: Re: [SR-Users] ser_ctl / serweb >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTik+dnTp-egK2xy1ua=tPnCcaDuuOA(a)mail.gmail.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Anyone know if ser_ctl (python utility) will be merged into master? Or is >going to be deprecated? >If it still alive, are there any examples/docs of the usage for DB users >provisoining? >kamctl and kamdbctl are recomended for its substitution? > >Which utility will be official for sip-router? >Sorry if it already were discussed, but I didn't find comments about it >since 2010. > >Thank you >Claudio > >On Tue, May 24, 2011 at 11:14 AM, Claudio Furrer <elcaio(a)gmail.com> wrote: > >> Hi all, >> >> Just want to ask if ser_ctl application will be merged into sr3.1 or future >> releases. >> By the moment I've found it here [1], [2] or [3]. >> >> Is it full compatible with sip-router v3.x? >> >> I know kamctl and kamdbctl would be suggested but these are for kamailio >> flavour. I don't find too much work based on SER flavour regarding to >> database/users administration and web provisioning. >> BTW, siremis only works with kamailio db structures, but for sip-router v3 >> ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if >> is >> it working well with the current new releases of the project. >> >> I would appreciate if someone can advice me if kamailio is the way to go, >> because have find that sip-router or ser flavours miss some docs and apps >> that >> kamailio already have. >> >> I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( >> >> [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ >> [2] >> >> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… >> [3] http://cvs.berlios.de/viewvc/ser/serctl/ >> >> [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ >> [5] http://developer.berlios.de/projects/serweb/ >> >> Thank you, >> Claudio >> >

1 0

Re: [SR-Users] sr-users Digest, Vol 72, Issue 67
by Boris 27 May '11

27 May '11

sr-users-request(a)lists.sip-router.org wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Aliases Using the Default Script; (David J.) > 2. Re: Green VoIP - energy efficiency and performances of v3.0 > (Jan Janak) > 3. Kamailio v3.1.4 Released (Daniel-Constantin Mierla) > 4. Re: ser_ctl / serweb (caio) > 5. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (I?aki Baz Castillo) > 6. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 26 May 2011 14:27:55 -0400 >From: "David J." <david(a)styleflare.com> >Subject: [SR-Users] Aliases Using the Default Script; >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <4DDE9BAB.4060401(a)styleflare.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >I am using the kamailio default script on 3.1.3. > >I was wondering what happens when I added an Alias in dbaliases? > >For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com > >when an invite comes in; it works perfect I got a 200 back. (1001 Device >rings.) > >If I add another alias to a remote server; ie. > >18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com > >I get a 404 back. > >I wonder what happens internally? > >What should I modify to enable this case; > >Thanks. > > > >------------------------------ > >Message: 2 >Date: Thu, 26 May 2011 20:49:39 +0200 >From: Jan Janak <jan(a)ryngle.com> >Subject: Re: [SR-Users] Green VoIP - energy efficiency and > performances of v3.0 >To: miconda(a)gmail.com >Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org> >Message-ID: <BANLkTikUb_iwBDEuqk977-ECmmHo4dEXvQ(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >No, I turned it off. > >-Jan > >On Thu, May 26, 2011 at 11:50, Daniel-Constantin Mierla ><miconda(a)gmail.com> wrote: >> Hello, >> >> out of curiosity, since you used the sources from GIT - was memory debugging >> on? It is usually enabled in master branch and that could have some impact >> in memory usage and performances... >> >> Thanks, >> Daniel >> >> On 5/25/11 3:00 PM, Jan Janak wrote: >>> >>> On Wed, May 25, 2011 at 06:54, Jeremya<jeremy(a)electrosilk.net> ?wrote: >>>> >>>> These figures pale into insignificance compared to the power required >>>> for standard SIP devices - typically 5-8 watts per device multiplied by >>>> the number of devices. >>>> >>>> When you factor in Gigabit Ethernet the power ups significantly. >>>> >>>> Optimisation at the server level is not significant on any scale. >>>> Optimisation on communications power: i.e. end-devices, DSL& ?switches >>>> is where the power savings are important. >>> >>> Sure, the total power consumption of the whole system is dominated by >>> the power consumption of end-point devices, there's no doubt about >>> that and the paper says that. >>> >>> Nevertheless, as an ITSP you are typically paying for the energy >>> consumed by your servers and in that case knowing what you can expect >>> and how many servers you need is useful. Modern data-center servers >>> have significant base-line power consumption and a portion of that >>> needs to be attributed to the SIP service running on those servers. >>> >>> -Jan >>> >>>> Daniel-Constantin Mierla wrote: >>>>> >>>>> Hello, >>>>> >>>>> Jan Janak conducted a very interesting research project regarding >>>>> energy efficiency of VoIP systems during 2010, a collaboration between >>>>> iptel.org and Columbia University. >>>>> >>>>> The team used the source code from sip-router.org GIT repository from >>>>> January 2010, which corresponds to Kamailio (former OpenSER) and SER >>>>> v3.0. The latest stable series v3.1 shares the same internal >>>>> architecture with v3.0. >>>>> >>>>> As part of the research work, Jan could also gather some figures about >>>>> capacity and performances of v3.0 with a quite complex configuration >>>>> file: etc/sip-router-oob.cfg (involving authentication and NAT >>>>> traversal as well). >>>>> >>>>> You can read the paper about energy efficiency at: >>>>> >>>>> - Green VoIP Article: http://asipto.com/u/2j >>>>> >>>>> The draft notes about capacity and performances of v3.0 are available >>>>> at: >>>>> >>>>> - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k >>>>> >>>>> Some interesting results: >>>>> >>>>> - one instance of SIP server with 500 000 online users (mixed users ? >>>>> behind and not NAT routers) ? consumed energy 210W >>>>> - one instance of SIP server with 1 000 000 online users (no NAT >>>>> involved) ? consumed energy 190W >>>>> - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for >>>>> SIP server, the server could support 43 000 simultaneous TLS >>>>> connections ? consumed energy 203W >>>>> - one SIP server instance with 80 000 permanent TCP connections, the >>>>> SIP server could still handle at least 1000 requests per second and a >>>>> connection arrival rate of 1000 new connections per second, done for >>>>> 20 000 new connections. CPU load generated by the SIP server was from >>>>> 6% to 8%. >>>>> >>>>> I added a new section to the draft notes to list the enhancements done >>>>> for the latest stable release (v3.1.x) that contribute to performance >>>>> improvements, like asynchronous TLS, fine tuning of memory for TLS >>>>> connections and raw UDP sockets. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > > > >------------------------------ > >Message: 3 >Date: Thu, 26 May 2011 21:00:42 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >Subject: [SR-Users] Kamailio v3.1.4 Released >To: kamailio <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org >Message-ID: <4DDEA35A.7070606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >Kamailio SIP Server v3.1.4 stable release is out. > >This is a maintenance release of latest stable branch, 3.1, that >includes fixes since release of v3.1.3. There is no change to database >schema or configuration language structure. Deployments running previous >v3.1.x versions are strongly recommended to be upgraded to v3.1.4. > >For more details about version 3.1.4, visit: > >http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ > >Cheers, >Daniel > >-- >Daniel-Constantin Mierla -- http://www.asipto.com >http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >------------------------------ > >Message: 4 >Date: Thu, 26 May 2011 17:59:42 -0300 >From: caio <elcaio(a)gmail.com> >Subject: Re: [SR-Users] ser_ctl / serweb >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTik+dnTp-egK2xy1ua=tPnCcaDuuOA(a)mail.gmail.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Anyone know if ser_ctl (python utility) will be merged into master? Or is >going to be deprecated? >If it still alive, are there any examples/docs of the usage for DB users >provisoining? >kamctl and kamdbctl are recomended for its substitution? > >Which utility will be official for sip-router? >Sorry if it already were discussed, but I didn't find comments about it >since 2010. > >Thank you >Claudio > >On Tue, May 24, 2011 at 11:14 AM, Claudio Furrer <elcaio(a)gmail.com> wrote: > >> Hi all, >> >> Just want to ask if ser_ctl application will be merged into sr3.1 or future >> releases. >> By the moment I've found it here [1], [2] or [3]. >> >> Is it full compatible with sip-router v3.x? >> >> I know kamctl and kamdbctl would be suggested but these are for kamailio >> flavour. I don't find too much work based on SER flavour regarding to >> database/users administration and web provisioning. >> BTW, siremis only works with kamailio db structures, but for sip-router v3 >> ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if >> is >> it working well with the current new releases of the project. >> >> I would appreciate if someone can advice me if kamailio is the way to go, >> because have find that sip-router or ser flavours miss some docs and apps >> that >> kamailio already have. >> >> I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( >> >> [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ >> [2] >> >> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… >> [3] http://cvs.berlios.de/viewvc/ser/serctl/ >> >> [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ >> [5] http://developer.berlios.de/projects/serweb/ >> >> Thank you, >> Claudio >> >

1 0

Re: [SR-Users] sr-users Digest, Vol 72, Issue 67
by Boris 27 May '11

27 May '11

sr-users-request(a)lists.sip-router.org wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Aliases Using the Default Script; (David J.) > 2. Re: Green VoIP - energy efficiency and performances of v3.0 > (Jan Janak) > 3. Kamailio v3.1.4 Released (Daniel-Constantin Mierla) > 4. Re: ser_ctl / serweb (caio) > 5. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (I?aki Baz Castillo) > 6. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 26 May 2011 14:27:55 -0400 >From: "David J." <david(a)styleflare.com> >Subject: [SR-Users] Aliases Using the Default Script; >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <4DDE9BAB.4060401(a)styleflare.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >I am using the kamailio default script on 3.1.3. > >I was wondering what happens when I added an Alias in dbaliases? > >For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com > >when an invite comes in; it works perfect I got a 200 back. (1001 Device >rings.) > >If I add another alias to a remote server; ie. > >18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com > >I get a 404 back. > >I wonder what happens internally? > >What should I modify to enable this case; > >Thanks. > > > >------------------------------ > >Message: 2 >Date: Thu, 26 May 2011 20:49:39 +0200 >From: Jan Janak <jan(a)ryngle.com> >Subject: Re: [SR-Users] Green VoIP - energy efficiency and > performances of v3.0 >To: miconda(a)gmail.com >Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org> >Message-ID: <BANLkTikUb_iwBDEuqk977-ECmmHo4dEXvQ(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >No, I turned it off. > >-Jan > >On Thu, May 26, 2011 at 11:50, Daniel-Constantin Mierla ><miconda(a)gmail.com> wrote: >> Hello, >> >> out of curiosity, since you used the sources from GIT - was memory debugging >> on? It is usually enabled in master branch and that could have some impact >> in memory usage and performances... >> >> Thanks, >> Daniel >> >> On 5/25/11 3:00 PM, Jan Janak wrote: >>> >>> On Wed, May 25, 2011 at 06:54, Jeremya<jeremy(a)electrosilk.net> ?wrote: >>>> >>>> These figures pale into insignificance compared to the power required >>>> for standard SIP devices - typically 5-8 watts per device multiplied by >>>> the number of devices. >>>> >>>> When you factor in Gigabit Ethernet the power ups significantly. >>>> >>>> Optimisation at the server level is not significant on any scale. >>>> Optimisation on communications power: i.e. end-devices, DSL& ?switches >>>> is where the power savings are important. >>> >>> Sure, the total power consumption of the whole system is dominated by >>> the power consumption of end-point devices, there's no doubt about >>> that and the paper says that. >>> >>> Nevertheless, as an ITSP you are typically paying for the energy >>> consumed by your servers and in that case knowing what you can expect >>> and how many servers you need is useful. Modern data-center servers >>> have significant base-line power consumption and a portion of that >>> needs to be attributed to the SIP service running on those servers. >>> >>> -Jan >>> >>>> Daniel-Constantin Mierla wrote: >>>>> >>>>> Hello, >>>>> >>>>> Jan Janak conducted a very interesting research project regarding >>>>> energy efficiency of VoIP systems during 2010, a collaboration between >>>>> iptel.org and Columbia University. >>>>> >>>>> The team used the source code from sip-router.org GIT repository from >>>>> January 2010, which corresponds to Kamailio (former OpenSER) and SER >>>>> v3.0. The latest stable series v3.1 shares the same internal >>>>> architecture with v3.0. >>>>> >>>>> As part of the research work, Jan could also gather some figures about >>>>> capacity and performances of v3.0 with a quite complex configuration >>>>> file: etc/sip-router-oob.cfg (involving authentication and NAT >>>>> traversal as well). >>>>> >>>>> You can read the paper about energy efficiency at: >>>>> >>>>> - Green VoIP Article: http://asipto.com/u/2j >>>>> >>>>> The draft notes about capacity and performances of v3.0 are available >>>>> at: >>>>> >>>>> - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k >>>>> >>>>> Some interesting results: >>>>> >>>>> - one instance of SIP server with 500 000 online users (mixed users ? >>>>> behind and not NAT routers) ? consumed energy 210W >>>>> - one instance of SIP server with 1 000 000 online users (no NAT >>>>> involved) ? consumed energy 190W >>>>> - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for >>>>> SIP server, the server could support 43 000 simultaneous TLS >>>>> connections ? consumed energy 203W >>>>> - one SIP server instance with 80 000 permanent TCP connections, the >>>>> SIP server could still handle at least 1000 requests per second and a >>>>> connection arrival rate of 1000 new connections per second, done for >>>>> 20 000 new connections. CPU load generated by the SIP server was from >>>>> 6% to 8%. >>>>> >>>>> I added a new section to the draft notes to list the enhancements done >>>>> for the latest stable release (v3.1.x) that contribute to performance >>>>> improvements, like asynchronous TLS, fine tuning of memory for TLS >>>>> connections and raw UDP sockets. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > > > >------------------------------ > >Message: 3 >Date: Thu, 26 May 2011 21:00:42 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >Subject: [SR-Users] Kamailio v3.1.4 Released >To: kamailio <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org >Message-ID: <4DDEA35A.7070606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >Kamailio SIP Server v3.1.4 stable release is out. > >This is a maintenance release of latest stable branch, 3.1, that >includes fixes since release of v3.1.3. There is no change to database >schema or configuration language structure. Deployments running previous >v3.1.x versions are strongly recommended to be upgraded to v3.1.4. > >For more details about version 3.1.4, visit: > >http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ > >Cheers, >Daniel > >-- >Daniel-Constantin Mierla -- http://www.asipto.com >http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >------------------------------ > >Message: 4 >Date: Thu, 26 May 2011 17:59:42 -0300 >From: caio <elcaio(a)gmail.com> >Subject: Re: [SR-Users] ser_ctl / serweb >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTik+dnTp-egK2xy1ua=tPnCcaDuuOA(a)mail.gmail.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Anyone know if ser_ctl (python utility) will be merged into master? Or is >going to be deprecated? >If it still alive, are there any examples/docs of the usage for DB users >provisoining? >kamctl and kamdbctl are recomended for its substitution? > >Which utility will be official for sip-router? >Sorry if it already were discussed, but I didn't find comments about it >since 2010. > >Thank you >Claudio > >On Tue, May 24, 2011 at 11:14 AM, Claudio Furrer <elcaio(a)gmail.com> wrote: > >> Hi all, >> >> Just want to ask if ser_ctl application will be merged into sr3.1 or future >> releases. >> By the moment I've found it here [1], [2] or [3]. >> >> Is it full compatible with sip-router v3.x? >> >> I know kamctl and kamdbctl would be suggested but these are for kamailio >> flavour. I don't find too much work based on SER flavour regarding to >> database/users administration and web provisioning. >> BTW, siremis only works with kamailio db structures, but for sip-router v3 >> ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if >> is >> it working well with the current new releases of the project. >> >> I would appreciate if someone can advice me if kamailio is the way to go, >> because have find that sip-router or ser flavours miss some docs and apps >> that >> kamailio already have. >> >> I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( >> >> [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ >> [2] >> >> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… >> [3] http://cvs.berlios.de/viewvc/ser/serctl/ >> >> [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ >> [5] http://developer.berlios.de/projects/serweb/ >> >> Thank you, >> Claudio >> >

1 0

Kamailio v3.1.4 Released
by Daniel-Constantin Mierla 26 May '11

26 May '11

Hello, Kamailio SIP Server v3.1.4 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.3. There is no change to database schema or configuration language structure. Deployments running previous v3.1.x versions are strongly recommended to be upgraded to v3.1.4. For more details about version 3.1.4, visit: http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Aliases Using the Default Script;
by David J. 26 May '11

26 May '11

I am using the kamailio default script on 3.1.3. I was wondering what happens when I added an Alias in dbaliases? For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com when an invite comes in; it works perfect I got a 200 back. (1001 Device rings.) If I add another alias to a remote server; ie. 18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com I get a 404 back. I wonder what happens internally? What should I modify to enable this case; Thanks.

1 0

preparing v3.1.4
by Daniel-Constantin Mierla 26 May '11

26 May '11

Later today I will package v3.1.4, so if you have something to commit in branch 3.1, please do it before 15:00GMT. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

packaging 3.1.4
by Juha Heinanen 26 May '11

26 May '11

Daniel-Constantin Mierla writes: > I am thinking of packaging v3.1.4 next week on Thursday -- last minor > release was more than 1.5month ago. > Any comments against it? the bug still exists, where AVP_CLASS_DOMAIN avps are not saved in the transaction. i guess jan forgot about it. -- juha

2 1

"ACK not received"
by Mokhtar Bengana 26 May '11

26 May '11

I am trying to forward a DID to a sip account registered to Kamailio. The call get established with two way audio and Kamailio shows it's comming from an Asterisk server. However the call get disconnected each time after 31s. Ngrep is showing "ACK not received" by the Kamailio test account before the call get disconnected. If Kamailio is suppose to send an ACK back to the asterisk server to keep the call up, I am not sure why it's not happening. Or is it the other way around? I have included the ngrep output. I appreciate any help. U 2011/05/26 00:27:12.525344 91.121.81.212:5060 -> 192.168.2.4:5060 INVITE sip:test@sipsoft.dyndns.org SIP/2.0. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. From: "3313144859493" <sip:3313144859493@188.165.98.47>;tag=as6c63c119. To: <sip:33980307151@sip.antisip.com:5060>. Contact: <sip:3313144859493@188.165.98.47:5060>. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 69. Date: Thu, 26 May 2011 00:27:10 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Type: application/sdp. Content-Length: 260. P-Antisip: forwarded with alias. P-PMR: mediarelay enabled. . v=0. o=root 10502 10502 IN IP4 91.121.81.212. s=session. c=IN IP4 91.121.81.212. t=0 0. m=audio 41316 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2011/05/26 00:27:12.526734 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. From: "3313144859493" <sip:3313144859493@188.165.98.47>;tag=as6c63c119. To: <sip:33980307151@sip.antisip.com:5060>. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Server: kamailio (3.1.3 (x86_64/linux)). Content-Length: 0. . U 2011/05/26 00:27:12.527013 192.168.2.4:5060 -> 192.168.2.8:65412 INVITE sip:test@192.168.2.8:65412;rinstance=befda46036a026ac SIP/2.0. Record-Route: <sip:192.168.2.4;lr=on>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. From: "3313144859493" <sip:3313144859493@188.165.98.47>;tag=as6c63c119. To: <sip:33980307151@sip.antisip.com:5060>. Contact: <sip:3313144859493@188.165.98.47:5060>. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 68. Date: Thu, 26 May 2011 00:27:10 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Type: application/sdp. Content-Length: 260. P-Antisip: forwarded with alias. P-PMR: mediarelay enabled. . v=0. o=root 10502 10502 IN IP4 91.121.81.212. s=session. c=IN IP4 91.121.81.212. t=0 0. m=audio 41316 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2011/05/26 00:27:12.601815 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 0. . U 2011/05/26 00:27:12.601967 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 0. . U 2011/05/26 00:27:22.568944 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:22.569043 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:23.080398 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:23.080549 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:24.080628 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:24.080697 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:26.080654 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:26.080816 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:30.080527 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:30.080594 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:34.080318 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:34.080436 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:37.137775 192.168.2.8:65412 -> 192.168.2.4:5060 . . U 2011/05/26 00:27:38.081552 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:38.081707 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:42.082163 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:42.082308 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:46.080414 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:46.080562 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:50.082011 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:50.082142 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:54.080488 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:54.080592 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:54.574347 192.168.2.8:65412 -> 192.168.2.4:5060 BYE sip:3313144859493@188.165.98.47:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport. Max-Forwards: 70. Route: <sip:192.168.2.4;lr>. Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Reason: SIP;description="ACK not received". Content-Length: 0. . U 2011/05/26 00:27:54.574553 192.168.2.4:5060 -> 91.121.81.212:5060 BYE sip:3313144859493@188.165.98.47:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK79bb.32809f01.0. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport=65412. Max-Forwards: 69. Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Reason: SIP;description="ACK not received". Content-Length: 0. . U 2011/05/26 00:27:54.705228 91.121.81.212:5060 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;rport=5060;received=99.42.204.9;branch=z9hG4bK79bb.32809f01.0. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport=65412. Record-Route: <sip:91.121.81.212;lr>. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. . U 2011/05/26 00:27:54.705377 192.168.2.4:5060 -> 192.168.2.8:65412 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport=65412. Record-Route: <sip:91.121.81.212;lr>. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. . U 2011/05/26 00:28:07.139142 192.168.2.8:65412 -> 192.168.2.4:5060 . .

3 3

Wildcard TLS
by Jon Farmer 25 May '11

25 May '11

Hi I am just wondering if it is possible to use a wildcard certificate with TLS. I have hundreds of customers who are using subdomains of the main domain name on a virtual PBX platform. i.e if the main domain is example.org I have customers using cust1.example.org cust2.example.org cust3.example.org ....... I want to start offering TLS on a per customer basis. However I don't want to have to create a TLS certificate per customer if I don't need to. So I need to know if I can create a *.example.org certificate and use that for all customers who want TLS? Is this possible? Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 4

3.1.x and Kamailio compatibility mode
by Klaus Darilion 25 May '11

25 May '11

Hi! Is #!KAMAILIO still necessary with Kamailio 3.1.x? If yes, which behavior is changed by #!KAMAILIO? Thanks Klaus

3 8

Service Execution Time
by Giuseppe Carella 25 May '11

25 May '11

Hi all, I'm using kamailio with the dispatcher module. I would like to know if there are some API to get an average of the service execution time. In other words, I need to get the time between sending the request and receiving the response for every request, but also to have this values separate per server. I know also that this could be a little heavy for the dispatcher, but for some test I need this information. Your help is kindly appreciated! Thanks in advance, Giuseppe

2 1

Manipulate From header
by alex pappas 25 May '11

25 May '11

Dear all, I'm trying to change the From uri and Dsplay but without sucess. My config is as follow: if(dp_translate("1111", "$avp(s:frm_user_name)/$avp(s:test_frm_user_name)")) --> i'm sending 2112202701 and I get back corectly 701 { $avp(s:display) = $avp(s:test_frm_user_name); $avp(s:fu_uri) = "sip:" + $avp(s:test_frm_user_name) + "@" + $fd; xlog("alx ------- The avp(s:display): $avp(s:display) ---- avp(s:fu_uri)=$avp(s:fu_uri) ----------\n"); --> I see values 701 and sip:701@my_IP_address uac_replace_from("$avp(s:display)","$avp(s:fu_uri)"); Here I log the fu and I see that nothing has changed.. } Do you have any ideas on that ? I use Kamailio 3.1.3 Thank you in advance Alex

2 2

undefined symbol error
by Bruno Bresciani 24 May '11

24 May '11

Hi, Someone know tell me why kamailio generate the following error when I try load the route.so module: May 24 10:21:43 sswpst00 kamailio: ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </home2/local/kamailio/lib/kamailio/modules/route.so>: /home2/local/kamailio/lib/kamailio/modules/route.so: undefined symbol: db_use_table I don't know why this error have been generate. I'm porting this module to use with kamailio 3.1.2.

2 3

htable ::size has wrong value
by Juha Heinanen 24 May '11

24 May '11

i have entries in htable with key name lpi::20033::prefix: # sip-proxy_ctl mi sht_dump htable | egrep 20033 lpi::20033::prefix[0]:: +3583796294 lpi::20033::prefix[1]:: +35837824030 lpi::20033::prefix::size:: 1 as you see in above, size of that key name is 1 although there clearly is two entries in the table with that key name. according to readme ::size is number of items: 3.7. array_size_suffix (str) The suffix to be added to store the number of items in an array. i verified with another key name where there was only one prefix, and then ::size was again 1. so sometimes ::size is correct and sometimes incorrect. this looks like a serious bug to me. any comments? -- juha

2 4

Telefonia ip
by rom ignacio 24 May '11

24 May '11

friend I am interested in learning more about VoIP or IP telephony, I read a lot about you and I still have more to read, as it can not released through theofficial documentation page errors Please let me send them am email, thanks I DEVELOP voip server to sip in Venezuela, I need all the help possible. thanks *Rommel Malave .M ** * *T.S.U Universitario Extracción y Producción de Gas * * T.S.U en Informatica*** 04148241688

1 0

Re: [SR-Users] app_lua in kamailio
by Daniel-Constantin Mierla 24 May '11

24 May '11

Hello, please do not write direct private emails for questions about kamailio, use public mailing list sr-users(a)lists.sip-router.org (cc-ed) -- private emails are ignored, after the first notification. exit is not a function exported by a module, it is implemented in core. On the other hand, you try to stop the kamailio cfg interpreter when lua interpreter is executing the Lua script, so it won't work straightforward. Set a variable in kamailio from Lua, then check it after executing the lua script to decide to further exit or not. Cheers, Daniel On 5/24/11 11:44 AM, Bret McDanel wrote: > I am having a terrible time trying to get a conditional response from a > lua script. > > Basically I want > > if error then > sr.sl.send_reply(404,"Not Found"); > sr.modf("exit"); -- this does not work! > end > > > How can I have such a conditional? I have tried with return values from > the script to see if that would change anything but that did not seem to > work, sr.modf("exit") is not found. > > Any assistance would be greatly appreciated. > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 3

struct run_act_ctx
by Bruno Bresciani 24 May '11

24 May '11

Hi, I'd like to know the functionality of the struct run_act_ctx. I need to adapt my route module to use this struct, the function do_action require this parameter. I'm migrating of the kamailio 1.5.0 to 3.1.2. Regards,

2 2

Re: [SR-Users] 403 Proxy forbidden replay for invite that send sip gateway
by hala alramli 24 May '11

24 May '11

thanks for your reply Klaus ,attached my config file could you please take a look ,i found the line that send that replay but i do not know why .? could you help me i am stuck --- Mer 18/5/11, Klaus Darilion <klaus.mailinglists(a)pernau.at> ha scritto: Da: Klaus Darilion <klaus.mailinglists(a)pernau.at> Oggetto: Re: [SR-Users] 403 Proxy forbidden replay for invite that send sip gateway A: sr-users(a)lists.sip-router.org Data: Mercoledì 18 maggio 2011, 13:04 Check your config and verify where 403 is sent. You have something like sl_send_reply("403","Proxy forbidden") in your config. regards Klaus Am 18.05.2011 10:48, schrieb hala alramli: > Hello every body, > could you help me ? > > i bought Geographical DID numbers from gateway provider , to receive > calls from pstn > > my problem that when gateway send invite my proxy send 403 forbidden ,i > dont know why what is wrong ,any help please. > > this is a track for th gateway invite. > > my proxy ip:79.12.254.153 > gateway IP :77.239.128.10 > my geographical number :0698962030 > > > > > > INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: <sip:0698962030@10.39.6.207:5060;user=phone> > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 INVITE > > User-Agent: CS2000_NGSS/8.0 > > Remote-Party-ID: <sip:0289089500@10.39.1.16;user=phone>; party=calling; > privacy=off; screen=yes > > Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK > > P-Asserted-Identity: <sip:0289089500@10.39.1.16;user=phone> > > Max-Forwards: 69 > > Contact: <sip:77.239.128.10:5060;transport=UDP> > > Supported: 100rel > > Content-Type: application/sdp > > Content-Length: 236 > > > > v=0 > > o=PVG 1305215180830 1305215180830 IN IP4 77.239.128.10 > > s=- > > p=+1 6135555555 > > c=IN IP4 77.239.128.10 > > t=0 0 > > m=audio 49264 RTP/AVP 18 8 0 13 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > a=fmtp:18 annexb=no > > > > > > > > SIP/2.0 403 Proxy forbidden > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 INVITE > > Server: Kamailio (1.4.4-notls (x86_64/linux)) > > Content-Length: 0 > > > > > > > > > > > > > > > > > > > > > > ACK sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 ACK > > Max-Forwards: 69 > > Content-Length: 0 > > > > > > > INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: <sip:0698962030@10.39.6.207:5060;user=phone> > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 INVITE > > User-Agent: CS2000_NGSS/8.0 > > Remote-Party-ID: <sip:026683011@10.39.1.16;user=phone>; party=calling; > privacy=off; screen=yes > > Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK > > P-Asserted-Identity: <sip:026683011@10.39.1.16;user=phone> > > Max-Forwards: 69 > > Contact: <sip:77.239.128.10:5060;transport=UDP> > > Supported: 100rel > > Content-Type: application/sdp > > Content-Length: 236 > > > > v=0 > > o=PVG 1305215458700 1305215458700 IN IP4 77.239.128.10 > > s=- > > p=+1 6135555555 > > c=IN IP4 77.239.128.10 > > t=0 0 > > m=audio 41892 RTP/AVP 18 8 0 13 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > a=fmtp:18 annexb=no > > > > > > > > SIP/2.0 403 Proxy forbidden > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 INVITE > > Server: Kamailio (1.4.4-notls (x86_64/linux)) > > Content-Length: 0 > > > > > > > > > > > > > > > > > > > > > > > > sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 ACK > > Max-Forwards: 69 > > Content-Length: 0 > > > > > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 4

SEMS 1.4.0 release
by Stefan Sayer 24 May '11

24 May '11

We are happy to announce the availability of the SIP Express Media Server version 1.4.0. SEMS is a high performance media and application server for SIP based VoIP networks. This release features a powerful Session Border Controller (SBC) module. From completely transparent B2BUA to customized URI/From/To, strictly filtered (messages, headers, codecs) with RTP anchoring, Session Timer enforcement, prepaid and call timer, the SBC facilitates interconnect and core routing in a simple and secure way. Thanks to the new multihoming support, SEMS can now be employed at the border of the networks. This addition also allows to overcome the bottleneck of one NIC - giving the possibility to fully exploit SEMS' great performance. In the app development area, the DSM language has matured to become a viable candidate also for implementing complex application logic, thanks to language constructs like for, if and functions. SEMS can be downloaded in source from its ftp site at: ftp.iptel.org/pub/sems/1.4/1.4.0/ Packages for debian/ubuntu should appear shortly on the OBS page: https://build.opensuse.org/project/show?project=home:team-sems Contributors to this release: Stefan Sayer, Raphael Coeffic, Bogdan Pintea, Matthew Williams, Peter Lemenkov, Anton Zagorskiy, Robert Szokovacs, Jon Bonilla, Thomas Gelf, Erik Dekkers, Tobias Lindgren, David J Many thanks to the companies sponsoring SEMS development, especially FRAFOS GmbH and TelTech Systems Inc. "E stands for Express." * Apologies if you receive multiple copies of this message. *

4 5

~ operator has no effect
by Juha Heinanen 24 May '11

24 May '11

according to core cookbook: ~ : bitwise NOT however, looks like it has no effect. i have $var(test) = ~2; xlog("L_INFO", "test is <$var(test)>\n"); and i get to syslog: May 17 18:10:40 sip /usr/sbin/sip-proxy[29280]: INFO: test is <2> have i misunderstood the operator or is there a bug? -- juha

3 5

Re: [SR-Users] IPv6
by Daniel-Constantin Mierla 24 May '11

24 May '11

On 5/20/11 6:33 PM, Jon Farmer wrote: > On 20 May 2011 17:25, Daniel-Constantin Mierla<miconda(a)gmail.com> wrote: > >> the module is good, but not for this case -- the warning seems to be from >> startup, saying that no reverse dns could be done to the ipv6 listen address >> and it is harmless (done for the purpose of building aliases list). >> >> Do you get the warning at runtime? >> >> Normally, the ipv6 should be in between square brackets. >> > Adding the address in question to my hosts file temporarily takes the > warning away :) ok, let me know how is going with the testing of ipv6 -- the support is in core since 2002, but not many entered the challenge so far :-) Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 4

newbie INVITE routing logic help
by Skyler 24 May '11

24 May '11

Hi all, Need some newbie advice on how to write INVITE routing for my scenario. I currently have kamailio+asterisk-realtime integration working in the following way: Outbound calls: USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then Asterisk(1.6.2.18) -> SIP/PSTN This works, though I'd prefer to use Kamailio + LCR and gateway failover as Asterisk doesn't do this well. Here's what I want to do: (for now, small steps) Outbound Calls: USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then Asterisk(1.6.2.18) -> Kamailio(3.1.3)+(lcr) -> SIP/PSTN I know I have to set kamailio as a GW in asterisk instead of the PSTN GW's and add the PSTN GW's to kamailio instead, though I'm at a loss as to how to write the routing logic so that kamailio sends calls from asterisk to PSTN. route[LOCATION] { ...non-INVITE check ends here... if(route(FROMASTERISK)) { # coming from Asterisk --- Does this mean check kamailio local users first before continuing? if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } ## -- If match 11 digits and starts with '1' or if match 10 digits and starts with [2-9] if($rU=~"^(1\d{10})$") || if($rU=~"^([2-9]\d{9})$") { xlog("L_INFO", " FROMASTERISK: This will become route(NORTHAMERICA); \n"); exit; ## to be removed } else { xlog("L_INFO", " FROMASTERISK: This will become route(INTERNATIONAL); \n"); exit; ## to be removed } # new call - send to Asterisk route(TOASTERISK); } Will this logic work? I haven't determined how to write route(NORTHAMERICA) or route(INTERNATIONAL) yet so I'm just logging to debug the logic and syntax. Thanks in advance.

1 0

Kamailio to handle call states
by Yeris Antonio Madrigal Castro 23 May '11

23 May '11

Hi I am planing to change my old T1-E1 pbx, wich is AVAYA, for an asterisk cluster. I am planning this for a call center and I spec 80 currents calls for 30 agents sometimes, so I need to handle big queues and call and agent states, is it possible for kamailio handle the call states and agent states, and leave asterisk for the call center functions? Thanks in advance

1 0

TLS - Kamailio Asterisk - Registration issue
by Lucas Alvarez 23 May '11

23 May '11

Hi, I have confgured Kamailio with TLS in an Kamailio-Asterisk implementation. The phones (Bria - Eyebeam - Aastra 57i) are registering in Kamailio without problems, but the registration is not being forwarded to Asterisk. Kamailio is listening on port tls:5061 only and Asterisk on udp:5080. Should Kamailio has to listen on port 5060 also for UAC module functions? If I enable the port 5060, how do I prevent common SIP - UDP registrations? I mean, I want the registrations to be over TLS only. I would appreciate if someone can put me on the right path. Thanks in advance. Lucas Alvarez

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change ptime problem
by hala alramli 22 May '11

22 May '11

i am trying to route to pstn provider,but i faced a problem . my provider accept ptime 20 while i am send the call with ptime 30 how i can change ptime i have kamailio with freeswitch any help please

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