- sr-users - kamailio.org

[Kamailio-Users] measuring calls per second
by Juan Perez 24 Mar '09

24 Mar '09

Hello What is the easiest way to know how many calls per second a proxy is handling at a given time? I looked thru the SNMPSTATS and STATISTICS modules w/o luck. cheers jp

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[Serusers] SER+Freeradius+CDRTool: rlm_sql problem Unsupported Acct-Status-Type Failed
by Ilya Pichugin 24 Mar '09

24 Mar '09

Hi, I installed openSER 1.3, Freeradius 1.1.6, CDRTool v6.7.2 Now I can see in my CDRtool web-interface successfully calls (OK 200), but I want to see unsuccessfully calls too (like Cancel, etc) In radius RFC this type of calls binded by VALUE Acct-Status-Type Failed 15 # RFC2866, acc So, in radius debug I see messages like: raddact: Mon Mar 23 09:14:51 2009 Acct-Status-Type = Failed Service-Type = Sip-Session Sip-Response-Code = 487 Sip-Method = Invite Event-Timestamp = "Mar 23 2009 09:14:51 MSK" Sip-From-Tag = "f758dc31" Sip-To-Tag = "5affff00ff10ff10ff00003048ffff26" Acct-Session-Id = "OTQxMDJmOGE4ZjQ0MjVlNTJlMjE0ZGNmYjg1YjlmZmE." Acct-Session-Id = "OTQxMDJmOGE4ZjQ0MjVlNTJlMjE0ZGNmYjg1YjlmZmE." Calling-Station-Id = "sip:33640000@mysipserver.com" Sip-Translated-Request-URI = "sip:2984545@xx.xxx.xxx.xx:5060" User-Name = "3360000" Media-Codecs = "call|3360000|mysipserver.com|<null>" Media-Info = "2984545|xxx.xx.xxx.xx|<null>" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Timestamp = 1237788891 and radius.log: Sat Mar 21 20:39:18 2009 : Info: rlm_sql (sql): Unsupported Acct-Status-Type = 15 I tried to install Freeradius-2.1.4 and I get same thing: Mon Mar 23 11:17:29 2009 : Error: rlm_radutmp: NAS localhost port 5060 unknown packet type 15) But! I see success processing Cancel-calls at AG-projects in screen like http://cdrtool.ag-projects.com/attachment/wiki/WikiStart/CDRTool-CallSearch… So, how may I fix this trouble? Thanks a lot. -- Best Regards, Ilya Pichugin

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Re: [Kamailio-Users] Siremis 0.9.2 with Kamailio 1.5 on Debian 5.0
by mike＠agcusa.com 23 Mar '09

23 Mar '09

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[Kamailio-Users] [LCR] About ping
by Iñaki Baz Castillo 22 Mar '09

22 Mar '09

Hi, some questions about LCR module ping mechanism: - How to know which gws are down? The only I see is a NOTICE log: --------- NOTICE:lcr:gw_set_state: trunk "99.99.99.99:5060" from group: <2> is OFFLINE --------- Is not possible a MI command to list the down gateways? - Why ping_interval cannot be less than 180 seconds? - In case of failure_route and "next_gw()", is the used gw (failing gw) automatically marked as down? (it would be useful so we don't need to wait "fr_timer" seconds for each request during "ping_interval"). Thanks. -- Iñaki Baz Castillo

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[Kamailio-Users] ACK when using NAT
by michel freiha 20 Mar '09

20 Mar '09

Dear All, Hi all, i'm trying to send FAX to an endpoint behind a NAT...The FAX is failing because the ACK message is sent to Private endpoint IP instead of PUBLIC IP....Can you please let me know what's wrong?Please find below my ACK method that i define in loose route methode f ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("AAAAKKKCCCCCCCNNNNNN"); force_rport(); t_relay(); exit; } Regards

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[Serusers] (no subject)
by Bill McNamara 20 Mar '09

20 Mar '09

We are having something of an issue with lookup_user() and lookup_contacts(). Our UA's are added to the proxy using "ser_ctl user add 1234567890(a)ourdomain.com -p password" which works great And the UA's register just fine. When a call comes in from a carrier, the invite is for 1234567890(a)12.34.56.78 where 12.34.56.78 is the IP address of our proxy. When we do a lookup to see if the call is destined for a local user via lookup_user("$tu.uid","@ruri") it fails because the registration is done using the URL and not the IP address. If I add the same user for the IP address domain, "ser_ctl user add 1234567890(a)12.34.56.78 -p password", the lookup_user() works, but the lookup_contacts(0 never does. I have tried adding aliases in the ser.cfg file: alias=12.34.56.78 alias=proxy.ourdomain.com but it does not work. I'm new to SER and the documentation I have been able to locate does not really address this issue, if, in fact, it really is an issue. If anyone could point me to something that will help me work through this problem, I'd greatly appreciate it. Thanks, Bill Bill McNamara billmc(a)connectmevoice.com ConnectMeVoice LLC 800-743-1208 x104

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[Kamailio-Users] [LCR] Can "prefix" be "empty" (any)?
by Iñaki Baz Castillo 20 Mar '09

20 Mar '09

Hi, I read in LCR doc: --------- each gateway group is associated with one or more <prefix, from pattern, priority> tuples. A gateway matches a request if user part of Request URI matches a prefix and caller's URI matches a from pattern in a tuple that belongs to the group of the gateway. ---------- Can "prefix" be empty (or NULL?) so any RURI matches it? Thanks. -- Iñaki Baz Castillo

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[Kamailio-Users] [LCR] Can I use a hostname in "ip_addr" field of "gw" table?
by Iñaki Baz Castillo 20 Mar '09

20 Mar '09

Hi, instead of an IP, can I use a hostname in "ip_addr" field of "gw" table? The hostname would have no DNS record (it would appear just in local /etc/hosts). Is it possible? or perhaps a real IP is required? Thanks. -- Iñaki Baz Castillo

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[Serusers] Deadly embrace with rtpproxy - Is it necessary? Can it be turned off?
by Frank Durda IV 20 Mar '09

20 Mar '09

I'm running into a problem with rtpproxy on this point, quoting from the README: - - - - - - - - - - - - after the session has been created, the proxy listens on the port it has allocated for that session and waits for receiving at least one UDP packet from each of two parties participating in the call. Once such packet is received, the proxy fills one of two ip:port structures associated with each call with source ip:port of that packet. When both structures are filled in, the proxy starts relaying UDP packets between parties; - - - - - - - - - - - However, a number of clients frequently fail to emit any audio when originating a call until they hear something from the TDM gateway, such as ring-back or the called party answering. So although rtpproxy is receiving a stream of audio, such as a voice mail menu robot, the calling party can't hear any of it unless they happen to make some noise or randomly and blindly press a DTMF key. This seems to be made worse on links with silence suppression, so there is no background noise to trigger two-way audio. This is being encountered between Class 4 carriers, so we don't have the option to get someone to adjust their phone/PBX settings or have them breathe heavier. Is there a setting adjustment to get rtpproxy to just pass the RTP packets from directed calling and called sources even if one party hasn't happened to make noise yet? I personally don't understand why this requirement for seeing audio from both sides before starting the flow in either direction if audio starts coming in even exists. It seems to have no benefit but is bound to cause this deadly embrace problem in many situations that may be beyond the control of the owners of the equipment passing traffic along to the site where rtpproxy is in use. Suggestions? Fix? I have looked at the latest snapshot of rtpproxy and the README is unchanged since 1.1 so apparently this behavior is still the same. Thanks in advance!

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Re: [Kamailio-Users] Problem starting Kamailio
by Sara EL KABIRI 20 Mar '09

20 Mar '09

On Thu, Mar 19, 2009 at 12:18 PM, Henning Westerholt < henning.westerholt(a)1und1.de> wrote: > > Hi Sara, > > ok, take a look to the plain output from strace, without the grep command. > Does it shows any errors from a file open, or a stat? Probably stupid > question, but do you've the strace command installed? > > Cheers, > > Henning Hello Henning, I do have strace :-) Yesterday i tried it with and without the grep part, and there was no error. So i did some IMS presence service tests... And Kamailio couldn't process a "SIP PUBLISH" because of an error "*ERROR 600 : Busy everywhere - Forwarding to S-CSCF failed*". For your information, i'm using the Fokus OpenIms Core as an ims server, and Kamailio as a presence server. An IFC (Initial Filter Criteria) is set up in OpenIms to route properly the presence messages to Kamailio. And by the way, i saw some lines in the logs about /tmp/kamailio_fifo opening successfully... Today, i launch the strace command and gess what... i'm getting the /tmp/kamailio_fifo opening error. Still a new kamailio_fifo is created indeed in my /tmp directory. And well, i am encountering the "busy everywhere" error again... Have anyone ever noticed the error 600? And what about the mi_fifo error that appears half the time? PS: Here you are some parts of the logs, and you can have a look at the attached file for the entire trace. ====LOG PARTS==== 1/ Error 600 : busy everywhere (in kamailio start) *Mar 19 10:46:28 [11850] DBG:core:parse_msg: SIP Request: Mar 19 10:46:28 [11850] DBG:core:parse_msg: method: <PUBLISH> Mar 19 10:46:28 [11850] DBG:core:parse_msg: uri: <sip:bob@test.ims.com<sip%3Abob(a)test.ims.com> > Mar 19 10:46:28 [11850] DBG:core:parse_msg: version: <SIP/2.0> Mar 19 10:46:28 [11850] DBG:core:parse_headers: flags=2 Mar 19 10:46:28 [11850] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKa813.b002d4e4.0>; state=16 Mar 19 10:46:28 [11850] DBG:core:parse_via: end of header reached, state=5 Mar 19 10:46:28 [11850] DBG:core:parse_headers: via found, flags=2 Mar 19 10:46:28 [11850] DBG:core:parse_headers: this is the first via Mar 19 10:46:28 [11850] DBG:core:receive_msg: After parse_msg... Mar 19 10:46:28 [11850] DBG:core:receive_msg: preparing to run routing scripts... Mar 19 10:46:28 [11850] DBG:core:parse_headers: flags=100 Mar 19 10:46:28 [11850] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKa813.41769183.0>; state=16 Mar 19 10:46:28 [11850] DBG:core:parse_via: end of header reached, state=5 Mar 19 10:46:28 [11850] DBG:core:parse_headers: via found, flags=100 Mar 19 10:46:28 [11850] DBG:core:parse_headers: parse_headers: this is the second via Mar 19 10:46:28 [11850] DBG:core:parse_via_param: found param type 235, <rport> = <5060>; state=6 Mar 19 10:46:28 [11850] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK315224847>; state=16 Mar 19 10:46:28 [11850] DBG:core:parse_via: end of header reached, state=5 Mar 19 10:46:28 [11850] DBG:core:parse_headers: via found, flags=100 Mar 19 10:46:28 [11850] DBG:core:parse_to: end of header reached, state=10 Mar 19 10:46:28 [11850] DBG:core:parse_to: display={}, ruri={ sip:bob@test.ims.com <sip%3Abob(a)test.ims.com>} Mar 19 10:46:28 [11850] DBG:core:get_hdr_field: <To> [25]; uri=[** sip:bob@test.ims.com <sip%3Abob(a)test.ims.com>**] Mar 19 10:46:28 [11850] DBG:core:get_hdr_field: to body [<** sip:bob@test.ims.com <sip%3Abob(a)test.ims.com>**> ] Mar 19 10:46:28 [11850] DBG:core:get_hdr_field: cseq <CSeq>: <20> <PUBLISH> Mar 19 10:46:28 [11850] DBG:maxfwd:is_maxfwd_present: value = 15 Mar 19 10:46:28 [11850] DBG:uri:has_totag: no totag Mar 19 10:46:28 [11850] DBG:core:parse_headers: flags=78 Mar 19 10:46:28 [11850] DBG:tm:t_lookup_request: start searching: hash=12682, isACK=0 Mar 19 10:46:28 [11850] DBG:tm:matching_3261: RFC3261 transaction matched, tid=a813.b002d4e4.0 Mar 19 10:46:28 [11850] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Mar 19 10:46:28 [11850] DBG:tm:t_lookup_request: transaction found (T=0xb5a0d430) Mar 19 10:46:28 [11850] DBG:tm:t_retransmit_reply: nothing to retransmit Mar 19 10:46:28 [11850] DBG:tm:t_check_trans: UNREF_UNSAFE: after is 0 Mar 19 10:46:28 [11850] DBG:core:destroy_avp_list: destroying list (nil) Mar 19 10:46:28 [11850] DBG:core:receive_msg: cleaning up Mar 19 10:46:31 [11852] DBG:core:parse_msg: SIP Reply (status): Mar 19 10:46:31 [11852] DBG:core:parse_msg: version: <SIP/2.0> Mar 19 10:46:31 [11852] DBG:core:parse_msg: status: <600> Mar 19 10:46:31 [11852] DBG:core:parse_msg: reason: <Busy everywhere - Forwarding to S-CSCF failed> Mar 19 10:46:31 [11852] DBG:core:parse_headers: flags=2 Mar 19 10:46:31 [11852] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKa813.192cfef5.0>; state=16 Mar 19 10:46:31 [11852] DBG:core:parse_via: end of header reached, state=5* 2/ /tmp/kamailio_fifo opening error (with strace kamctl) *rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 stat64("/tmp/kamailio_fifo", {st_mode=S_IFIFO|0660, st_size=0, ...}) = 0 geteuid32() = 1000 getegid32() = 1000 getuid32() = 1000 getgid32() = 1000 access("/tmp/kamailio_fifo", W_OK) = -1 EACCES (Permission denied) rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 ioctl(1, SNDCTL_TMR_TIMEBASE or TCGETS, {B38400 opost isig icanon echo ...}) = 0 rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 write(1, "\33[37;31m\33[1mERROR: Error opening"..., 72ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ) = 72 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 ioctl(1, SNDCTL_TMR_TIMEBASE or TCGETS, {B38400 opost isig icanon echo ...}) = 0 rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 brk(0x9302000) = 0x9302000 write(1, "\33[37;31m\33[1mERROR: Make sure you"..., 123ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ) = 123 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 ioctl(1, SNDCTL_TMR_TIMEBASE or TCGETS, {B38400 opost isig icanon echo ...}) = 0 rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 write(1, "\33[37;31m\33[1mERROR: and also have"..., 64ERROR: and also have loaded the mi_fifo module. ) = 64 rt_sigprocmask(SIG_BLOCK, NULL, [], 8) = 0 rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 rt_sigprocmask(SIG_SETMASK, [], NULL, 8) = 0 exit_group(2) = ? Process 12215 detached* ================== -- S.

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Re: [Kamailio-Users] ERROR:error opening Openser's FIFO FIFO
by Henning Westerholt 20 Mar '09

20 Mar '09

On Friday 20 March 2009, Metshet Seleshi wrote: > Really > appreciate your replay, I am sorry, I wasn’t able to replay time sooner, > > Answers > to the questions > > I > am using OpensER-1.3.4-tls , > > Server > starts up fine, and register, > > And below is part of the config file, I hope it helps > [..] > loadmodule "mi_fifo.so" > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") Hi Metshet, the cfg looks good so far. Can you do a "ls -la /tmp/openser_fifo" and quote the output? With what user do you've tried to run the openserctl tool? Cheers, Henning

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Re: [Kamailio-Users] server isnt writing to "acc" table - how to debug?
by Daniel-Constantin Mierla 20 Mar '09

20 Mar '09

please keep the mailing list in the recipient list, so the others facing same problem learn the right solutions. Thanks, Daniel On 03/19/2009 06:14 PM, Joao Gomes Pereira wrote: > That's it :) > Thanks for the help > Joao Pereira > > Daniel-Constantin Mierla wrote: >> Hello, >> >> have you set db_url for acc module? >> >> modparam("acc", "db_url", "mysql://dbusername:dbpass@localhost/openser") >> >> Cheers, >> Daniel >> >> >> On 03/19/2009 05:55 PM, Joao Gomes Pereira wrote: >>> Hello to all >>> I enabled the Kamailio acc modules but it isn't writing to "acc" or >>> "missed_calls" table. >>> How can I see in real time the SQL querys that Kamailio is sending >>> to Mysqld? >>> Here are some parts of my configuration file. >>> >>> >>> (...) >>> loadmodule "db_mysql.so" >>> loadmodule "acc.so" >>> >>> (...) >>> >>> # ----- acc params ----- >>> /* what sepcial events should be accounted ? */ >>> modparam("acc", "early_media", 1) >>> modparam("acc", "report_ack", 1) >>> modparam("acc", "report_cancels", 1) >>> modparam("acc", "detect_direction", 0) >>> /* account triggers (flags) */ >>> modparam("acc", "failed_transaction_flag", 3) >>> modparam("acc", "log_flag", 1) >>> modparam("acc", "log_level", 2) >>> modparam("acc", "log_missed_flag", 2) >>> modparam("acc", "log_extra", >>> "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") >>> >>> /* uncomment the following lines to enable DB accounting also */ >>> modparam("acc", "db_flag", 1) >>> modparam("acc", "db_missed_flag", 2) >>> modparam("domain", "db_url", >>> "mysql://dbusername:dbpass@localhost/openser") >>> modparam("acc", >>> "db_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") >>> >>> >>> >>> Flags 1 and 2 are in the configuration (the same way as in the >>> sample that comes with Kamailio) >>> >>> (...) >>> >>> if (has_totag()) { >>> # sequential request withing a dialog should >>> # take the path determined by record-routing >>> if (loose_route()) { >>> if (is_method("BYE")) { >>> setflag(1); # do accounting ... >>> setflag(3); # ... even if the >>> transaction fails >>> } >>> route(1); >>> (...) >>> >>> # account only INVITEs >>> if (is_method("INVITE")) { >>> setflag(1); # do accounting >>> } >>> >>> So, if accounting is enabled and the flags in the right place... why >>> is the "acc" table still empty? >>> Thanks >>> Regards >>> Joao Pereira >>> >>> >> > > -- Daniel-Constantin Mierla SIP Router Masterclass - Kamailio (OpenSER) Training http://www.asipto.com/index.php/sip-router-masterclass/

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[Kamailio-Users] T38 NAT
by michel freiha 20 Mar '09

20 Mar '09

Dear All, I'm trying to send FAX using T38...Everything looks fine when using endpoints with real IP...As soon as I use endpoint behind NAT everything looks not working Any clue? Regards

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Re: [Kamailio-Users] Problem starting Kamailio
by Sara EL KABIRI 19 Mar '09

19 Mar '09

On Tue, Mar 17, 2009 at 1:35 PM, Henning Westerholt < henning.westerholt(a)1und1.de> wrote: > > Hi Sara, > > this is strange. If you increase the debug level to 4, you should see some > message about the module loading. Is the mi_fifo module correctly loaded? > Perhaps you configured another path in another modparam statement, so that > the first one is overwritten, and subsequently not found from the ctl > command? > > Cheers, > > Henning > Hello, The log i get after increasing the debug level to 4 is in the attached. Hope it's fine to attach files in this mailing list :-) -- S.

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[Kamailio-Users] point in the right direction
by Geoffrey Mina 19 Mar '09

19 Mar '09

hello, I am running Kamailio 1.5 and siremis. I want to deploy some IP based ACL services. Basically checking the source IP on an INVITE to see if it's in a list of trusted IP addresses. I see in the sirmeis configuration under "ACL Services" there is a tab for "trusted addresses", but I can't figure out what module those are in support of, or what exported functions I would call to check against the list of trusted IPs. I have had a high fever for 2 days, so perhaps my brain is fried and I am missing something totally obvious :) Thanks! Geoff

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[Kamailio-Users] Contact Header
by John 19 Mar '09

19 Mar '09

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[Kamailio-Users] Dentist.s Listing for the United States
by Earnest N Stinson 19 Mar '09

19 Mar '09

Top quality: **> 164,251 De'ntists **> 158,743 Addresses **> 163,412 Telephone Numbers **> 77,326 Office Fax Numbers **> 45,520 Email Addys from now until Friday the new lowered price is $299 Send an email to MiguelChampagne(a)mylistsource.com to inquire about this and other Database we have to adjust your subscription status email to takeoff(a)mylistsource.com

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[Kamailio-Users] Select a 3XX branch inmediatelly and CANCEL a ringing branch
by Iñaki Baz Castillo 18 Mar '09

18 Mar '09

Hi, let's me explain a simple escenario and the behaviour I want: Kamailio phone1 phone2-1 phone2-2 phone3 (201) (202) (202) (203) 1) 201 calls 202 so both phone2-1 and phone2-2 start ringing (180) in parallel. 2) phone2-1 replies a redirect "302" to user 203. 3) Kamailio cancels the branch with phone2-1 and replies 302 upstream *inmediatelly*. This is however not the most "compliant" behaviour, since Kamailio should wait for phone2-1 to reply a final response and then choose which one response to reply upstream. And that is of course the default behaviour in Kamailio. Is it possible however the behaviour I want? I find no way to get it. Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Hor to strip() $rU dynamically
by Iñaki Baz Castillo 18 Mar '09

18 Mar '09

Hi, I receive INVITE like this: INVITE sip:client_xxx2004@domain.org SIP/2.0 and I get from an $avp(CLIENT) the value "client_xxx". I need to strip from $rU the lenght of $avp(CLIENT), this is: # Original $rU => client_xxx2004 strip($avp(CLIENT)); # New $rU => 2004 Unfortunatelly "strip()" doesn't accept a PV as parameter, but a static integer. Any suggestion on how to achieve my goal? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] I don't want "rpid" column but get an ERROR in "auth_db"
by Iñaki Baz Castillo 18 Mar '09

18 Mar '09

Hi, I don't want/need the "rpid" column in "subscriber" table but if I delete it I get an error: ------- Mar 18 14:49:47 [22377] ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'rpid' in 'field list' Mar 18 14:49:47 [22377] ERROR:core:db_do_query: error while submitting query Mar 18 14:49:47 [22377] ERROR:auth_db:get_ha1: failed to query database ------- After readin "auth" module documentation I've found a parameter: ------------ rpid_avp (string) Full AVP specification for the AVP which stores the RPID value. It used to transport the RPID value from authentication backend modules (auth_db or auth_radius) or from script to the auth function append_rpid_hf and is_rpid_user_e164. If defined to NULL string, all RPID functions will fail at runtime. ------------ I expected that if I set: modparam("auth", "rpid_avp", "NULL") then "auth_db" module wouldn't require "rpid" column, but I must be wrong since I get an error: ------ ERROR:auth:init_rpid_avp: malformed or non AVP NULL AVP definition ------ So, is there any way to avoid "rpid" column usage? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] Using Asterisk as media server
by Mark Sayer 18 Mar '09

18 Mar '09

Pickup with OpenSER >> Asterisk 1.4.17 works just fine. Mark At 12:26 p.m. 18/03/2009, you wrote: >No, I don't need app_pickup... but thanks for the heads up. > >On Tue, Mar 17, 2009 at 8:21 PM, Iñaki Baz Castillo <ibc(a)aliax.net> wrote: > > El Miércoles, 18 de Marzo de 2009, Geoffrey Mina escribió: > >> I tend to not agree with some of what people are saying here. We have > >> deployed a Kamailio/Asterisk platform with GREAT success. Although, I > >> have the luxury of having many asterisk servers to achieve a common > >> goal. I have found asterisk to be VERY stable, > > > >> I am running 1.4.23. > > > > Then I hope you don't need Asterisk pickup code (since it doesn't work in > > 1.4.23 due to a regression). > > > > XD > > > > > > -- > > Iñaki Baz Castillo > > > > _______________________________________________ > > Kamailio (OpenSER) - Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > >_______________________________________________ >Kamailio (OpenSER) - Users mailing list >Users(a)lists.kamailio.org >http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] The hash_table_2 global variable is defined twice?
by catalina oancea 18 Mar '09

18 Mar '09

Hi, I have a question regarding the global hash_table_2 variable. It seems to me that it is defined twice in different modules: ./modules/permissions/trusted.c:47:struct trusted_list **hash_table_2; /* Pointer to hash table 2 */ ./modules/domain/domain_mod.c:85:struct domain_list **hash_table_2 = 0; /* Pointer to hash table 2 */ Could you please explain why this variable is defined twice?

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[Kamailio-Users] Dialog SNMP stats.
by Alex Balashov 18 Mar '09

18 Mar '09

Is it possible to monitor via SNMPStats the amount of dialogs currently open in a particular profile? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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Re: [Kamailio-Users] Using Asterisk as media server
by Juan Perez 18 Mar '09

18 Mar '09

Alex, I understand your point and appreciate your input highly, but let's say that asterisk is needed as the media server for other reasons. It may not scale as well as the other solutions but we need them to be there to provide other services. I just wanted to know whether there was any difference aside from the scalabilty issue. We were under the impression that it was imperative to use either media-proxy or rtp-proxy but after examining in more detail how the 2 solutions work we find no reason not to use asterisk as the media proxy as well. Unless of course we are missing the big picture here and surely will find out once we set out to test it. But given the deep knowledge of a lot of people in this forum we deemed necessary to ask about the feasibility of this workaround. If somebody detects any flaw on this solution or if there is someting else critical or not that we are missing we would love to hear it . thank you in advance and thank you Alex again, jp ----- Original Message ---- From: Alex Balashov <abalashov(a)evaristesys.com> To: Juan Perez <jperezsip2008(a)yahoo.com> Sent: Monday, March 16, 2009 6:51:34 PM Subject: Re: [Kamailio-Users] Using Asterisk as media server No, because the amount of throughput you get per media/RTP proxy is much higher than you would get through something whose entire core is structured to behave as a PBX, with all the attendant features and latent event loop characteristics. A media relay does one thing and one thing only, and it does it well. Asterisk does many things, among them media relay. So, while you could use Asterisk in this capacity, I don't see the point. It'd be a waste of time. Also, you don't really want to get Asterisk involved in the signaling layer (as a B2BUA) if you can help it. It introduces unnecessary complexity and possible points of interop problems and failure. With a media relay that is controlled by API hooks into Kamailio, you don't have that problem. On Mon, 16 Mar 2009 15:22:27 -0700 (PDT), Juan Perez <jperezsip2008(a)yahoo.com> wrote: > > thanks alex, but actually there will be several asterisk servers, managed > by the LCR module, that should be OK right? > it should be the same as adding several media-proxies or rtp-proxies, am I > wrong? > > > > > > > > ----- Original Message ---- > From: Alex Balashov <abalashov(a)evaristesys.com> > To: Juan Perez <jperezsip2008(a)yahoo.com> > Cc: users(a)lists.kamailio.org > Sent: Monday, March 16, 2009 3:41:46 PM > Subject: Re: [Kamailio-Users] Using Asterisk as media server > > No, nothing needs to be modified. > > However, in this scenario you are limited to Asterisk's performance > characteristics in terms of the number of concurrent calls that can be > handled. Just about any media relay will be much more scalable and > lightweight. > > Asterisk is a PBX and a nice feature server. You don't build SIP service > delivery platforms out of Asterisk used as a general-purpose call router. > > Juan Perez wrote: > >> Hello >> >> In this scenario, customers behind NAT register in kamailio, records are > stored in a db etc. Kamailio uses Asterisk boxes as gateways to send calls > to PSTN. >> Is it possible to use Asterisk as the media server instead of > media-proxy or rtp-proxy? >> In this case the SDP that Asterisk sends in the 180 message needs to me > modified? >> >> thank you >> jp >> >> >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > -- Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (678) 237-1775 -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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[Serusers] Record-Route issues
by Stephen Helfen 17 Mar '09

17 Mar '09

I have a 0.9.6 SER proxy running and one particular UA that is creating some problems with a RECORD-ROUTE header I am not handling correctly. Here is how I have my record_route configured (XXXs are replaced by my actual proxy IP): if (!(uri=~^"sip:XXX.XXX.XXX.XXX")) { record_route_preset("XXX.XXX.XXX.XXX:5060;nat=yes"); }; When inbound traffic arrives for this particular UA, the INVITE is passed along. when the 200 OK comes back, there is a Record-Route header that looks like this (YYYs are replaced with the UA's public IP): RECORD-ROUTE: <sip:YYY.YYY.YYY.YYY:5060;lr;transport=udp> Instead of adding a new Record-Route header with my IP before this one, the message is simply proxied back to the call origination. The ACK is then sent DIRECTLY to the UA, bipassing my proxy...as well as the BYE/200 OK that terminates the call....thus rendering my stateful proxy as stateless. Can I remove the RECORD-ROUTE that my terminating UA is attaching? If not, why am I not adding my own Record-Route header like normal? What am I missing? Thanks, Steve _________________________________________________________________ Windows Live™ Groups: Create an online spot for your favorite groups to meet. http://windowslive.com/online/groups?ocid=TXT_TAGLM_WL_groups_032009

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[Kamailio-Users] 487 Request Terminated and BYE
by Guillaume Lacroix 17 Mar '09

17 Mar '09

Hi, I am facing problem with handling CANCEL/487 Request Terminated in a case of a call hunting. The problem is that the call terminating carrier doesn't process the ACK properly and keeps sending 487 messages. Here is an example. Let's say UAC1 has the following rules of call hunting : 1. call +33123456 for 10s 2. call +33789456 for 10s Let's say that an incoming call to UAC1 triggers the call hunting. Here is a diagram : openSER -> Terminating Carrier (TC) : INVITE +33123456 ... (time out : +33123456 didn't answer in 10s, so openSER sends an INVITE to +33789456 - and let's say to an other Terminating carrier TC2 - and CANCEL the current INVITE to TC) openSER -> TC2 : INVITE : +33789456 openSER -> TC : CANCEL TC -> openSER : 487 Request Terminated openSER -> TC : ACK The problem is now TC doesn't process the ACK correctly and keeps sending 487. So, in the case of +33789456 answering the call (a 200 OK is sent to openSER), openSER will keep relaying the 487 to TC2 and TC2 will then send a BYE a terminate the call : TC -> openSER : 487 openSER -> TC : ACK TC -> openSER : 487 openSER -> TC : ACK TC2 -> openSER : OK openSER -> TC2 : ACK <-- The call is taking place --> TC -> openSER : 487 openSER -> TC : ACK openser -> TC2 : 487 (openSER relays the 487 once the call has been established to TC2) TC2 -> openSER : ACK TC2 -> openSER : BYE <-- Call is ended but should not --> According to the RFC, once a call has been OKed and a 487 is received, TC2 may go on with the call or send a BYE (up to it). So it behaves the right way (chapter 15. end of 3rd paragraph : "If the INVITE results in 2xx final response(s) to the INVITE, this means that a UAS accepted the invitation while the CANCEL was in progress. The UAC MAY continue with the sessions established by any 2xx responses, or MAY terminate them with BYE."). My question is then : is there a way to prevent this behavior when a terminating carrier doesn't behave correctly, either by preventing relaying of the 487 once it has been ACKed or once the call has been OKed (but I guess we are not RFC compliant then) ? Thanks Bye, Guillaume

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[Kamailio-Users] kamailio at FFG 2009
by Henning Westerholt 17 Mar '09

17 Mar '09

Hi all, i'll present in the next week the Kamailio project at the FFG 2009 conference. Topics of my talk are: - overview about Kamailio - applications and usage scenarios - new and interesting stuff in the 1.5.0 release - future developments, the sip-router project Date: Thursday, 12.03.2009 Place: Karlsruhe, Germany Link to the abstract (in german as the talk will be also held in german): http://www.guug.de/veranstaltungen/ffg2009/abstracts.html#3_2_1 More information about the event: http://www.guug.de/veranstaltungen/ffg2009/ I'll provide the slides and eventual some notes after the talk as usual. Cheers, Henning

1 1

[Kamailio-Users] Build for debug
by mayamatakeshi 17 Mar '09

17 Mar '09

Hello, sometimes when I'm investigating problems in my cfg files i think it could be easier to understand them if i could put some breakpoints on a gdb session. I don't grok makefiles yet. But I suppose there would be a command line option for the kamailio Makefile to instruct it to build the core and modules with debug symbols. So, how do you guys usually compile kamailio with debug symbols? regards, takeshi.

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Re: [Kamailio-Users] Problem starting Kamailio
by Daniel-Constantin Mierla 17 Mar '09

17 Mar '09

Please cc all the time to mailing lists. If you reported an issue and solved it, then it is nice to close the discussion thread by saying what was wrong and how was solved. Others coming later and reading mailing list archive can learn from your experience. Thanks, Daniel On 03/11/2009 10:47 AM, Sara EL KABIRI wrote: > Hello, > > Thanks a lot for you help. > I've already solved the problem :-) > > Regards, > Sara > > On Wed, Mar 11, 2009 at 8:09 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 03/09/2009 08:48 PM, Sara EL KABIRI wrote: > > > Hello, > > I've installed kamailio 1.4.3 (without tls) on ubuntu by > following the provided INSTALL guide. I want to use it as a > presence server but i've got some problems starting it. > > "kamctl start" returns nothing. > > check the syslog file for errors. There you can get some hint. > > Alternatively, set log_stderror=yes in your config file and run > "kamailio" (/usr/sbin/kamailio or /usr/local/sbin/kamailio if > installed from sources) and see what you get on the screen. > > Cheers, > Daniel > > and "kamctl moni" returns "Error opening kamailio's FIFO > /tmp/kamailio_fifo" > I've checked in my .cfg file and the 'loadmodule' and > 'modparam' lines related to mi-fifo module are uncommented. > I've checked also the mi-fifo module was installed indeed > (mi_fifo.so appears in /usr/local/lib/kamailio/modules directory). > > > Could anyone help please? > > -- > Sara > ------------------------------------------------------------------------ > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > > > -- > S. -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] Problem compiling Jabber module
by Joao Gomes Pereira 16 Mar '09

16 Mar '09

I'm using CentOS 5.2 ( 2.6.18-92.1.22.el5) I tried to install it with yum, but it says: "Package expat - 1.95.8-8.2.1.i386 is already installed" Could be some other package missing? Thanks Joao Pereira TCB wrote: > you need to install expat-dev , what distro are you using ? > > On Mon, Mar 16, 2009 at 8:54 PM, gomespereira <gomespereira(a)startel.pt > <mailto:gomespereira@startel.pt>> wrote: > > Hello > While compiling Kamailio's jabber module.... this error appears: > > In file included from xjab_jcon.c:43: > xode.h:31:19: error: expat.h: No such file or directory > In file included from xjab_jcon.c:43: > xode.h:281: error: expected specifier-qualifier-list before > âXML_Parserâ > make[1]: *** [xjab_jcon.o] Error 1 > make[1]: Leaving directory > `/usr/src/kamailio/kamailio-1.5.0-notls/modules/jabber' > make: *** [modules] Error 2 > > Its only works if I exclude the jabber module. > What could be the problem? Are there any modules dependencies? > Thanks > Regards > Joao Pereira > > > > > -- > StarTel - A Rede Livre > Joao Gomes Pereira > www.startel.pt <http://www.startel.pt> > +351 304500650 > sip: gomespereira(a)startel.pt <mailto:gomespereira@startel.pt> > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > > -- > TC -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

1 1

[Kamailio-Users] Using Asterisk as media server
by Juan Perez 16 Mar '09

16 Mar '09

Hello In this scenario, customers behind NAT register in kamailio, records are stored in a db etc. Kamailio uses Asterisk boxes as gateways to send calls to PSTN. Is it possible to use Asterisk as the media server instead of media-proxy or rtp-proxy? In this case the SDP that Asterisk sends in the 180 message needs to me modified? thank you jp

2 1

[Kamailio-Users] Handling of REGISTER messages when Kamilio is in front of Asterisk
by Juan Perez 16 Mar '09

16 Mar '09

Hello We are implementing an architecture with Kamailio and Asterisk with real time and are debating who should be the Registrar for the SIP clients. We are biased towards leaving that to Kamailio and use Asterisk to provide the media services: vmail, conference etc. It looks like this is the recommended way as per several tutorials on the net suggest that solution. As a down side we will have to use a combination of near-end [STUN, ALG] and far-end Nat solutions [rtp-proxy] to handle the media and also are kind of worried about how we should handle the Cass 5 features tipical of a residential line. But a member of our team suggests to use Asterisk for that and have Kamailio forwarding the REGISTER request from the clients to the Asterisk. Asterisk can handle media and also users behind a NAT. We have never done it that way and we are not sure whether it is possible or not, whether it is acceptable SIP-wise or not. For example: if the proxy is load balancing the REGISTER messages then the clients should be given the IP of the proxy to send the REGISTERs and the NAT binding in the customer's NAT/Firewall will be associated to the IP of the proxy, not the IP of the Asterisk. On then other hand, Asterisk as a PBX supports a lot of Class 5 features natively. We would like to hear your input, suggestions and comments about this, what other people are doing. If we had only a few user then 1 Asterisk box should be enough but we are planning on having way more that what an Asterisk machine alone can handle, that is why we want to load balance using kamailio. regards jp

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[Kamailio-Users] ACK reply
by John kader 16 Mar '09

16 Mar '09

Hi all, Can anyone tell me please about which function should I use to reply ACK to PSTN gateway after receiving a REINVITE? Thanks

4 3

[Kamailio-Users] use as proxy to support NAT users
by gomespereira 16 Mar '09

16 Mar '09

Hello My Kamailio isn't working with users behind NAT. If just one of the clients is behind NAT, the calls work fine. When both clients are behind NAT, there is no audio. How can I force the streams to pass through Kamailio server when the users are detected to be behind NAT? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

2 3

[Kamailio-Users] Installing Kamailio in UBUNTU Desktop 8.1110
by Meftah Tayeb 16 Mar '09

16 Mar '09

Hello my friends please anyone help me ? i'm looking for a one friends from the Kamailio Community to install Kamailio for me in my UBUNTU Desktop using SSH or Remote Desktop Viewer this Installation is onlyy for Home usage for Meetings with other Blind users like me. i'm waiting for your responce. thanks my friend

5 7

[Kamailio-Users] Using fr_inv_timer_avp
by mayamatakeshi 16 Mar '09

16 Mar '09

Hello, I'm seeing the following issues when using the latest kamailio 1.4 (Revision: 5689) to set an AVP to control fr_inv_timer: - if I set the AVP to an empty string, according to tm doc, the value of the timer would fall back to the default value. This doesn't happen and the fr_inv_timer is set to zero (not a problem as I just have to set it to the static value. It is just inconsistent with the doc). - if I try to set the timer to something bigger than 60 seconds, the setting doesn't work and it seems the default value is used instead (I tested this with both hard-coding the assignment in the cfg and loading the avp from the usr_preferences). regards, takeshi

2 3

[Kamailio-Users] Incoming T38 Public IP
by michel freiha 15 Mar '09

15 Mar '09

Dear All, Does anyone has an experience with Incoming FAX T38 using a PUBLIC IP on endpoint? Regards

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[Kamailio-Users] LDAP Module and TLS/SSL
by Sven Schulz 15 Mar '09

15 Mar '09

HI, Ive tested the LDAP authentication using Kamailio 1.4.2 and it works great. However, I'd like prevent the passwords being sent back "in the clear" from the LDAP server. Is there a way to use the Kamailio ldap module with TLS/SSL? I already have openssl and openldap-client on the kamailio box and can successfully use the ldapsearch command line tool via TLS. Sven Schulz Penn State University

3 4

[Kamailio-Users] About "476 Unresolvable destination"
by Iñaki Baz Castillo 14 Mar '09

14 Mar '09

Hi, "476 Unresolvable destination" is returned by Kamailio when the destination domain has no resolution. Is it a custom reply code? is it part of some RFC/draft? Thanks. -- Iñaki Baz Castillo

2 1

[Kamailio-Users] ERROR:error opening Openser's FIFO FIFO
by Metshet Seleshi 13 Mar '09

13 Mar '09

please help , why do i get ERROR:error opening Openser's FIFO FIFO ERROR:make sure you have the line 'Modparam('mi_fifo', 'fifo_name', 'FIFO' in your config---which I have ERROR:and also have loaded the mi_fifo module.- yes, it is loaded, I cant do openserctl moni, any idea?? regards ________________________________ From: "users-request(a)lists.kamailio.org" <users-request(a)lists.kamailio.org> To: users(a)lists.kamailio.org Sent: Thursday, March 12, 2009 2:36:05 PM Subject: Users Digest, Vol 46, Issue 39 Send Users mailing list submissions to users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.kamailio.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request(a)lists.kamailio.org You can reach the person managing the list at users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: kamailio start with error Access denied for user.. (svenp) 2. Using fr_inv_timer_avp (mayamatakeshi) 3. DID assignment & routing (Robert McGilvray) 4. Does Kamailio can be used as ptt server? (doolin wu) 5. Siremis: login problems (Luntras Loredana) 6. Re: Siremis: login problems (Luntras Loredana) ---------------------------------------------------------------------- Message: 1 Date: Thu, 12 Mar 2009 04:44:16 -0700 (PDT) From: svenp <s.pastorik(a)westfalen-blatt.de> Subject: Re: [Kamailio-Users] kamailio start with error Access denied for user.. To: users(a)lists.kamailio.org Message-ID: <22474624.post(a)talk.nabble.com> Content-Type: text/plain; charset=us-ascii Hello, i foud the error in my config. I think user and pw is in config file kamailio.cfg an alias from file kamctlrc. original: modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") The right Username und PW must configured in the kamailio.cfg too. My kamailio Server is now running :). -- View this message in context: http://www.nabble.com/kamailio-start-with-error-Access-denied-for-user..-tp… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. ------------------------------ Message: 2 Date: Thu, 12 Mar 2009 20:54:49 +0900 From: mayamatakeshi <mayamatakeshi(a)gmail.com> Subject: [Kamailio-Users] Using fr_inv_timer_avp To: kamailio <users(a)lists.kamailio.org> Message-ID: <15b9404e0903120454s34c0b9eqd57bf5313d89ba75(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Hello, I'm seeing the following issues when using the latest kamailio 1.4 (Revision: 5689) to set an AVP to control fr_inv_timer: - if I set the AVP to an empty string, according to tm doc, the value of the timer would fall back to the default value. This doesn't happen and the fr_inv_timer is set to zero (not a problem as I just have to set it to the static value. It is just inconsistent with the doc). - if I try to set the timer to something bigger than 60 seconds, the setting doesn't work and it seems the default value is used instead (I tested this with both hard-coding the assignment in the cfg and loading the avp from the usr_preferences). regards, takeshi ------------------------------ Message: 3 Date: Thu, 12 Mar 2009 10:01:40 -0400 From: "Robert McGilvray" <rmcgilvr(a)globeop.com> Subject: [Kamailio-Users] DID assignment & routing To: <users(a)lists.kamailio.org> Message-ID: <888FFEF718B66147A7F4EEAED6E3B385025BADEA(a)wpf1malprd4.globeop.com> Content-Type: text/plain; charset="utf-8" I'm deploying Kamailio in an enterprise environment and I'm looking for a better way to handle the mapping of external DIDs to SIP usernames, number re-writing etc. I'm currently using ENUM in my test setup but I'd prefer to go with SQL storage and have that data written out to a faster database for lookups by Kamailio. I have a mix of users that register directly to the proxy and also several PBXs that I need to route blocks of numbers to through Cisco routers. For location A I would need to route any four digit number that starts with a 3 (3759), EXCEPT ones specifically defined in my database since they might be mapped directly to a user. Also for any outbound call (9 prefix) I need to check the DB to see if it's local. If I move to DB storage how do I handle things like parallel/sequential forking? ENUM makes it very easy - for example: 1.0.0.0.1 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:8700@172.20.152.132!" . NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:901191986701####@172.20.152.132!" . NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:901191996700####@172.20.152.132!" . If I have a SIP user "bob(a)domain.com" a simple location lookup works for that, but I also need to be able to map a e.164 did to him, and a short form with just the last 4 digits for my internal dialing. I would need a database schema that does the equivalent to this: 0.1.5.3.3.9.2.4.1.9.1 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:bobm@domain.com!" . 0.1.5.3 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:bobm@domain.com!" . Any suggestions? Thanks Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. (W1)

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[Serusers] Questions about strip() and altering ANI/From: numbers
by Frank Durda IV 13 Mar '09

13 Mar '09

When using strip() on a E.164 format number (eg +17895551212), does strip(2) delete the +1, or is it going to delete "17" in the shown example? I seem to be getting the latter behavior. If the latter is what is intended, how does one get rid of the plus? Second, is there something equivalent to strip() for editing the ANI/From: number? As I am doing PSTN gateway conversions, I have to strip the +1 from a E.164 ANI/From: (if sent) at some point so that the destination PSTN switch can correctly do the Calling Name database lookup. The North American Calling Name databases will only match if exactly ten digits are provided, so the ANI/From: has to be scrubbed down to ten digits if the country code in the ANI/From: begins with +1. Thanks!

2 2

[Kamailio-Users] lcr-module, ping and load_gws_from_grp (kamailio 1.5)
by Stefan Prelle 12 Mar '09

12 Mar '09

Hi, I just noticed something weird when working with the lcr module. My setup has 5 gateways - 4 in group 1, one in group 2. The lcr-table has an entry for each group, both reacting on the same prefix. I am using load_gws_from_grp("1") to get a gateway for a target. This usually works fine. If now all 4 gateways in group 1 ar disabled because the alive check failed, load_gws_from_grp("1") returns the gateway from group 2 (or other groups where the prefix from the lcr-table matches. I would consider this a bug, since I explicitly requested gateways from a specific group. Regards, Stefan

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[Serusers] Problem with Record-Route not being added
by Frank Durda IV 12 Mar '09

12 Mar '09

I have recently noticed that despite having a $record_route_nated = true; in the route[PSTN_FORWARD] of ser.cfg, the INVITES and such sent on do not have a Record-Route header. In our NATed situation, probably the only thing this would cause problems with would be when the called party hangs up first, SER and the calling party never find out about it, since the calling party IP is not reachable from the PSTN switch NATed address space, so the single BYE message sent by the PSTN switch probably whizzes by to an unreachable destination address. So the calling party remains off-hook and just gets dead air. When the calling party finally disconnects, the SER passes the caller-initiated BYE to the PSTN switch which returns a 481 because it thinks the call was over ages earlier. Or at least this is what appears to be happening. Here is the PSTN_FORWARD in use, not that different from the PSTN gateway example ser.cfg: route[PSTN_FORWARD] { # Set where packets should be sent. If you don't do this here, # OPTIONS and other messages will be transmitted back to the # outside interface and processed all over again, and you eventually # get a Too Many Hops error on every OPTIONS message. # Code to make this point in multiple directions would be needed # to implement two-way origination. rewritehost("10.x.x.x"); #Send to SIP-TDM switch if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); } else if (method == "INVITE") { # Force record routing if either party is behind NAT $record_route_nated = true; # log(1, "NAT Mangle Force RTP Proxy\n"); force_rtp_proxy("e"); } t_on_reply("PSTN_REPLY"); # if this is called from the failure route we need to open a new branch if (isflagset(FLAG_FAILUREROUTE)) { append_branch(); } # if this is an initial INVITE (without a To-tag) we might try another # (forwarding or voicemail) target after receiving an error if (method=="INVITE" && @to.tag=="") { t_on_failure("FAILURE_ROUTE"); } # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); } drop; } The resulting INVITE that is being sent on to the PSTN switch looks like: INVITE sip:+1xxxxxxxxxx@10.x.x.x:5060;npdi=yes SIP/2.0 Via: SIP/2.0/UDP 10.x.x.x;branch=z9hG4bK6804.1cb4cb23.0 Via: SIP/2.0/UDP 10.x.x.x:5060;branch=z9hG4bK1kpu0b20e0fgogk9c6d0.1 f: <sip:+1xxxxxxxxxx;jip=xxxxxx@10.x.x.x:5060;user=phone>;tag=d0a-13c4-83e003-64bd22fa-83e003 t: <sip:+1xxxxxxxxxx@10.x.x.x:5060;user=phone> i: a249e890d8184d0a13c483e0033230d1ff3b57654c1d4018-0143-8166 CSeq: 1 INVITE User-agent: CS2000_NGSS/9.0 P-Asserted-Identity: <sip:+1xxxxxxxxxx;jip=xxxxxx@10.x.x.x;user=phone> Max-Forwards: 16 k: 100rel Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK m: <sip:10.x.x.x:5060;transport=udp> c: application/SDP l: 231 v=0 o=- 15109843 15109843 IN IP4 10.x.x.x s=- c=IN IP4 10.x.x.x t=0 0 m=audio 44464 RTP/AVP 0 101 b=AS:80 a=ptime:20 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=nortpproxy:yes So, any ideas as to why the "$record_route_nated = true" doesn't seem to be having any effect, eg no Record-Route shows up in outbound messages? I know SER is passing through the above code because I can uncomment the "NAT Mangle" comment and see it in the logs. Is there perhaps some module that must be present that we may have eliminated that makes Record-Route get emitted? Thanks again!

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[Kamailio-Users] DID assignment & routing
by Robert McGilvray 12 Mar '09

12 Mar '09

I'm deploying Kamailio in an enterprise environment and I'm looking for a better way to handle the mapping of external DIDs to SIP usernames, number re-writing etc. I'm currently using ENUM in my test setup but I'd prefer to go with SQL storage and have that data written out to a faster database for lookups by Kamailio. I have a mix of users that register directly to the proxy and also several PBXs that I need to route blocks of numbers to through Cisco routers. For location A I would need to route any four digit number that starts with a 3 (3759), EXCEPT ones specifically defined in my database since they might be mapped directly to a user. Also for any outbound call (9 prefix) I need to check the DB to see if it's local. If I move to DB storage how do I handle things like parallel/sequential forking? ENUM makes it very easy - for example: 1.0.0.0.1 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:8700@172.20.152.132!" . NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:901191986701####@172.20.152.132!" . NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:901191996700####@172.20.152.132!" . If I have a SIP user "bob(a)domain.com" a simple location lookup works for that, but I also need to be able to map a e.164 did to him, and a short form with just the last 4 digits for my internal dialing. I would need a database schema that does the equivalent to this: 0.1.5.3.3.9.2.4.1.9.1 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:bobm@domain.com!" . 0.1.5.3 NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:bobm@domain.com!" . Any suggestions? Thanks Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. (W1)

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[Kamailio-Users] SQLOps Module
by Andreas Heise 12 Mar '09

12 Mar '09

Hi, first of all thanks for the great sqlops module. Is "|sql_result_free(result)|" needed before I leave the script by "exit;" to avoid memory trouble or is it only needed to query again in the same request processing? Is there any way to check if the query was not successful (db unreachable, systax error in query)? I've seen a warning in the syslog, but I need to check it in the script. BTW in the doc the example for 1.4.2. seems to be wrong? regards, Andreas

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[Kamailio-Users] Does Kamailio can be used as ptt server?
by doolin wu 12 Mar '09

12 Mar '09

Hi, I'm trying to find a open source ptt (push to talk) server? Does Kamailio can act as this role? Or do you know any open source ptt server? Thanks! -- Steven S.W

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[Kamailio-Users] Siremis: login problems
by Luntras Loredana 12 Mar '09

12 Mar '09

Helo, I've installes Siiremis and I have the following error when I try to log in. "An exception occurred while executing this script: Error message: #256, No Database information found in the config file. Script name and line number of error: /var/www/siremis-0.9.1/openbiz/bin/Configuration.php:42" I have only change the db_name in Config.xml from openser_siremis to openser. (the only data base that I have is openser that I've installed with opensedbctl create). Can anybody help me, please? Thanks, Loredana

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[Kamailio-Users] kamailio start with error Access denied for user..
by svenp 12 Mar '09

12 Mar '09

hello, ich have installed the latest version from kamailio kamailio-1.5.0-tls_src.tar.gz Compiled and installed with this parameters NICER=1 make all make include_modules="db_mysql" all make install the database tables created with /usr/local/sbin/kamdbctl create and then start kamailio with kamctl start the output on cmd is : gateway2:/usr/local/sbin # kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed my logfile: ... Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing record_route, line 310 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing is_method, line 313 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:parse_methods: methods 0x1 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:textops:fixup_method: using id for method [INVITE/1] Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing append_hf, line 320 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing is_method, line 334 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:parse_methods: methods 0x2200 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:textops:fixup_method: using id for methods [PUBLISH,SUBSCRIBE/8704] Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing is_method, line 337 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:parse_methods: methods 0x80 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:textops:fixup_method: using id for method [REGISTER/128] Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:fix_actions: fixing save, line 339 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:db_do_init: connection 0x654220 not found in pool Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:db_mysql:db_mysql_new_connection: opening connection: mysql://xxxx:xxxx@localhost/openser Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:db_mysql:db_mysql_new_connection: driver error: Access denied for user 'openser'@'localhost' (using password: YES) Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:core:db_do_init: could not add connection to the pool Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:usrloc:register_udomain: failed to open database connection Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:registrar:domain_fixup: failed to register domain Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 339 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: CRITICAL:core:fix_expr: fix_actions error Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: ERROR:core:main: failed to fix configuration with err code -1 Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:destroy_modules: shutting down module auth_db Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:destroy_modules: shutting down module auth Mar 9 13:17:19 gateway2 /usr/local/sbin/kamailio[9496]: DBG:core:destroy_modules: shutting down module acc .... username and pw ist default. i try it on console and login is ok: gateway2:/usr/local/sbin # mysql -uopenser -h localhost -popenser Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5894 to server version: 5.0.26 Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql> use openser Database changed mysql> Bye what is wrong in my config? -- View this message in context: http://www.nabble.com/kamailio-start-with-error-Access-denied-for-user..-tp… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] How terminate inbound pstn calls
by carl Lougher 12 Mar '09

12 Mar '09

Howdy, Can anyone help me configure my .cfg file to point inbound calls from the pstn gateway to my Trixbox on a sip trunk to Kamailio? Outbound calls work fine and trunk is registered in local DB. Cheers, Taff...

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[Kamailio-Users] Modifying the URI in the header From more than once
by mayamatakeshi 12 Mar '09

12 Mar '09

Hello, I'm trying to add CallForwarding to my kamailio.cfg. Before relaying an INVITE from one subscriber to another, I'm required to modify the From username to the PSTN number of the caller. So I use uac_replace_from and this works fine. I can see kamailio sendind this record-route: Record-Route: <sip:192.168.33.2;lr=on;ftag=qunug47bd1hc74ir26iv;vsf=AAAAAERUW1NLXloEeFVZKRYIDAZKBwRPEQYEH1oLCQAEFS5qcA--;did=b76.5c17ed25> But then, when the callee fails to answer and failure_route is called, I need to modify the From username to the PSTN number of the callee before doing the forward. If I simply try to call uac_replace_from again inside failure_route, record-route is sent with a duplicate vsf parameter like this: Record-Route: <sip:192.168.33.2;lr=on;ftag=qunug47bd1hc74ir26iv;vsf=AAAAAERUW1NLXloEeFVZKRYIDAZKBwRPEQYEH1oLCQAEFS5qcA--;did=b76.5c17ed25;vsf=AAAAAERUW1NLXloEeFVeKRYIDAZKBwRPEQYEH1oLCQAEFS5qcA--> And this causes two problems: - the message becomes too big and the GW replies with "513 Message too large" - the reply is sent back to the caller with duplicate From header and some UAs do not acknowledge the reply. So, is it possible to modify the From header more than once? regards, takeshi

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[Kamailio-Users] PIKE module
by Juan Perez 12 Mar '09

12 Mar '09

Hello I need some clarification here; the Help file for this module says this: ///////////////////////////////////////////////////////////////////////////////////////////////////////////////////// The module keeps trace of all (or selected ones) incoming request's IP source and blocks the ones that exceeded some limit. Works simultaneous for IPv4 and IPv6 addresses. The module does not implement any actions on blocking - it just simply reports that there is a high traffic from an IP; what to do, is the administator decision (via scripting). ///////////////////////////////////////////////////////////////////////////////////////////////////////////////////// I understand from the first part that it blocks the IPs, but last sentence says otherwise. How is it supposed to work? txs jp

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[Kamailio-Users] htable - grabbing a value randomly
by Mik Cheez 11 Mar '09

11 Mar '09

I'm hoping to store several gateway IP's in a hash table using HTable, then grab one of them randomly. Does anyone have a recommendation as to how this can be done, or perhaps a better way to do it? I chose HTable because a) I want to reload the values every x minutes and b) I'm using it for other purposes. Mike

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[Kamailio-Users] Announcement Server
by Bruno Rodrigues 11 Mar '09

11 Mar '09

Hi All, First, sorry I have a bad english. Second, Anybody know if is possible redirect the call to one MediaServer when the destination answer the call? In the Media Server, the callee will hear a announcement, and the destination will hear a another announcement. The announcement has the same duration. When the announcement finish, the RTP will be established between the UA1 and UA2 The signaling I think will be this UA1 Kamailio UA2 Media Server Invite -----------------> - Received a normal Invite Invite----------------> - Forward The Invite <------------------200 - Receive the 200 OK <------------------200 - Forward the 200 OK <-------------re-Invite re-Invite ------------> - Kamailio will update the media part to the Media Server <----RTP Announce 1-----> - RTP are established <----------------------RTP Announce 2 ---------------------------------> - RTP are established <-------------re-Invite re-Invite ------------> - Kamailio after the announcement are finished update the Media part to UA1 ip media and UA2 ip media <-----------------RTP-------------------------> - RTP between UA1 and UA2 If have another way to do this, please let me know Thank You in advanced Bruno Rodrigues

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[Kamailio-Users] DNS lookup inside cfg
by Uriel Rozenbaum 11 Mar '09

11 Mar '09

Genlemen, I'll be quick: is there any way to perform a DNS lookup within the .cfg? The example would be: avp(s:server_name) = "kamailio.org"; avp(s:server_ip) = "192.168.1.100"; if (DNS_LOOKUP_FUNCTION(avp(s:server_name)) == avp(s:server_ip)) { ... do something ... } else { ... do something else ... } Thanks in advance, Uriel

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[Kamailio-Users] [OT] sip router masterclass
by Daniel-Constantin Mierla 11 Mar '09

11 Mar '09

Hello, my apologizes if this email look business oriented for this list, but to answer some questions in the last time, there is now a dedicated professional training program for kamailio (openser) and sip-router.org: http://www.asipto.com/index.php/sip-router-masterclass/ Community wise, each time will be a Social Networking Event (dinner) where everybody can join and have a chat with other folks developing or using these projects. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] new module: memcached
by Henning Westerholt 11 Mar '09

11 Mar '09

Hi all, i just added a new module to the repository - 'memcached'. Its allows to connect to a memcached server, and supports add, get, delete and atomic operations via pseudo-variables. Further details can be found in the module documentation: http://kamailio.org/docs/modules/devel/memcached.html If the pv module is imported into the sip-router it should be also work there, i'll migrate the module then to this tree. Best regards, Henning Westerholt

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[Kamailio-Users] Strange behaviour when RADIUS server is down
by Antonio Reale 11 Mar '09

11 Mar '09

Hi all, I see a very strange behaviour using accounting on RADIUS. If RADIUS server is up and running, all is OK. If it is down dialog module (in db mode) can't work properly: In the log I see: after 200 OK to INVITE CRITICAL:dialog:log_next_state_dlg: bogus event 6 in state 2 for dlg 0xb61c0918 [360:515907930] with clid 'B8108B3CEBD9B8E3E1CC23DE8361D35F@sipsvr' and tags 'C319BA6F3E777E04867202774CE6443B' '' after BYE CRITICAL:dialog:log_next_state_dlg: bogus event 7 in state 2 for dlg 0xb61c0918 [360:515907930] with clid 'B8108B3CEBD9B8E3E1CC23DE8361D35F@sipsvr' and tags 'C319BA6F3E777E04867202774CE6443B' '' and no record is written in the dialog table. All come back to work after starting RADIUS server. This is not the main problem... In some circumstance kamailio fails to route properly the calls, due retransmissions; for instance after the client answers I see a lot of 200 OK sent to kamailio because it doesn't send the ACK immediately. In the log I see for each 200 OK: ... Mar 9 11:34:01 [27466] DBG:tm:timer_routine: timer routine:3,tl=0xb61a8ecc next=(nil), timeout=75 Mar 9 11:34:01 [27466] DBG:tm:delete_handler: removing 0xb61a8e68 Mar 9 11:34:01 [27466] DBG:tm:delete_cell: delete_cell 0xb61a8e68: can't delete -- still reffed (1) Mar 9 11:34:01 [27466] DBG:tm:set_timer: relative timeout is 2 Mar 9 11:34:01 [27466] DBG:tm:insert_timer_unsafe: [3]: 0xb61a8ecc (77) Mar 9 11:34:01 [27466] DBG:tm:delete_handler: done ... I was not able to replicate this problem. Solved after starting RADIUS server, but the problem doesn't occur if I stop it. I'm using kamailio 1.4.3, but the last problem occurred also on a production system with openser 1.2.3. After customer's RADIUS server went down, openser wasn't able to route all calls. Thank you very much for support. Regards, Antonio.

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[Kamailio-Users] sl README does not know send_reply()
by jh＠tutpro.com 11 Mar '09

11 Mar '09

the new send_reply() function is not documented in sl/README. another related thing: would it be logical to also have corresponding reply_error() function especially when t_relay seems to create transaction even in case of some internal errors. -- juha

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[Kamailio-Users] Problem starting Kamailio
by Sara EL KABIRI 11 Mar '09

11 Mar '09

Hello, I've installed kamailio 1.4.3 (without tls) on ubuntu by following the provided INSTALL guide. I want to use it as a presence server but i've got some problems starting it. "kamctl start" returns nothing. and "kamctl moni" returns "Error opening kamailio's FIFO /tmp/kamailio_fifo" I've checked in my .cfg file and the 'loadmodule' and 'modparam' lines related to mi-fifo module are uncommented. I've checked also the mi-fifo module was installed indeed (mi_fifo.so appears in /usr/local/lib/kamailio/modules directory). Could anyone help please? -- Sara

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[Kamailio-Users] Force rtp proxy
by michel freiha 11 Mar '09

11 Mar '09

Hi all, I have a question regarding rtp proxy...I need to force rtp proxy in order that all rtp packets pass through...The issue now is the first INVITE does contains the PSTN GW IP instead of RTP Proxy IP...My config is as below: if (is_method("INVITE")) { # setflag(4); # do accounting fix_nated_sdp("1"); } if(!cr_route("default", "0", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In cas of failure, re-route the request t_on_failure("1"); t_on_reply("1"); force_rtp_proxy(); t_relay(); } onreply_route[1] { # if (t_check_status("2[0-9][0-9]") ) if((t_check_status("200|183|180") && search("Content_Type: application/sdp")) || search("Content-Type: application/sdp")) { force_rtp_proxy(); } exit; } Please let me know what's wrong here and how i can fix this issue Regards

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[Kamailio-Users] Forgetting to call unforce_rtp_proxy
by mayamatakeshi 11 Mar '09

11 Mar '09

Hello, is there any risk of having resources leaking if I fail to call unforce_rtp_proxy? Like having rtpproxy not freeing ports? I'm mean, my cfg is becoming bigger and I may fail to call this function somewhere. Should I be extremely careful about this? regards, takeshi

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[Kamailio-Users] LCR With 300,000 Routes
by Geoffrey Mina 10 Mar '09

10 Mar '09

I have spec'd out how my LCR table would look based my current PSTN gateway providers. Looks like I would need to load roughly 300,000 routes to properly represent everything. This server needs to be able to support spikes of 100 call setups per second. What is the general consensus on this? And, if it's a No-Go, what alternatives do I have? I have looked briefly at the CarrierRoute module, but haven't actually played with it. Thanks! Geoff

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Re: [Kamailio-Users] handling of DNS-SRV in kamailio-1.4.3 & 1.5.0
by Klaus Feichtinger 10 Mar '09

10 Mar '09

Hello Daniel, I've done several test with different scenarios: 1) including 4 hosts in the answer of the SRV reply message; the 1st and 2nd host are unreachable, the third an fourth are reachable - the SIP message never reached the target (= host 3) 2) including 4 hosts in the answer of the SRV reply message; the SIP service is not running on the first and second host; the third and fourth have the service activated - the SIP message is transmitted to the second host after the fr_timeout of the TM module, but is not forwarded to the target (= host 3); it is retransmitted to host 2 until SIP timeout The conclusion is: when only on master and slave are in the answer of the SRV reply message it works fine. But starting with the third host the service hangs on the second host and does not try reaching an alternative target (as listed in the SRV response). In the syslog I can only see entries like this (but no ERROR message): Mar 9 15:36:21 lennysrv /usr/local/sbin/kamailio[25041]: DBG:core:mk_proxy: doing DNS lookup... Mar 9 15:36:21 lennysrv /usr/local/sbin/kamailio[25041]: DBG:core:a2dns_node: storing kamailio2.test.loc:5080 Mar 9 15:36:21 lennysrv /usr/local/sbin/kamailio[25041]: DBG:core:a2dns_node: storing kamailio3.test.loc:5060 Mar 9 15:36:26 lennysrv /usr/local/sbin/kamailio[25043]: DBG:core:mk_proxy: doing DNS lookup... Mar 9 15:36:28 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=408, flags=1 Mar 9 15:36:28 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:t_should_relay_response: trying DNS-based failover Mar 9 15:36:28 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:do_dns_failover: new destination available Mar 9 15:36:31 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:is_3263_failure: dns-failover test: branch=1, last_recv=408, flags=1 Mar 9 15:36:31 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:t_should_relay_response: trying DNS-based failover Mar 9 15:36:31 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:do_dns_failover: new destination available Mar 9 15:36:34 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:is_3263_failure: dns-failover test: branch=2, last_recv=408, flags=1 Mar 9 15:36:34 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:t_should_relay_response: trying DNS-based failover Mar 9 15:36:34 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:do_dns_failover: new destination available Mar 9 15:36:37 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:is_3263_failure: dns-failover test: branch=3, last_recv=408, flags=1 Mar 9 15:36:37 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:t_should_relay_response: trying DNS-based failover Mar 9 15:36:37 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:do_dns_failover: new destination available Mar 9 15:36:40 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:is_3263_failure: dns-failover test: branch=4, last_recv=408, flags=1 Mar 9 15:36:40 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:t_should_relay_response: trying DNS-based failover Mar 9 15:36:40 lennysrv /usr/local/sbin/kamailio[25047]: DBG:tm:do_dns_failover: new destination available Mar 9 15:36:43 lennysrv /usr/local/sbin/kamailio[25045]: DBG:tm:is_3263_failure: dns-failover test: branch=5, last_recv=487, flags=1 The syslog output seems to be okay, because kamailio theoretically makes a dns_failover. But practically it is only sent to the first and second target. Do you have any other idea what could be wrong? I haven't found any specific parameters that could influence this behaviour of kamailio.... regards, Klaus F. -------- Original-Nachricht -------- > Datum: Fri, 06 Mar 2009 10:59:28 +0200 > Von: Daniel-Constantin Mierla <miconda(a)gmail.com> > An: Klaus Feichtinger <klaus.feichtinger(a)gmx.net> > CC: users(a)lists.kamailio.org > Betreff: Re: [Kamailio-Users] handling of DNS-SRV in kamailio-1.4.3 & 1.5.0 > Hello, > > On 03/05/2009 02:51 PM, Klaus Feichtinger wrote: > > Hello, > > > > I have to relay SIP requests to a special "redundant" destination (= > FQDN - e.g. "test.kamailio.loc") that consists of 3 hosts with different > priorities. > > > > After startup Kamailio creates a DNS-SRV request to the DNS server and > receives a response with following answer (e.g.): > > _sip._udp.test.kamailio.loc: type SRV, class IN, priority 10, weight > 100, port 5060, target host1.test.kamailio.loc > > _sip._udp.test.kamailio.loc: type SRV, class IN, priority 20, weight > 90, port 5060, target host2.test.kamailio.loc > > _sip._udp.test.kamailio.loc: type SRV, class IN, priority 30, weight > 80, port 5060, target host3.test.kamailio.loc > > > > In worst case - if the prime and secondary host (host1 and host2) are > unavailable - I expect that kamailio tries relaying the request to the third > host in the list. But it doesn't. It makes retransmission of the original > invite to the second host and does not try reaching the third host. > > > > For tuning the switchover to the alternative target(2) I have set the > module parameter "fr_timer" for TM to 3 seconds (modparam("tm", "fr_timer", > 3)). > > > > > > Is this behaviour of Kamailio-1.4.3 and Kamailio-1.5.0 (I've tested it > with both versions) as expected (= limited to support only two answers in a > DNS SRV reply)? Does anybody have experience with this scenario? Can > anybody give me a hint? > > > should be no such limitation. If you try with 4 hosts what happens? Do > you get any errors in the syslog? > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com -- Computer Bild Tarifsieger! GMX FreeDSL - Telefonanschluss + DSL für nur 17,95 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a

2 1

[Kamailio-Users] t_relay with result code -6 creates transaction
by jh＠tutpro.com 10 Mar '09

10 Mar '09

i was playing with t_relay and tcp and noticed that when forwarding to a tcp contact fails, transaction is still created although README says: * 0x02 - do not internally send a negative reply in case of forward failure (due internal error, bad RURI, bad message, etc). When a forward failure occurs, no SIP request is relayed and therefore no negative reply or timeout will show up on the failure_route (if one is set). It applies only when the transaction is created. By default one negative reply is sent. Useful if you want to implement a serial forking in case of failure. and In case of error, the function returns the following codes: ... * -6 - generic send failed while testing this i had two contacts with different q values: Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: DBG:tm:t_load_contacts: loaded contact <sip:test@192.98.101.10:5074#012#012#012udp:192.98.101.10:5060#0120#012d> with q_flag <0> Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: DBG:tm:t_load_contacts: loaded contact <sip:jh@192.98.101.10:5555;transport=tcp#012#012#012#01216#0124> with q_flag <4> i then called next_contacts() followed by t_relay("0x03") and tested the result: while (1) { xlog("L_INFO", "INFO: Routing initial $rm to <$ru> and <$bR>\n"); t_relay("0x03"); switch ($retcode) { case 1: xlog("L_INFO", "INFO: retcode is <1>\n"); exit; case -1: case -2: case -3: sl_reply_error(); exit; case -4: xlog("L_INFO", "INFO: retcode is <-4>\n"); case -5: xlog("L_INFO", "INFO: retcode is <-5>\n"); case -6: xlog("L_INFO", "INFO: retcode is <-6>\n"); if (!t_next_contacts()) { send_reply("480", "Temporarily Unavailable"); exit; }; }; }; i got to syslog: Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: DBG:tm:t_next_contacts: entering next_contacts WITHOUT transaction Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: DBG:tm:t_next_contacts: R-URI contact is <sip:jh@192.98.101.10:5555;transport=tcp#012#012#012#01216#0124> Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: INFO: Routing initial INVITE to <sip:jh@192.98.101.10:5555;transport=tcp> and <<null>> Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: ERROR:core:tcp_blocking_connect: poll error: flags 18 Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: ERROR:core:tcp_send: connect failed Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: ERROR:tm:msg_send: tcp_send failed Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: ERROR:tm:t_forward_nonack: sending request failed Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: INFO: retcode is <-6> the first debug line above is a new one that i added to the beginning of next_contacts(): t = get_t(); if (!t || (t == T_UNDEFINED)) { LM_DBG("entering next_contacts WITHOUT transaction\n"); } else { LM_DBG("entering next_contacts WITH transaction\n"); } then the script goes on and calls t_next_contacts()/t_relay("0x03") again and now i get: Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: DBG:tm:t_next_contacts: entering next_contacts WITH transaction Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: DBG:tm:t_next_contacts: next branch contact is <sip:test@192.98.101.10:5074#012#012#012udp:192.98.101.10:5060#0120#012d> Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: INFO: Routing initial INVITE to <sip:jh@192.98.101.10:5555;transport=tcp> and <<sip:test@192.98.101.10:5074>;q=0> Mar 10 06:50:07 taimen /usr/sbin/kamailio[18101]: INFO: retcode is <1> how is it possible that the transaction was created even when the first t_relay failed with result code -6? this example should prove that scripting tm like this NOT user friendly and prone to errors. -- juha ps. i think we agreed earlier that failing forward to a tcp contact is NOT an error and should not be printed to syslog as an error.

2 1

[Kamailio-Users] bug in subscribe when contact uses tcp transport?
by jh＠tutpro.com 10 Mar '09

10 Mar '09

> The modules from 1.5 should work out of the box with 1.4. My plan is to > test them a while in 1.5 and backport before next minor release > 1.4.x. daniel, unfortunately the above does not seem to be true. when i copy presence module from 1.5 and try to make it, i get make[1]: Entering directory `/usr/src/4.1-src/openxg-kamailio/modules/presence' publish.c:46:26: error: ../../cmpapi.h: No such file or directory Compiling bind_presence.c ... Compiling notify.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"presence"' -DNAME='"kamailio"' -DVERSION='"1.4.3-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DSVNREVISION='"2:5638M"' -DUSE_TLS -DUSE_SCTP -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/ssl/include -I/usr/local/include -I/usr/include/openssl -I/usr/include/libxml2 -I/usr/local/include/libxml2 -I/usr/local/include -c notify.c -o notify.o notify.c: In function ??get_p_notify_body??: notify.c:728: error: ??EVENT_DIALOG_SLA?? undeclared (first use in this function) notify.c:728: error: (Each undeclared identifier is reported only once notify.c:728: error: for each function it appears in.) -- juha

1 0

[Kamailio-Users] bug in subscribe when contact uses tcp transport?
by jh＠tutpro.com 09 Mar '09

09 Mar '09

if user subscribes using tcp transport, kamailio presence server replies with 200 ok where contact specifies udp transport. this looks like a bug to me, because re-subscribe to contact uri then uses udp transport, which may not even be enabled by firewall between ua and kamailio. do others agree that this is a bug? if so, any volunteers to fix it? -- juha +++ 6-3-2009 10:27:39.692364 INFO SIP ::send_sip_tcp Send to: tcp:192.98.101.10:5060 SUBSCRIBE sip:test@test.fi SIP/2.0 Via: SIP/2.0/TCP 192.98.101.10:5074;rport;branch=z9hG4bKrzmiknas Max-Forwards: 70 To: <sip:test@test.fi> From: "Juha Heinanen" <sip:jh@test.fi>;tag=gonhy Call-ID: ljkaayblocflyyu@taimen CSeq: 62 SUBSCRIBE Contact: <sip:jh@192.98.101.10:5074;transport=tcp> Accept: application/pidf+xml Event: presence Expires: 3600 User-Agent: Twinkle/1.4.1 Content-Length: 0 +++ 6-3-2009 10:27:39.707261 INFO SIP ::process_sip_msg Received from: tcp:192.98.101.10:5060 SIP/2.0 202 OK Record-Route: <sip:192.98.101.10;r2=on;lr> Record-Route: <sip:192.98.101.10;transport=tcp;r2=on;lr> Via: SIP/2.0/TCP 192.98.101.10:5074;received=192.98.101.10;rport=54705;branch=z9hG4bKrzmiknas To: <sip:test@test.fi>;tag=3d2810ff0e005fca9b24aee8694a9a3d-a5a3 From: "Juha Heinanen" <sip:jh@test.fi>;tag=gonhy Call-ID: ljkaayblocflyyu@taimen CSeq: 62 SUBSCRIBE Expires: 3600 Contact: <sip:192.98.101.10:5082> Server: OpenXg Kamailio (1.5.0-pre2-tls (i386/linux)) Content-Length: 0

4 16

[Kamailio-Users] Forget about my first mail..
by Sara EL KABIRI 09 Mar '09

09 Mar '09

The mysql database problem is solved, but i have now another problem... I'm sending a second mail... Sorry for the spam.

1 0

[Kamailio-Users] Problem starting Kamailio
by Sara EL KABIRI 09 Mar '09

09 Mar '09

Hello, I've installed kamailio 1.4.3 (without tls) on ubuntu by following the provided INSTALL guide. I want to use it as a presence server but i've got some problems creating the MySQL database: I got this error : *ERROR* *1105* (HY000): *MyISAM* *table* ‘version′ *is* *in* *use* (*most* * likely* *by* *a* *MERGE* *table*). *Try* *FLUSH* *TABLES*… Could anyone help please? -- Sara

1 0

[Kamailio-Users] lcr-module, ping and load_gws_from_grp (kamailio 1.5)
by Stefan Prelle 09 Mar '09

09 Mar '09

Hi, I just noticed something weird when working with the lcr module. My setup has 5 gateways - 4 in group 1, one in group 2. The lcr-table has an entry for each group, both reacting on the same prefix. I am using load_gws_from_grp("1") to get a gateway for a target. This usually works fine. If now all 4 gateways in group 1 ar disabled because the alive check failed, load_gws_from_grp("1") returns the gateway from group 2 (or other groups where the prefix from the lcr-table matches. I would consider this a bug, since I explicitly requested gateways from a specific group. Regards, Stefan

1 0

[Kamailio-Users] Send a status from reply route
by Kévin L'heveder 09 Mar '09

09 Mar '09

Hi, I work with Kamailio as Proxy and a call manager. From a MI_datagram script I manage kamailio. My script check the current data rate on different links. So if the max data rate is reached my script must cancel the call. That works good because I can use t_reply or sl_send_reply from request route and send cancel or status error. But in one case I need to cancel the call after receiving the trying (after INIVTE) from the call manager (because the call manager could preempt a call to free some space one the link) My trouble is that i can't use t_reply or sl_send _reply from REPLY_ROUTE and to cancel the call properly i need to send a status error, so i can't use t_uac_dlg from my management interface (because this function sends request only). I didn't find out a solution to do this, so is someone have one ...? Thanks, Koon _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

1 0

[Kamailio-Users] AUTO: Miodrag Radulovic is out of the office (returning 16-03-2009)
by Miodrag Radulovic 09 Mar '09

09 Mar '09

I am out of the office until 16-03-2009. Privremeno nedostupan - Out-of-Office Message. Poruka koju ste poslali je prispela i bice sacuvana, napominjem da je za vreme mog puta pristup elektronskoj posti veoma ogranicen. The message that you have sent will be saved in my inbox, but please note that I will be out of the country and my access to my email will be very limited. Miodrag Radulovic General Manager COMUTEL Note: This is an automated response to your message "Users Digest, Vol 46, Issue 27" sent on 9.3.2009 4:07:38. This is the only notification you will receive while this person is away.

1 0

[Kamailio-Users] reseting of t_on_branch, t_on_reply and t_on_failure
by jh＠tutpro.com 09 Mar '09

09 Mar '09

i have noticed that i need to call again t_on_branch, t_on_reply and t_on_failure in failure route before calling again t_relay, i.e., successful call of t_relay resets these settings and they need to be re-armed. how about if t_relay fails and transaction is not created? are the t_on_X calls still in effect or do they need to be re-armed again? i haven't found in tm/README any discussion about when the t_on_X calls get reset. -- juha

1 0

[Kamailio-Users] fork.cfg
by jh＠tutpro.com 09 Mar '09

09 Mar '09

looks like it was forgotten to fix examples/fork.cfg to accommodate tcp transport setup failure. -- juha

1 0

[Kamailio-Users] SIP Re-Routing using Private Header
by Geoffrey Mina 08 Mar '09

08 Mar '09

Hello, I have a problem which I am not sure the best way to solve with Kamailio. I have many asterisk servers which use Kamailio as their outbound gateway to route calls to the PSTN. This works great, I use the LCR engine to control routing. What I want to do is have the ability to dial a random SIP URI from my asterisk servers, but route the call through my Kamailio server for accounting and security purposes. My asterisk servers are not allowed SIP messaging from anything other than my Kamailio gateway. What I am considering doing is something like this: Since asterisk is fairly limited in your ability to route calls, I need to do a little magic to make the call route through a proxy. Maybe I'm wrong, but I haven't yet been able to figure it out. My theory is that I will add a special header at the asterisk level and send the invite to Kamailio. [test] exten => 1,1,SipAddHeader("P-Forward-URI: bob(a)somedomain.com") exten => 1,n,Dial(SIP/forward@kamailio,30) [From Asterisk To Kamailio] INVITE sip:forward@10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: <sip:forward@10.1.1.1> From: "" <sip:5555551212@pc33.atlanta.com>;tag=1928301774 Call-ID: a84b4c76e66710(a)pc33.atlanta.com CSeq: 314159 INVITE Contact: <sip:5555551212@pc33.atlanta.com> Content-Type: application/sdp Content-Length: 142 P-Forward-URL: bob(a)somedomain.com if(is_present_hf("P-Forward-URL")){ //what do i do here to rewrite the To and INVITE parts before doing record_route() and t_relay() } Maybe I'm totally off track here, but this is all I have come up with so far! Perhaps there is a mechanism in SIP which already allows me to do this, and I don't know about it... I don't know what I don't know :) Thanks, Geoff

3 2

[Kamailio-Users] compile problem on debian lenny
by Andreas Heise 07 Mar '09

07 Mar '09

Dear all, I tried to compile trunk version on debian lenny, but it failed. flex, bison and so on are installed. During make the file lex.yy.c is created by flex cfg.lex, but it seems to be that lex.yy.c is compiled twice in the second try lex.yy.c is missing?! Any hint would be great ;-) thx, Andreas kamailio:~/kamailio# make deb prefix=/usr all|more bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex rm -f debian ln -sf packaging/debian dpkg-buildpackage -rfakeroot -tc dpkg-buildpackage: warning: using a gain-root-command while being root dpkg-buildpackage: use CFLAGS from environment: -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls dpkg-buildpackage: set CPPFLAGS to default value: dpkg-buildpackage: use LDFLAGS from environment: -Wl,-O2 -Wl,-E dpkg-buildpackage: set FFLAGS to default value: -g -O2 dpkg-buildpackage: set CXXFLAGS to default value: -g -O2 dpkg-buildpackage: source package kamailio dpkg-buildpackage: source version 1.5.0-svn1 dpkg-buildpackage: source changed by Henning Westerholt <henning.westerholt(a)1und1.de> dpkg-buildpackage: host architecture i386 ... Compiling lex.yy.c gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DNAME='"kamailio"' -DVERSION='"1.6.0-dev0-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/etc/kamailio/"' -DPKG_MALLOC -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DUSE_LOCAL_ROUTE -DF_MALLOC -DSVNREVISION='"2:5668"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -c lex.yy.c -o lex.yy.o gcc: lex.yy.c: No such file or directory gcc: no input files make: *** [lex.yy.o] Error 1 kamailio:~/kamailio# -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01

3 3

[Kamailio-Users] Need help terminating inbound and outbound calls from PSTN!!!!
by carl Lougher 07 Mar '09

07 Mar '09

Hi, I finally managed to get outbound calls working to the PSTN but am struggling with the inbound ones. Current setup: Kamailio 1.5 Sip trunk to Trixbox using username and password stored in Kamailio db PSTN - Gateway only using ip address and no auth. Questions: 1. Am i using the correct method for terminating outbound calls from the Trunk? Currently using rewritehostport off invite. 2. I've tried using allow trusted for the inbound calls from the PSTN GW but they do not connect to the sip trunk. Although i do see the call coming in using ngrep. 3. How do i pass multiple inbound calls to the trunk when it only has one number associated to it in the db? 4. Is there a better method for terminating calls in and out from asterisk/trixbox on sip trunks? cfg file: Last login: Sat Mar 7 14:25:33 2009 from 121-73-118-198.cable.telstraclear.net [root@zedd ~]# cat /usr/local/etc/kamailio/kam kamailio.cfg kamailio.cfg.working kamailio.cfg.working2 kamctlrc [root@zedd ~]# cat /usr/local/etc/kamailio/kamailio.cfg # # $Id: kamailio.cfg 5652 2009-03-02 11:13:38Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's/#m#//g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's/#a#//g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=no children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" #port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp::5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "permissions.so" loadmodule "group.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ #modparam("uri_db", "use_uri_table", 0) #modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", #"mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") #a#modparam("auth_db", "db_url", #a# "mysql://openser:openserrw@localhost/openser") #a#modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ------permissions------- modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://openser:openserrw@localhost/openser") modparam("permissions", "db_mode",1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting if (uri=~"sip:.*$") # Here we check the number dialed { #authorize if a call is going to PSTN if(!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); return; }; xlog("L_INFO", "CALL: Call to international number\n"); rewritehostport("xxx.xx.xx.xx:5060"); # rewriting SIP headers route(1); } if(!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403","Forbidden, use FROM=ID"); exit; }; }; } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } # requests for Media server if(is_method("INVITE") && !has_totag() && uri=~"sip:0[0-9]") { route(4); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; }; } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ # direct to sip provider if (uri =~ "sip:0[0-9]" ) { # route to Asterisk Media Server rewritehostport("xx.xxx.xx.xxx"); route(1); } # force_rport(); # if (nat_uac_test("19")) { # if (method=="REGISTER") { # fix_nated_register(); # } else { # fix_nated_contact(); # } # setflag(5); # } # return; #} # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ #route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE) #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } Thanks!!! Taff.

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[Kamailio-Users] LCR Documentation
by Geoffrey Mina 07 Mar '09

07 Mar '09

Hello, I am currently deploying the LCR module. I have some questions about fields and what they do... I searched the docs, but couldn't find anything. Can someone please fill in the blanks? [LCR Gateways] Gateway Name: description of the gateway Group ID: Numeric identifier used to link LCR Rules IP Address: The IP of the SIP gateway Hostname: The hostname of the SIP gateway (not required if IP provided?) Port: The port the SIP gateway listens on URI Scheme: (well documented) NULL = sip, 1 = sip and 2 = sips Transport: (well documented) NULL = none, 1 = udp, 2 = tcp, 3 = tls, and 4 = sctp Strip: ? Tag: ? Weight: ? Ping: ? Flags: ? [LCR Rules] Prefix: The matching prefix. In the US most would do this by area code, so I would assume I would put something like 1678 or 1770 to route Atlanta calls. Would putting a NULL or Empty String force a default route for everything? >From URI: If I don't care about the from URI can I leave this NULL or BLANK? Group ID: The numeric identifier matching to LCR Gateways Priority: I am assuming if there are 2 records which match the prefix and from URI the high priority will be chosen first? Also, say for example, one of my gateway providers has a primary and a secondary gateway. If the primary is down I need to route calls to the secondary. How would this fit into an LCR scheme? Would I need duplicate routes for the secondary with a lower priority? Thanks, Geoff p.s. for anyone holding out, I just installed Siremis today and it rocks! I am really quite pleased with the solution.

4 10

[Kamailio-Users] controlling inbound calls
by Juan Perez 06 Mar '09

06 Mar '09

hello guys I need to control by IP,port,R-URI who can send calls to our proxy. Looking thru the doc I saw only 1 module that I can use, PERMISSIONS with either Trusted Requests or Address Permissions modes enabled.. Is this the only method or am I missing something? this is for kamailio 1.4.3_no_tls txs a lot jp

3 4

[Serusers] SER and cisco call manager express
by Vittorio Giuffredi 06 Mar '09

06 Mar '09

Hi, I'm relatively new to SER. I'm investigating if a Cisco Router with CME can be connected with a server running SER so I can place a call from my softphone (registered under SER) toward an IP phone that belongs to the Cisco device described above. regards Vittorio Giuffredi ______________________________________________________________________ This email has been scanned by the Candy Group Email Security System. ______________________________________________________________________ Before printing this message, please make sure this is absolutely necessary. Protecting the environment depends on you too! ______________________________________________________________________

1 0

[Kamailio-Users] Normalising
by Brian Chamberlain 06 Mar '09

06 Mar '09

Hi List!, I am trying to write a normalised uri into the radius record for accounting reasons. I have a route[5] that does the transformation for me I translate the uri and store it in a variable. $var(translated_uri) = $ru; $ru = $var(ruri_store); In the accounting module parameters I store Sip-Translated-Request- URI= $var(translated_uri) = $ru; I added the following debug in my route[5] to check the variables are being transformed and they are as I want. xlog("L_INFO", "[TEMP1] Accounting translation: $ru\n"); xlog("L_INFO", "[TEMP2] Accounting translation: $var(translated_uri)\n"); The Sip-Translated-Request-URI is equal to 0 on the radius record.. What am I missing here? I hope I have explained myself well Thanks, Brian

3 4

[Kamailio-Users] Where to add pstn route to .cfg file?
by carl Lougher 06 Mar '09

06 Mar '09

Howdy, I'm getting a little further now but have got stuck at the pstn stage. Now if i delete all the current routes in the kamailio.cfg file and just add a rewritehostport then outbound calls work. Now where exactly do i add this in the .cfg file with all the other routes? I'm not concerned about groups or specfic patterns .* can get sent on. Also for the auth to Trixbox am i better to put as a trusted gw or leave as username/password macth in db? Copy of .cfg file below: [root@zedd kamailio]# cat kamailio.cfg # # $Id: kamailio.cfg 5652 2009-03-02 11:13:38Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's/#m#//g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's/#a#//g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=no children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" #port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:xxx.xxx.xxx.xxx:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ #modparam("uri_db", "use_uri_table", 0) #modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", #"mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") #a#modparam("auth_db", "db_url", #a# "mysql://openser:openserrw@localhost/openser") #a#modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; }; } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ #route[4]{ # force_rport(); # if (nat_uac_test("19")) { # if (method=="REGISTER") { # fix_nated_register(); # } else { # fix_nated_contact(); # } # setflag(5); # } # return; #} # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE) #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }

2 1

[Kamailio-Users] t_uac_dlg syntax
by kévin l'heveder 06 Mar '09

06 Mar '09

Hi, Each time I try to use t_uac_dlg from a management script using MI_DATAGRAM (Kamailio v1.4), Kamailio answers "bad headers". I didn't find out some working examples, so i hope someone could help me. Here's what I send on my UDP socket: ":t_uac_dlg:\nCANCEL\nsip:154@192.168.10.250 <nsip%3A154(a)192.168.10.250>\n.\n.\nFrom: sip:655@192.168.10.250 <sip%3A655(a)192.168.10.250>\nTo: sip:154@192.168.10.250 <sip%3A154(a)192.168.10.250>\n" ( \n.\n. : because i don't specify next hop and the socket ; also i don't specify body) If someone could correct me or give me a working example, that would be very nice. Thanks, Koon

2 1

[Serusers] Ser base billing
by Parikhita Behera 06 Mar '09

06 Mar '09

Dera Sir In the SER based Switch , Billing is possible? If possible , what are the option possible like cdr , monthly trafic , termination status etc.... Waiting for your reply Thanks and Regards Parikhita Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

2 1

[Kamailio-Users] notls vs tls sources
by Luciano Afranllie 06 Mar '09

06 Mar '09

Hi everybody, May be this is a stupid question but I want to know what is the difference between kamailio-1.x.0-notls_src.tar.gz and kamailio-1.x.0-tls_src.tar.gz I guess it is tls support but INSTALL file of notls version talks about compile with TLS support, that is why I am asking... ... - compile with TLS or SCTP support make TLS=1 make SCTP=1 Thanks Luciano

2 1

[Kamailio-Users] handling of DNS-SRV in kamailio-1.4.3 & 1.5.0
by Klaus Feichtinger 06 Mar '09

06 Mar '09

Hello, I have to relay SIP requests to a special "redundant" destination (= FQDN - e.g. "test.kamailio.loc") that consists of 3 hosts with different priorities. After startup Kamailio creates a DNS-SRV request to the DNS server and receives a response with following answer (e.g.): _sip._udp.test.kamailio.loc: type SRV, class IN, priority 10, weight 100, port 5060, target host1.test.kamailio.loc _sip._udp.test.kamailio.loc: type SRV, class IN, priority 20, weight 90, port 5060, target host2.test.kamailio.loc _sip._udp.test.kamailio.loc: type SRV, class IN, priority 30, weight 80, port 5060, target host3.test.kamailio.loc In worst case - if the prime and secondary host (host1 and host2) are unavailable - I expect that kamailio tries relaying the request to the third host in the list. But it doesn't. It makes retransmission of the original invite to the second host and does not try reaching the third host. For tuning the switchover to the alternative target(2) I have set the module parameter "fr_timer" for TM to 3 seconds (modparam("tm", "fr_timer", 3)). Is this behaviour of Kamailio-1.4.3 and Kamailio-1.5.0 (I've tested it with both versions) as expected (= limited to support only two answers in a DNS SRV reply)? Does anybody have experience with this scenario? Can anybody give me a hint? regards, Klaus F. -- Computer Bild Tarifsieger! GMX FreeDSL - Telefonanschluss + DSL für nur 17,95 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a

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[Kamailio-Users] install kamailio (openser) 1.5.0 from svn
by Daniel-Constantin Mierla 06 Mar '09

06 Mar '09

Hello, for those interested in installing kamailio (openser) 1.5.0 from SVN repository, I have created a new wiki page: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn If you find something missing or some error there, please update the page. Hope it's useful! Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Where are going my pua_dialoginfo publishes ?
by Eric PTAK 06 Mar '09

06 Mar '09

Hi, I'm trying to use *_dialoginfo modules but I'm facing to a missunderstanding, or a bug. In fact, the presence server is working and I'm also able to subscribe to dialog event. But when pua_dialoginfo uses pua to send the Publish, I don't know where it going... I even not sure publishes are really sent. This the log : DBG:dialog:run_create_callbacks: dialog=0xb59d30b0 DBG:pua_dialoginfo:__dialog_created: new INVITE dialog created: from=sip:alice@domain DBG:core:grep_aliases: comparing host [0:domain:0] with us [0:domain:0] DBG:core:grep_aliases: match found for: [0:domain:0] DBG:core:check_self: host == me DBG:pua_dialoginfo:build_dialoginfo: new_body: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:alice@domain"> <dialog id="MzQ3N2FiMzhiZGIwNzhhNDhhNjcxN2Y5OGJhY2M0N2Q." call-id="MzQ3N2FiMzhiZGIwNzhhNDhhNjcxN2Y5OGJhY2M0N2Q." direction="initiator"> <state>Trying</state> <remote> <identity>sip:bob@domain</identity> <target uri="sip:bob@domain"/> </remote> <local> <identity>sip:alice@domain</identity> <target uri="sip:alice@domain"/> </local> </dialog> </dialog-info> DBG:pua_dialoginfo:dialog_publish: publish uri= sip:alice@domain DBG:pua_dialoginfo:print_publ: publ: DBG:pua_dialoginfo:print_publ: uri= sip:alice@domain DBG:pua_dialoginfo:print_publ: id= DIALOG_PUBLISH.MzQ3N2FiMzhiZGIwNzhhNDhhNjcxN2Y5OGJhY2M0N2Q. DBG:pua_dialoginfo:print_publ: expires= 0 DBG:pua:send_publish: pres_uri=sip:alice@domain DBG:pua:search_htable: core_hash= 118 DBG:pua:search_htable: record not found DBG:pua:send_publish: insert type DBG:pua:send_publish: UPDATE_TYPE and no record found DBG:pua:send_publish: request for a publish with expires 0 and no record found DBG:dialog:get_dlg_timeout: invalid AVP value, use default timeout DBG:tm:run_reqin_callbacks: trans=0xb59d3270, callback type 1, id 0 entered Regards, Eric.

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[Kamailio-Users] Siremis 0.9.2 with Kamailio 1.5 on Debian 5.0
by mike＠agcusa.com 05 Mar '09

05 Mar '09

3 2

Re: [Kamailio-Users] Newb doco request
by carl Lougher 05 Mar '09

05 Mar '09

Ha ha thanks guys. Who wants the 25cents? Ok i dont have an issue with Asterisk/Trixbox as i've been using that for years. I also dont have an issue with Kamailio installation. The issue i have though is that being a newbie to Kamailio i am struggling to understand and come up with a cfg file to allow the following. 1.To allow Kamailio to accept calls from a specific asterisk/trixbox (authentication would be best). 2.To forward all calls from Kamailio to the Sip provider (how do i set that up? No auth required for sip provider). 3.To apply rtp proxy or nathelper if required. 4.To send cdrs to a mysql database. I've had several attempts at writing stuff to the route part of the cfg file but it always fails to load. Is the best method to find the failure to use /var/log/messages? Thats about it and oh my amazon purchase of "Building Telphony systems with Openser" book turned up today so hopefully reading that might make a difference. Taff.. --- On Thu, 5/3/09, olivier.taylor(a)gmail.com <olivier.taylor(a)gmail.com> wrote: > From: olivier.taylor(a)gmail.com <olivier.taylor(a)gmail.com> > Subject: Re: [Kamailio-Users] Newb doco request > To: "Henning Westerholt" <henning.westerholt(a)1und1.de> > Cc: users(a)lists.kamailio.org > Date: Thursday, 5 March, 2009, 3:36 PM > > > > > > > Pesos, of course :) > > > > Olivier > > > > Henning Westerholt a écrit : > > On Thursday 05 March 2009, olivier taylor wrote: > > > 25 cents bounty, just to understand what's > his problem ;) > > > > euro- or dollar-cent? ;-) > > Henning > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

2 1

[Kamailio-Users] problem with : arithmetic_operations
by BERGANZ François 05 Mar '09

05 Mar '09

Hello, I need to do arithmetic operations. I could do some test with having looks at : http://kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#arithmetic_operati ons Simple opérations like $var(a)=1+1 works. But, when we have : $var(n)=0; while ($var(j)<$dbr(nb_max=>rows)){ $var(n) = $var(n) + $dbr(nb_max=>[0,$var(j)]); xlog("L_NOTICE", "log5 $var(n) + $dbr(nb_max=>[0,$var(j)]) \n"); ##à give us log5 0+10 where is the problem? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 3

Re: [Kamailio-Users] [SR-Dev] git on sourceforge
by Daniel-Constantin Mierla 05 Mar '09

05 Mar '09

IIRC some devels in the past asked for conversion from svn to gitin kamailio (openser), but at that time SF didn't offer it. Andrei started the discussion of how to maintain the parallel source trees for the two projects. We should get to a decision soon. Cheers, Daniel On 03/02/2009 03:01 PM, Jan Janak wrote: > FYI: > > http://apps.sourceforge.net/trac/sitedocs/wiki/Git > > Jan. > > _______________________________________________ > 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

1 1

[Kamailio-Users] Newb doco request
by carl Lougher 05 Mar '09

05 Mar '09

Hi, I'm after some useful doco to allow me to setup trunks from Trixbox to Kamailio and push all calls to a sip provider. Would possibly need rtpproxy or nathelper as well. Can anyone point me in the right direction as i've been combing the web for days and unable to find anything with good explanations and basic enough for a Kamailio beginner. Thanks, Taff.

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Re: [Kamailio-Users] HOWTO rewrite to: header
by alexander merkulov 04 Mar '09

04 Mar '09

thats work for invite nice, but not work fro cancel even this (very simple form) not change cancel if (method == "BYE" || method == "CANCEL") { # xlog("bye\n");a xlog("L_INFO", "CANCEL - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(t_check_trans()){ xlog("triing to rewrite"); if ( subst('/192/aaa/ig') ) {}; t_relay(); exit; }else{ xlog("no trans"); exit; } 2009/3/4 alexander merkulov <arheops(a)gmail.com> > unfortanly it not work after > location() > known limitation: use non-rewrited message. > > 2009/3/4 Ovidiu Sas <osas(a)voipembedded.com> > > Take a look at textops module: >> http://www.kamailio.net/docs/modules/1.5.x/textops.html#id2467719 >> >> Regards, >> Ovidiu Sas >> >> On Wed, Mar 4, 2009 at 4:20 PM, alexander merkulov <arheops(a)gmail.com> >> wrote: >> > here is correct reqwest and reqwest via kamailio >> > >> > is there any way to rewrite To: field in kamailio >> > because now it nto work >> > >> > correct reqwest(asterisk) >> > RECEIVING FROM: 69.70.173.195:5060 >> > CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 >> > Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK4c97bd4a >> > From: "Unknown" <sip:Unknown@voip1.bravotelecom.com<sip%3AUnknown(a)voip1.bravotelecom.com> >> >;tag=as366a0ba0 >> > To: <sip:34249@193.110.78.12:8484> >> > Call-ID: 322db75c71be4ba45a61c7666b93e6be(a)voip1.bravotelecom.com >> > CSeq: 102 CANCEL >> > User-Agent: Asterisk PBX >> > Max-Forwards: 70 >> > Content-Length: 0 >> > >> > incorrect reqwest(multi-homed kamailio, asterisk originate to internal >> eth) >> > >> > RECEIVING FROM: 69.70.173.195:5060 >> > CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 >> > Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK1b1b.3c42cea6.0 >> > From: "Unknown" <sip:Unknown@69.70.173.195<sip%3AUnknown(a)69.70.173.195> >> >;tag=as6075df60 >> > Call-ID: 4d784fa972c4e9c9795032417f46cb26(a)voip1.bravotelecom.com >> > To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> >> > CSeq: 102 CANCEL >> > Max-Forwards: 70 >> > asterisk >> > Content-Length: 0 >> > >> > -- >> > Merkulov Alexander >> > >> > _______________________________________________ >> > Kamailio (OpenSER) - Users mailing list >> > Users(a)lists.kamailio.org >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> > >> > > > > -- > Merkulov Alexander > -- Merkulov Alexander

3 2

[Kamailio-Users] Redundancy and fault-tolerance
by Uriel Rozenbaum 04 Mar '09

04 Mar '09

Hi gentlemen, I'm trying to find a suitable topology to deploy on our SIP network. We started testing UCARP, but it seems it's not going to do for us. Our topology involves 2 servers with physical IPs on them and 2 floating IPs (to balance load 50% on each). The balancing is manual, we are going to point half our servers (measured in load) to one floating IP and so on. When a server fails, the other will handle the load assuming the other floating IP. We found an issue managing IPs, if we have to make Kamailio listen to a floating IP we have to set: listen=0.0.0.0 and then record_route will just pass that IP on the record route headers, making destination servers not to respond. If we try to use record_route_preset() there, when server A fails, all responses come with the "wrong" IP on record_route when server B assumes A's IP. I'd like to know if some of you found a tidier solution and would like to share some topology and implementation tips. DNS SRV is also on our list, but maybe there is something out there that can help us with this. Thanks in advance, Uriel

6 32

[Kamailio-Users] How to send a to-tag in a CANCEL request generated by carrierroute?
by mayamatakeshi 04 Mar '09

04 Mar '09

Hello, I'm using the module carrierroute. I'm setting the carrierfailureroute table to try a second route if the first one times out. The first route is a GW that sends 183 Ringing with SDP and to-tag. But when the failure_route is called due to timeout and the second route is resolved I see kamailio sending a CANCEL without a to-tag and the GW replies with "481 Call Leg/Transaction Does No Exist". So, This fails to cancel the first request. How can I make kamailio send the to-tag in the CANCEL request? Also, I noticed kamailio sends the INVITE to the second route before sending the CANCEL to the first one. Was this coded on purpose? (I believe this doesn't go against the RFC, but it is something curious). If you bother to look at my cfg, here's the relevant parts: route { .........some stuff ..... DOMAIN_AVP = ""; if (!cr_route("CARRIER_AVP", "DOMAIN_AVP", "$rU", "$rU", "call_id", "HOST_AVP")) { sl_send_reply("404", "No Route"); exit; } t_on_failure("ROUTE_FIRST_ATTEMPT_FAILURE"); if (subst_uri('/(sip:.*);nat=yes/\1/')) { setbflag(NAT_BFLAG); } if (isbflagset(NAT_BFLAG)) { fix_nated_contact(); if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { force_rtp_proxy("cfr"); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } t_on_reply("ONREPLY"); if (!t_relay()) { sl_reply_error(); } } failure_route[ROUTE_FIRST_ATTEMPT_FAILURE] { if (isbflagset(NAT_BFLAG)) { unforce_rtp_proxy(); } revert_uri(); if (!cr_next_domain("CARRIER_AVP", "DOMAIN_AVP", "$rU", "HOST_AVP", "$T_reply_code", "DOMAIN_AVP")) { exit; } if (!cr_route("CARRIER_AVP", "DOMAIN_AVP", "$rU", "$rU", "call_id", "HOST_AVP")) { exit; } t_on_failure("ROUTE_SECOND_ATTEMPT_FAILURE"); append_branch(); route(RELAY_REQUEST); } failure_route[ROUTE_SECOND_ATTEMPT_FAILURE] { if (isbflagset(NAT_BFLAG) ) { unforce_rtp_proxy(); } }

2 2

[Kamailio-Users] htable question
by Klaus Darilion 04 Mar '09

04 Mar '09

Hi! Sorry for not knowing how hash table works, thus my questions may be a bit stupid: How many items can be stored in hashtable? Is it limited to "size" parameter, e.g. 10 means max. 1024 entries? Or is it as much as memory is available (what for is the size parameter in this case)? How can I delete a key from htable? Example: I want to track concurrent calls (lets pretend there is no dialog module :-): # pseudo language if INVITE $sht(a=>$ci) = $ts; ... elseif BYE $avp(s:duration) = $ts - $avp(s:duration); # how to delete this key now from the htable? Is it possible to iterate over all entries in the htable? thanks klaus

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Re: [Serusers] Need easiest way to set/clear a flag and refer to it later in same call session
by Miklos Tirpak 04 Mar '09

04 Mar '09

Please read the getting started documentation from http://www.iptel.org/doc. This is a bit old, and I strongly suggest to use ser 2.x instead of 0.9.x, therefore some additional documents might be also useful: http://www.iptel.org/ser/doc/010whatsnew. Regards, Miklos p.s. Please post all the questions/answers to the serusers list, others might be interested as well. On 03/03/2009 08:35 PM, Martin Lane wrote: > Where can I learn the syntax or regexp format of the ser config file? > > On Thu, Feb 26, 2009 at 3:50 AM, Miklos Tirpak <miklos(a)iptel.org> wrote: > >> Hello, >> >> for simple and very fast boolean checks you can use the flags: >> >> - declare the flag(s) at the beginning of the config: >> >> flags >> FLAG_FOO, >> FLAG_BAR; >> >> - Call the setflag(FLAG_FOO) and resetflag(FLAG_FOO) functions to set and >> clear the flag when the INVITE is processed. >> - Use "if (isflagset(FLAG_FOO)) {...}" to check the state of the flag in >> the reply route. >> >> For storing integer or string values you can use AVPs: >> >> # during INVITE processing >> $foo = "myvalue"; >> $bar = 2; >> >> # in reply_route: >> if ($foo == "myvalue") {...} >> if ($bar) {...} >> >> Have a look at etc/ser-oob.cfg for examples. The values are attached to the >> transaction in both cases, therefore you need to forward the request with tm >> module. >> >> Miklos >> >> >> On 02/24/2009 06:40 AM, Frank Durda IV wrote: >> >>> I need a simple way to set either a boolean flag (or integer >>> value) during the INVITE of a given call in ser.cfg, and test >>> that value via "if" elsehere in ser.cfg during the 183/200 >>> response messages for that same call. >>> This would be a data object unique to a given call, >>> which should disappear/lose persistence/end scope when >>> a given call ends. Not global, no need to write the >>> value to disk, doesn't need to survive a restart of SER, etc. >>> >>> The various SER documents describe a number of things that >>> could do this, but most seem overly complex, storing the object >>> in a MYSQL database on disk or something, and these all seem >>> so excessive. There also appear to be a few built-in flags, >>> but the documentation suggests that all are actually there >>> for specific uses and so use for other things might not work. >>> >>> I simply want to make a determination of something during >>> the INVITE for call 12345, and be able to refer to that >>> flag or integer value later on when the various response >>> messages for the same call (#12345) are coming back >>> through. I can't make the same determination on each >>> message after the INVITE because the original src_ip >>> address value where the INVITE came from doesn't seem >>> to be in a testable variable when handling reply >>> messages, or at least not in one that is clearly >>> documented. >>> >>> >>> I've chased a number of possible ways this might be done, >>> but tire of being referred away from the doc that came with >>> the source and over to web sites that tell me this or that module >>> that might do part of what I need is deprecated and to use >>> this other module, but the web site for the other module says >>> that the first module is better suited to what I want to >>> do (a circle), doesn't show how the set value can actually >>> be tested, or the web page just doesn't work anymore. >>> >>> So, assuming there is a simple way to do this, I need a >>> quick code snip showing how to set/clear/assign a value to such >>> a beast within INVITE handling, and a quick code snip of how >>> to test it in an "if" statement during 183/200/110 type >>> response messages. Whatever it is, the value set, cleared >>> or being tested in one call should not affect any other >>> call. >>> >>> >>> Thanks in advance! >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >

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[Kamailio-Users] how to suppr a SDP param
by BERGANZ François 04 Mar '09

04 Mar '09

Hello, In my SDP, I have Media Attribute (a): silenceSupp:off - - - - How can I suppr it ?because, I have a phone wich have error when it have that! Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] Trixbox>openser is this possible?
by carl Lougher 04 Mar '09

04 Mar '09

Howdy, Is it possible to connect a trixbox trunk to openser and then flick the cdr to another box and allow rtp streams to go direct from customer trixbox to sip provider? What cdr billing methods are best from openser? I would like to dump in a mysql database if poss? Thanks, Taff..

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[Kamailio-Users] Kamailio (OpenSER) v1.5.0 Released
by Daniel-Constantin Mierla 03 Mar '09

03 Mar '09

Hello, Kamailio (OpenSER) 1.5.0 has been released. It represent the major version with the biggest new features boost since project was started back in 2005. Release 1.5.0 comes after a shorter development period than usual - about five months of coding and more than one month of extensive testing, but brings lot of new features and improvements, with special care to stability, performance, security and flexibility. A short summary of what the release brings out: * configuration file caching system - generic storage engine in memory via htable module allowing to share data between processes and reduce SQL operations by storing results in cache * execute SQL queries against multiple databases and access the results in a fast and natural fashion * execute config file route blocks on timer * send SIP requests from config file * perform HTTP queries from config file - integration with web service * dialog info presence support - RFC4235 * support for TLS server_name extension (aka SNI=server name indication) * support for non-blocking logging * many enhancements to least cost routing (LCR), load balancing (dispatcher) and carrier routing (carrierroute) * access in config file to both request or reply during transaction processing * perl-like regular expression support (PCRE library) * sanity check of SIP message structure * protection against SIP reply attack * twelve new modules * code restructuring in core - slimmer and less exposed to issues * web management interface - SIREMIS - http://siremis.asipto.com You can download the tarball of the released sources at * http://www.kamailio.org/pub/kamailio/latest/src/ Binary packages for several distributions you can find at * http://www.kamailio.org/pub/kamailio/latest/bin/ * http://www.kamailio.org/pub/kamailio/latest/packages/ Packages will be uploaded as soon as they are submitted. Documentation: * modules' documentation: http://www.kamailio.org/docs/modules/1.5.x/ * migration guide: http://www.kamailio.org/dokuwiki/doku.php/install:1.4.x-to-1.5.0 Full release notes at: * http://www.kamailio.org/mos/view/Kamailio-OpenSER-v1.5.0-Release-Notes Many thanks to all that made this release possible: the management board, developers, testers and community. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Carrierrout issue
by Ali Jawad 03 Mar '09

03 Mar '09

Hi I do use carrieroute to balance calls to 3 servers and those 3 servers do use a2billing and they forward calls to my pstn gw. I do have a graph that has the active calls on them randomly and it does not depend on the load at hand the calls spike up on all 3 servers together. And most of those calls are in ringing state but with no callerID. Normally in the log for carrieroute I have something like this: Mar 3 13:39:10 [4288] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxx ACC: transaction answered: timestamp=1236087551;method=INVITE;from_tag=rMciw2o;to_tag=as7761f5cf;ca ll_id=c745983c;code=200;reason=OK ACC: request acknowledged: timestamp=1236087551;method=ACK;from_tag=rMciw2o;to_tag=as7761f5cf;call_ id=c745983c;code=200;reason=OK Mar 3 13:39:11 [4286] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxxx Mar 3 13:39:12 [4287] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxx ACC: transaction answered: timestamp=1236087552;method=INVITE;from_tag=yJ+hSWA;to_tag=as26649b96;ca ll_id=a1c42904;code=183;reason=Session Progress ACC: transaction answered: timestamp=1236087552;method=INVITE;from_tag=CAJpAXg;to_tag=as5d24ff4f;ca ll_id=a35563d5;code=183;reason=Session Progress ACC: transaction answered: timestamp=1236087552;method=INVITE;from_tag=7i2kutG;to_tag=as6c125d0e;ca ll_id=2739b561;code=183;reason=Session Progress Mar 3 13:39:12 [4285] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxxxxxx ACC: transaction answered: timestamp=1236087553;method=INVITE;from_tag=tMs9y8M;to_tag=as18b7f087;ca ll_id=7d65d34b;code=183;reason=Session Progress However when this problem happens the log screen only has the something like the below and as said the calls are in ringing state with no caller ID: Mar 3 13:39:11 [4286] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxxx Mar 3 13:39:12 [4287] INFO:carrierroute:cr_do_route: uri xxxxxxxxxxxx was rewritten to sip:xxxxxxxxxxxxxx

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[Kamailio-Users] preparing 1.5.0 release
by Daniel-Constantin Mierla 03 Mar '09

03 Mar '09

Hello, preparations for 1.5.0 release starts. Any developer that has to commit any code, please do it quickly. Branch will be created very soon. Many thanks for those helping with development and testing. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] 1.5.0 is awesome.
by Alex Balashov 03 Mar '09

03 Mar '09

Thanks, everyone! Very useful and excellent new features. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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[Kamailio-Users] sqlops problem
by BERGANZ François 03 Mar '09

03 Mar '09

I follow the module instructions and I do : sql_query("ca", "select server from servers where etat_ping=oui & etat_asterisk=oui", "nb_srv"); $var(i) = 0; $var(s)=$dbr(nb_srv=>[server,$var(i)]); ### line 166 I have that error: ERROR:sqlops:sql_parse_index: bad number <server> ERROR:sqlops:pv_parse_dbr_name: invalid index in [[server,$var(i)]] ERROR:core:pv_parse_spec: pvar "dbr" has an invalid name param [nb_srv=>[server,$var(i)]] ERROR:core:pv_parse_spec: wrong char [)/41] in [$dbr(nb_srv=>[server,$var(i)])] at [29 (5)] CRITICAL:core:yyerror: parse error in config file, line 166, column 110-140: unknown script variable Anyone know why? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] How to change request uri with Header value?
by Ramu 02 Mar '09

02 Mar '09

Hi All, I have situation where I need to change the request uri with To header value. Scenario: OpenServ will receive INVITE as follows INVITE sip:2000@open-ser-ip SIP/2.0. To: "2000"<sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com>>. Now I would like to change the request uri as follows: INVITE sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com> SIP/2.0. To: "2000"<sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com>>. and will do a lookup location. Ramu

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Re: [Kamailio-Users] Handling CANCEL and 487
by Daniel-Constantin Mierla 02 Mar '09

02 Mar '09

Hello, ok. I am cc-ing to mailing list to close the discussion thread. The issue was actually caused by a mis-configuration. Cheers, Daniel On 03/02/2009 05:53 PM, Juan Perez wrote: > Thank you Daniel, I applied that code and it is working now. > I appreciate it a lot. > cheers > jp > > ------------------------------------------------------------------------ > *From:* Juan Perez <jperezsip2008(a)yahoo.com> > *To:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *Sent:* Friday, February 27, 2009 1:27:55 PM > *Subject:* Re: [Kamailio-Users] Handling CANCEL and 487 > > thank you a lot, I will do it and let you know > question: if the call is not cancelled, after the 200 OK there is an > ACK, why that ACK does not have any prob? > > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *To:* Juan Perez <jperezsip2008(a)yahoo.com> > *Sent:* Friday, February 27, 2009 11:36:17 AM > *Subject:* Re: [Kamailio-Users] Handling CANCEL and 487 > > > > On 02/27/2009 06:32 PM, Juan Perez wrote: > > thank you a lot, I am attaching the config file > in the main route block you have nothing that process the ack, so it > gets drop. > > Try to add: > > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > t_relay(); > exit; > } else { > exit; > } > } > > Daniel > -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] fix_nated_contact onreply
by samuel 02 Mar '09

02 Mar '09

I have seen lots of default config files where in the reply route only after checking the message (client_nat_test(1)) fix_nated is called. Why is not called when the NAT flag is set upong lookup_XX? In ser-oob I think the reply_route should include the case of a user called behind a NAT and the reply is not fixed due to some router in the middle. Will it hurt including fix_nated_contact in the case of checking the flag? Friday thoughts... Samuel

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Re: [Kamailio-Users] how to know if my gw servers are active ?
by BERGANZ François 02 Mar '09

02 Mar '09

For the database, asterisk and SER work on the sameone. in sip.conf create one for SER and insecure=invite. It accept all from SER In the SER, configure the auth_db module to check the username col and password col. SER is a wall between client and asterisk(s). The same SER can be a wall between asterisk and provider. It is just a gymnastic! To test, it, the best is to have all in the same UC. And have one virtual networkInterface per Asterisk and SER. Cordialement, BERGANZ François http://www.acropolistelecom.net  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Abdul Hakeem [mailto:alhakeem100@hotmail.com] Envoyé : lundi 2 mars 2009 12:14 À : 'BERGANZ François' Objet : RE: [Kamailio-Users] how to know if my gw servers are active ? Hi, Thanks for the prompt reply. My comments are below: -----Original Message----- From: BERGANZ François [mailto:francois@acropolistelecom.net] Sent: 02 March 2009 07:59 To: alhakeem(a)ipextelecom.net Cc: users(a)lists.kamailio.org Subject: RE: [Kamailio-Users] how to know if my gw servers are active ? Abdul Hakeem, I use Asterisk-SER for another way. SER authenticate users and asterisk(s) accept all SIP frames from the SER. >>How did you create the users databases on Asterisk and Ser ? Also, how did you configure Asterisk to forward registration requests to Ser ? If you could send me sample Asterisk/Ser configuration, I'll be much grateful. Then, asterisk(s) forward the call to the provider. When a call come from the provider, asterisk forward it to the SER and the SER just do a lookup(location)! >>What happens if a call comes in to SER directly, & it has to do a quick lookup to see which Asterisk the user is currently located ? The reason I ask is because for example User-A with e164 number/DID +447931800952 might be registered to Asterisk-X now and 10 minutes later register with Asterisk-Y, Did you implement some refresh mechanism on the SER database ?. Because in this case inbound IP calls are hitting SER first before being redirected to the appropriate Asterisk server. I look forward to your reply. Regards, Abdul Hakeem Cordialement, BERGANZ François ? Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Abdul Hakeem [mailto:alhakeem@gmail.com] Envoyé : samedi 28 février 2009 09:41 À : 'BERGANZ François' Cc : users(a)lists.kamailio.org Objet : RE: [Kamailio-Users] how to know if my gw servers are active ? Hello, How did you setup your Asterisk to update SER/OpenSER/Kamalito's database ? I am trying to setup a solution for 2 Asterisk or more servers to allow dynamic and not static user registration. i.e. any user can register with any of the Asterisk servers. Let's say if UserA register with Asterisk1, this will reflect on SER's database where to route the call. In this scenario of UserA now register with Asterisk4 5 minutes later, SER's database will be automatically updated and calls to UserA will be automatically routed to Asterisk4. What I am hoping to achieve is not to do this via manual intervention. Any this will be much appreciated. Cheers, Abdul Hakeem -----Original Message----- From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of BERGANZ François Sent: 27 February 2009 10:24 To: 'Daniel-Constantin Mierla' Cc: users(a)lists.kamailio.org Subject: Re: [Kamailio-Users] how to know if my gw servers are active ? Yes, my asterisk update a database to help the SER Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -----Message d'origine----- De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] Envoyé : vendredi 27 février 2009 11:19 À : BERGANZ François Cc : 'Henning Westerholt'; users(a)lists.kamailio.org Objet : Re: [Kamailio-Users] how to know if my gw servers are active ? On 02/27/2009 12:10 PM, BERGANZ François wrote: > I need to dispatch to the asterisk which have less users connected > How do you get the asterisk with the less users connected? You have some data in a db table? Cheers, Daniel > > Cordialement, > BERGANZ François > P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > > -----Message d'origine----- > De : Henning Westerholt [mailto:henning.westerholt@1und1.de] > Envoyé : vendredi 27 février 2009 11:07 À : BERGANZ François Cc : > users(a)lists.kamailio.org Objet : Re: [Kamailio-Users] how to know if > my gw servers are active ? > > On Friday 27 February 2009, BERGANZ François wrote: > >> If I cant use dispatcher module? >> Because, I don’t use the module algo to choose the dst. >> > > Well, then perhaps you need to use another solution, or adapt the dispatcher > code to your needs. May i ask what is the problem with existing algorithms? > What exactly do you want to achive? > > Cheers, > > Henning > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] how to know if my gw servers are active ?
by BERGANZ François 02 Mar '09

02 Mar '09

hello, I have a list of servers in my database. I need that openser send OPTIONS to servers. for example, I could have a col 'active' to know if each server is active or not. In fact, I need to do an other style of dispatch, and, I cant use the dispatcher module! is there a solution for that ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] CANCEL over nat
by alexander merkulov 01 Mar '09

01 Mar '09

i have async rtp proxy setup on multihomed host. config for transfer cancel and invites: if(lookup("location")){ log("found loc"); fix_nated_contact(); $avp(i:11)=$fU; $var(fr)="sip:"+$avp(i:11)+"@69.70.173.195"; xlog("replace $var(fr)"); if (method == "BYE" || method == "CANCEL"){ unforce_rtp_proxy(); # force_rtp_proxy("AIEOC"); }else{ uac_replace_from("$avp(i:11)","$var(fr)"); if(force_rtp_proxy("FAIEOC")) t_on_reply("2"); } route(1); exit(); } cancel is routed ok, ATA get cancel in form of (ip changed): RECEIVING FROM: 69.70.xxx.xx:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK6e35.c727f225.0 From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as704e6b0d Call-ID: 735a38dd3d77f4f20f29f28b335f11f4(a)192.168.1.180 To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> CSeq: 102 CANCEL Max-Forwards: 70 asterisk Content-Length: 0 but ignore it. same with any other ATA type(including softphones) can u please explaine me what is incorrect in CANCEL? cancel produced by asterisk.CALL-id is correct. here is corespondoing invite 16:06:56.0 RECEIVING FROM: 69.70.173.195:5060 INVITE sip:34249@193.110.78.12:8484 SIP/2.0 Record-Route: <sip:69.70.173.195;r2=on;lr=on;ftag=as704e6b0d;last_from=bWVyYWxtZXJhbG1ldWFwdGRqaHVvZXJgaQ--> Record-Route: <sip:192.168.2.170;r2=on;lr=on;ftag=as704e6b0d;last_from=bWVyYWxtZXJhbG1ldWFwdGRqaHVvZXJgaQ--> Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK6e35.c727f225.0 Via: SIP/2.0/UDP 192.168.1.180:5060;rport=5060;branch=z9hG4bK49fa4d6d From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as704e6b0d To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> Contact: <sip:Unknown@192.168.1.180:5060> Call-ID: 735a38dd3d77f4f20f29f28b335f11f4(a)192.168.1.180 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Sun, 01 Mar 2009 14:05:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 447 v=0 o=root 8300 8300 IN IP4 69.70.173.195 s=session c=IN IP4 69.70.173.195 t=0 0 m=audio 10336 RTP/AVP 0 8 110 97 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes -- Merkulov Alexander

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[Kamailio-Users] dial the prefix #31# to mask the number
by ALAEDDINE abbech 01 Mar '09

01 Mar '09

Hi When i dial #31# 011111with a sip phone i have an INVITE in OPENSER with a number %2331%23 011111. Who know how can i send an INVITE whith prefix #31# using a phone or softphone. thanks,

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Re: [Kamailio-Users] [OpenSIPS-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
by Antonio Reale 27 Feb '09

27 Feb '09

Sorry for the delay. I did not read your reply. Nice solution. I need to check that IOS based pbx can read a custom header. Regards. Antonio. > -----Original Message----- > From: users-bounces(a)lists.opensips.org [mailto:users- > bounces(a)lists.opensips.org] On Behalf Of Iñaki Baz Castillo > Sent: Saturday, February 21, 2009 3:42 AM > To: users(a)lists.opensips.org > Subject: Re: [OpenSIPS-Users] [OpenSER-Users] [OT] How to handle different > DID's in incoming calls for a registered client? > > El Viernes, 20 de Febrero de 2009, Antonio Reale escribió: > > Hi Iñaki, > > did you find a way to solve this problem. I have the same problem with > > devices that can't read the To: value. > > For customers with 10/100 assigned number I routed the entire root number > > to the customer's IP (loosing auth...), but for users with only one > > supplementary number it is not practicable. > > Has someone a suggestion for this problem? > > Thanks to all. > > Regards. > > In the proxy I add a custom header: > P-Called-Number: +34123123123 > so the pbx receiving this INVITE can read that header. > > -- > Iñaki Baz Castillo > > _______________________________________________ > Users mailing list > Users(a)lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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[OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
by Iñaki Baz Castillo 27 Feb '09

27 Feb '09

Hi, maybe this question is a bit off-topic so I'm sorry for that. My question is about SIP providers using OpenSer that associate PSTN numbers to their local clients (SIP accounts): Usually the client must register to OpenSer in order to receive calls. Then it will appear in location table with "Contact=sip:clientX@IP". Suppose clientX has two PSTN numbers associated in a ENUM entry: +34999000111 +34999000222 When anyone in PSTN world calls to +34999000222 the call will arrive to the OpenSer from a gateway in an INVITE like: INVITE sip:+34999000222@gateway SIP/2.0 To: <sip:+34999000222@gateway> The OpenSer will do the lookup in location table and finally send this INVITE to the clientX: INVITE sip:clientX@IP_clientX SIP/2.0 To: <sip:+34999000222@gateway> The info about the called PSTN number is just available in "To" header, so a way to get different behaviour for each associated PSTN number is matching "To" URI. Is common to do it? which other alternatives are there? Thanks for any comment. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [Kamailio-Users] ACC with mysql
by Daniel-Constantin Mierla 27 Feb '09

27 Feb '09

According to the config, you do not even set the accounting flag for ACK, or I missed something there? Can you compile with memory debug turned on and test? Watch the syslog for critical errors. It can turn to be a buffer overflow somewhere. Once again, what is the version are you using? Cheers, Daniel On 02/27/2009 01:45 PM, Uriel Rozenbaum wrote: > Here it is > > On Fri, Feb 27, 2009 at 9:31 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 02/27/2009 12:57 PM, Uriel Rozenbaum wrote: > > I´ve checked the captures again and all ACK messages have the > RPID, for 487 or 200 messages. > > ok, but is the ack you sent causing the trouble? It does not look > as an ack for 200ok, because it does not have a Route header. > Non-200ok ACK do not get accounted. > > > I'm just using a plain build of kamailio downloaded from the > site including carrerroute module. > > Would it be useful if I sent the .cfg? > > > Could help, yes. > > Daniel > > > On Fri, Feb 27, 2009 at 8:34 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com> > <mailto:miconda@gmail.com <mailto:miconda@gmail.com>>> wrote: > > I tried to reproduce, but no success. The crash is > indicated when > processing the lumps, a very old piece of code, so I guess > there > is somewhere a buffer overlow. > > Are you having some private modules that are you using the > config > file? > > Thanks, > Daniel > > > > On 02/27/2009 11:39 AM, Daniel-Constantin Mierla wrote: > > Hello, > > is this of an ACK due to 200ok? Are you doing > record-routing? > Because Route header is missing. > > If you don't do record-routing, then the ACK should be > end-to-end between caller and callee and you cannot > account. > > Anyway, the reported issue should be fixed. > > Cheers, > Daniel > > > On 02/24/2009 10:05 PM, Uriel Rozenbaum wrote: > > Daniel, I've been trying a litte longer with the server > and I captured one ACK that definetely has RPID: > > /ACK sip:1054111556446967@cc1int.x.com.ar > <mailto:sip%3A1054111556446967@cc1int.x.com.ar> > <mailto:sip%3A1054111556446967@cc1int.x.com.ar > <mailto:sip%253A1054111556446967@cc1int.x.com.ar>> > <mailto:sip%3A1054111556446967@cc1int.x.com.ar > <mailto:sip%253A1054111556446967@cc1int.x.com.ar> > <mailto:sip%253A1054111556446967@cc1int.x.com.ar > <mailto:sip%25253A1054111556446967@cc1int.x.com.ar>>> SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.200.11:5060;branch=z9hG4bK31a69f48;rport > From: "541160911100" > <sip:541160911100@192.168.200.11 > <mailto:sip%3A541160911100@192.168.200.11> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11>> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11> > <mailto:sip%253A541160911100@192.168.200.11 > <mailto:sip%25253A541160911100@192.168.200.11>>>>;tag=as799b3334 > > To: <sip:1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%3A1054111556446967@cc1int.cpsnet.com.ar> > <mailto:sip%3A1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%253A1054111556446967@cc1int.cpsnet.com.ar>> > <mailto:sip%3A1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%253A1054111556446967@cc1int.cpsnet.com.ar> > > <mailto:sip%253A1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%25253A1054111556446967@cc1int.cpsnet.com.ar>>>>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.90f4 > > > > Contact: <sip:541160911100@192.168.200.11 > <mailto:sip%3A541160911100@192.168.200.11> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11>> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11> > <mailto:sip%253A541160911100@192.168.200.11 > <mailto:sip%25253A541160911100@192.168.200.11>>>> > > Call-ID: > 5d7adcce1399bafd45c194b049479a1b(a)192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11> > > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11>> > > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11> > > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11>>> > CSeq: 102 ACK > User-Agent: Metrotel > Max-Forwards: 70 > Remote-Party-ID: "541160911100" > <sip:541160911100@192.168.200.11 > <mailto:sip%3A541160911100@192.168.200.11> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11>> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11> > <mailto:sip%253A541160911100@192.168.200.11 > <mailto:sip%25253A541160911100@192.168.200.11>>>>;privacy=off;screen=no > > > Content-Length: 0/ > > Hope it helps, > Uriel > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

2 1

[Kamailio-Users] ACC with mysql
by Uriel Rozenbaum 27 Feb '09

27 Feb '09

Hey guys, I'm using kamailio-1.4.3-notls with ACC module on mysql. I'm trying to benchmark it installed on a virtual server (Xen on CentOS) and I have everything going OK for my basic configuration but when I turn on ACC (by flagging transactions) I get the following lines on the log: Feb 19 08:48:57 ops3 /usr/local/sbin/kamailio[4895]: INFO:core:handle_sigs: child process 4899 exited by a signal 11 Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4895]: INFO:core:handle_sigs: core was generated Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4895]: INFO:core:handle_sigs: terminating due to SIGCHLD Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4902]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4903]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4901]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4897]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4904]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4898]: INFO:core:sig_usr: signal 15 received This issue only happens when i account messages using the following lines: if (is_method("INVITE|BYE|CANCEL")) { setflag(1); setflag(2); } and the configuration for the modules are ass follows: # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # Descomentar para guardar parametros adicionales # modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "acc_to_tag_column", "to_tag") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "acc_time_column", "time") modparam("acc", "db_extra", "cseq=$cs;privacy=$dip;reason=$dir;from_uri=$fu;from_username=$fU;from_displayname=$fn;from_domain=$fd;req_username=$rU;req_domain=$rd;rpid=$re;source_ip_addr=$si;source_port=$sp;to_uri=$tu;to_username=$tU;to_displayname=$tn;to_domain=$td;original_uri=$ou;original_uri_username=$oU;original_uri_domain=$od;unix_timestamp=$Ts") Do you think there is something I'm missing on the configuration? I tried to modify parameters on mysql but get the same errors all the time? Thanks in advance! Uriel

3 17

Re: [Kamailio-Users] how to know if the next gw is up?
by Daniel-Constantin Mierla 27 Feb '09

27 Feb '09

Please direct your questions to users mailing lists. Private messages of such type are not honored very fast, they usually end in my very long unread mails pool. See the dispatcher module, maybe it helps you. Thanks, Daniel On 02/27/2009 11:18 AM, BERGANZ François wrote: > Hello, > > I have 1SER and 5 Asterisk. > I need to choose the dst with my algo... > But, I need to have my server list and choose a server which is online. > The best thing could be to have OPTIONS send to my asterisk and write responses in the database! > > Have you an idea? > > > Cordialement, > BERGANZ François >  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > -----Message d'origine----- > De : users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] De la part de Daniel-Constantin Mierla > Envoyé : vendredi 27 février 2009 09:10 > À : c_lougher(a)yahoo.co.uk > Cc : users(a)lists.kamailio.org > Objet : Re: [Kamailio-Users] Kamailio Newb questions > > > > On 02/27/2009 12:07 AM, carl Lougher wrote: > >> So if i just want to hook off the rtp stream between endpoint and sip provider then am i better off to go for a stun server than nathelper/rtpproxy? >> >> > nathelper/rtpproxy does rtp relay on server, so the rtp goes: [caller] > === [rtpproxy] === [callee] > > In case of dealing with symmetric NAT, it is the only feasible way now > to make it work. > > If STUN is used, the rtp goes: [caller] === [callee] . But does not work > for symmetric nat. > > The best is to combine both of them so you get as less as possible RTP > relaying on server. > > Cheers, > Daniel > > >> What are you using rtpproxy for that is different than stun? >> >> >> --- On Thu, 26/2/09, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >> >> >> >>> From: Daniel-Constantin Mierla <miconda(a)gmail.com> >>> Subject: Re: [Kamailio-Users] Kamailio Newb questions >>> To: c_lougher(a)yahoo.co.uk >>> Cc: users(a)lists.kamailio.org >>> Date: Thursday, 26 February, 2009, 12:27 PM >>> On 02/26/2009 01:19 PM, carl Lougher wrote: >>> >>> >>>> Thanks for that. So does it mean by using rtpproxy you >>>> >>>> >>> will therefore carry all the rtp streams through that server >>> yes, that is the role of RTPProxy - to proxy the RTP >>> streams, therefore those go via the server. >>> >>> If you want end-to-end RTP stream, then look at STUN, if >>> the phones are not behind symmetric nat, it can help. >>> >>> >>> >>>> or can it be redirected to the sip provider from the >>>> >>>> >>> endpoint? >>> >>> >>>> Also how do you put the kamailio server in the >>>> >>>> >>> equation? Do you set it up as an external proxy for the >>> clients or do you register the clients to it then just use >>> asterisk for the media/vmail etc? >>> >>> >>>> >>>> >>>> >>> I do everything in kamailio but the media services which i >>> do with asterisk (vmail, ivr, ...) - authentication, >>> registration, call routing is done in kamailio. >>> >>> Cheers, >>> Daniel >>> >>> >>> >>>> --- On Thu, 26/2/09, Daniel-Constantin Mierla >>>> >>>> >>> <miconda(a)gmail.com> wrote: >>> >>> >>>> >>>> >>>> >>>>> From: Daniel-Constantin Mierla >>>>> >>>>> >>> <miconda(a)gmail.com> >>> >>> >>>>> Subject: Re: [Kamailio-Users] Kamailio Newb >>>>> >>>>> >>> questions >>> >>> >>>>> To: c_lougher(a)yahoo.co.uk >>>>> Cc: users(a)lists.kamailio.org >>>>> Date: Thursday, 26 February, 2009, 9:16 AM >>>>> Hello, >>>>> >>>>> On 02/26/2009 12:59 AM, carl Lougher wrote: >>>>> >>>>> >>>>> >>>>>> Howdy, >>>>>> I'm trying to remove the media/rtp streams >>>>>> >>>>>> >>> from an >>> >>> >>>>>> >>>>>> >>>>>> >>>>> asterisk server for natted users so would like to >>>>> >>>>> >>> know if >>> >>> >>>>> this is possible with kamailio. >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> yes it is possible. nathelper+rtpproxy is the >>>>> >>>>> >>> option I use >>> >>> >>>>> and prefer because of flexibility and >>>>> >>>>> >>> performances. You can see an >>> >>> >>>>> example at: >>>>> >>>>> >>>>> >>> http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy >>> >>> >>>>> >>>>> >>>>> >>>>>> Qu's: >>>>>> What is the best option? >>>>>> rtpproxy/mediaproxy? >>>>>> nathelper? >>>>>> >>>>>> If i use kamailio to achieve this does it mean >>>>>> >>>>>> >>> that i >>> >>> >>>>>> >>>>>> >>>>>> >>>>> still have to carry the rtp streams through the >>>>> >>>>> >>> kamailio >>> >>> >>>>> server instead? >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> through the rtpproxy server, which can be located >>>>> >>>>> >>> on same >>> >>> >>>>> or different machine than kamailio. >>>>> >>>>> >>>>> >>>>> >>>>>> Also will i need to change the logon info for >>>>>> >>>>>> >>> the >>> >>> >>>>>> >>>>>> >>>>>> >>>>> clients so they now logon to kamailio then i just >>>>> >>>>> >>> point >>> >>> >>>>> registrar to asterisk? >>>>> >>>>> >>>>> >>>>>> Can i use kamailio for sip trunks to asterisk >>>>>> >>>>>> >>> and >>> >>> >>>>>> >>>>>> >>>>>> >>>>> carry rtp and natted clients media streams rather >>>>> >>>>> >>> than >>> >>> >>>>> register to asterisk? >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> Yes, you can register to kamailio, see registrar >>>>> >>>>> >>> and usrloc >>> >>> >>>>> modules. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> >>>>> >>>>> >>>>>> Many thanks, >>>>>> Taff.. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>> _______________________________________________ >>> >>> >>>>>> Kamailio (OpenSER) - Users mailing list >>>>>> Users(a)lists.kamailio.org >>>>>> >>>>>> >>>>>> >>>>>> >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> >>>>> >>>>> >>>>> >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> -- Daniel-Constantin Mierla >>>>> http://www.asipto.com >>>>> >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (OpenSER) - Users mailing list >>>> Users(a)lists.kamailio.org >>>> >>>> >>>> >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> >>>> >>>> >>>> >>> -- Daniel-Constantin Mierla >>> http://www.asipto.com >>> >>> >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Kamailio Newb questions
by carl Lougher 27 Feb '09

27 Feb '09

Howdy, I'm trying to remove the media/rtp streams from an asterisk server for natted users so would like to know if this is possible with kamailio. Qu's: What is the best option? rtpproxy/mediaproxy? nathelper? If i use kamailio to achieve this does it mean that i still have to carry the rtp streams through the kamailio server instead? Also will i need to change the logon info for the clients so they now logon to kamailio then i just point registrar to asterisk? Can i use kamailio for sip trunks to asterisk and carry rtp and natted clients media streams rather than register to asterisk? Many thanks, Taff..

3 9

[Kamailio-Users] default config file
by Elena-Ramona Modroiu 26 Feb '09

26 Feb '09

Hi, looking at the default config file, we deliver something pretty useless and very insecure. I propose to add to it several features so it becomes really usable: - mysql support - even some use radius for aaa, mysql is for sure used for usrloc. Those using other db drivers can just replace the module - auth against db and check auth username against from/to username to avoid spoofing - accounting - at least to syslog to store more than default data which gives nothing useful right now. Thinking to from/ruri username and domain at least in order to have caller and callee details - usrloc to database - nat traversal support with nathelper and rtpproxy - if rtpproxy is not running, will be just warnings at startup - the first part of the config will have some help of how to configure the environment to start the sip server (e.g., add user, ...) What do you think? Regards, Ramona

5 7

[Serusers] Opensips--Ser
by Parikhita Behera 26 Feb '09

26 Feb '09

Dear Sir I am new to SER . So I want to know , is Opensips is Open source?. Does it need license?. What operating system needed to install the SER. Does SER support totally telephony, pc-pc , pc-phone, calling card , call back , IPPBX etc.. How many systems needed to install to work the complete solution. If possible , you can give me the installation and configuration ducumentation so that I can try to do the complete solution. Waiting for your reply. Thanks and Regards Parikhita Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

2 1

[Serusers] Need easiest way to set/clear a flag and refer to it later in same call session
by Frank Durda IV 26 Feb '09

26 Feb '09

I need a simple way to set either a boolean flag (or integer value) during the INVITE of a given call in ser.cfg, and test that value via "if" elsehere in ser.cfg during the 183/200 response messages for that same call. This would be a data object unique to a given call, which should disappear/lose persistence/end scope when a given call ends. Not global, no need to write the value to disk, doesn't need to survive a restart of SER, etc. The various SER documents describe a number of things that could do this, but most seem overly complex, storing the object in a MYSQL database on disk or something, and these all seem so excessive. There also appear to be a few built-in flags, but the documentation suggests that all are actually there for specific uses and so use for other things might not work. I simply want to make a determination of something during the INVITE for call 12345, and be able to refer to that flag or integer value later on when the various response messages for the same call (#12345) are coming back through. I can't make the same determination on each message after the INVITE because the original src_ip address value where the INVITE came from doesn't seem to be in a testable variable when handling reply messages, or at least not in one that is clearly documented. I've chased a number of possible ways this might be done, but tire of being referred away from the doc that came with the source and over to web sites that tell me this or that module that might do part of what I need is deprecated and to use this other module, but the web site for the other module says that the first module is better suited to what I want to do (a circle), doesn't show how the set value can actually be tested, or the web page just doesn't work anymore. So, assuming there is a simple way to do this, I need a quick code snip showing how to set/clear/assign a value to such a beast within INVITE handling, and a quick code snip of how to test it in an "if" statement during 183/200/110 type response messages. Whatever it is, the value set, cleared or being tested in one call should not affect any other call. Thanks in advance!

2 1

[Serusers] Opensips--Ser
by Parikhita Behera 26 Feb '09

26 Feb '09

Dear Sir I am new to SER . So I want to know , is Opensips is Open source?. Does it need license?.. What operating system needed to install the SER. Does SER support totally telephony, pc-pc , pc-phone, calling card , call back , IPPBX etc.. How many systems needed to install to work the complete solution. If possible , you can give me the installation and configuration ducumentation so that I can try to do the complete solution. Waiting for your reply. Thanks and Regards Parikhita Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

1 0

[Serusers] How to set the query into database and get answers with exec_msg?
by 邱磊 26 Feb '09

26 Feb '09

Hi ,I am a student studying in VoIP and implementing my theory in SER. I try to access mysql database to get some information, but have a problem aboout paramaters of exe_msg. My goal is :count the number of contact addresses that certain user registered, and take this is my solution: exec_msg(' Count="select count(contact) from location where username=\"$SIP_HF_CONTACT\" "' CountResult=`mysql -user -pheslo -e "$Count" ser`; ') What is the result of "CountResult" ? I suppose originally that it will be the the number of the query, but it seems not......... and i try the exec_msg('rm -r test'); but it failed; Dose the function wored? and how to use it,Hhw can I get my answer what I want to get ? Hoping someone can give me some advises ^^. Thannks in advance. 2009-02-26 邱磊

1 0

[Kamailio-Users] Handling CANCEL and 487
by Juan Perez 25 Feb '09

25 Feb '09

Hello I am having a prob handling the CANCEL in this scenario Kamailio 1.4 doing basic load balancing with LCR module. Calling party makes a call, sends it to Kamailio, Kamailio looks up the destination gw and sends the call to the dest IP. I am using statefull mode and loose routing. The calling party makes the call and while it is ringing it hangs up. Kamailio receives the CANCEL and forward it to the dest IP, it receives the 487 from dest IP and pass it back to the calling party. This is when it gets weird. The calling party sends the ACK to the 487 and it looks like the proxy [kamailio] ignores it and keep sending the 487 to the calling party. Calling party IP: 192.168.1.200 kamailio[proxy]: 192.168.1.50 dest IP: 192.168.11 PLease see below a tethereal output and my config file. I would appreciate any help to help me fix this prob. [root@proxy1 ~]# tethereal -i eth1 -S -w /root/call-with-cancel3.pcap udp port 5060 Running as user "root" and group "root". This could be dangerous. Capturing on eth1 0.000000 192.168.1.200 -> 192.168.1.50 SIP/SDP Request: INVITE sip:664050495179280@192.168.1.50, with session description 0.000702 192.168.1.50 -> 192.168.1.200 SIP Status: 100 Giving a try 0.000774 192.168.1.50 -> 192.168.1.11 SIP/SDP Request: INVITE sip:664050495179280@192.168.1.11:5060;transport=udp, with session description 0.044219 192.168.1.11 -> 192.168.1.50 SIP Status: 100 Trying 0.731527 192.168.1.11 -> 192.168.1.50 SIP/SDP Status: 183 Session Progress, with session description 0.731585 192.168.1.50 -> 192.168.1.200 SIP/SDP Status: 183 Session Progress, with session description 1.869161 192.168.1.200 -> 192.168.1.50 SIP Request: CANCEL sip:664050495179280@192.168.1.50 1.869626 192.168.1.50 -> 192.168.1.200 SIP Status: 200 canceling 1.869738 192.168.1.50 -> 192.168.1.11 SIP Request: CANCEL sip:664050495179280@192.168.1.11:5060;transport=udp 1.869942 192.168.1.11 -> 192.168.1.50 SIP Status: 487 Request Terminated 1.869946 192.168.1.11 -> 192.168.1.50 SIP Status: 200 OK 1.870243 192.168.1.50 -> 192.168.1.11 SIP Request: ACK sip:664050495179280@192.168.1.11:5060;transport=udp 1.870273 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 1.871112 192.168.1.200 -> 192.168.1.50 SIP Request: ACK sip:664050495179280@192.168.1.50 2.317041 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 3.317025 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 5.317028 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 9.317036 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 13.317043 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 17.317049 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 21.317057 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 25.317064 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 29.317071 192.168.1.50 -> 192.168.1.200 SIP Status: 487 Request Terminated 23 packets captured This is my routing logic: ####### Routing Logic ######## # main routing logic route{ xlog("L_INFO", "mylog: starting_main_logic.\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_INFO","mylog: Too Many Hops.\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); xlog("L_INFO","mylog: Message too big.\n"); exit; }; if (!method=="REGISTER") { record_route(); xlog("L_INFO","mylog: Recording Route. Method: [$rm].\n"); } if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); xlog("L_INFO","mylog: Loose Route section. Method: [$rm].\n"); route(1); }; if (method=="INVITE") { if (!load_gws()) { sl_send_reply("503", "Unable to load gateways"); exit; } if(!next_gw()){ sl_send_reply("503", "Unable to find a gateway"); exit; } route(1); } #CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } } route[1] { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); if (!t_relay()) { xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); sl_reply_error(); }; exit; }

2 1

Re: [Kamailio-Users] Need to handle an upstream carrier error - Duplicate INVITE Messages
by Geoffrey Mina 25 Feb '09

25 Feb '09

Hello, I have a carrier who provides PSTN gateway services. They have multiple redundant sip gateway devices in their network. The problem occurs when one of their devices starts to have issues. I will receive an INVITE request from both gateways with the same call-id. The problem is that my Kamailio system doesn't detect that I already set a call up for the INVITE once, and forwards the request to another server in the dispatcher list. What I end up with is a call on two asterisk servers, but only one has the actual RTP stream. The BYE request gets routed to the wrong server, and everything just gets screwy. If anyone could provide any hint on how I might be able to deal with this scenario, I would really appreciate it. I have attached my current config file, and the following is a link to a google spreadsheet which shows the SIP trace. http://spreadsheets.google.com/ccc?key=pU5i2J6Ck3b519-_M6Et3cw I have masked my IP addresses for my own sanity. XX.XX.XX.179 - Kamailio SIP Gateway XX.XX.XX.189 - Asterisk1 XX.XX.XX.186 - Asterisk2 Thanks! Geoff

2 5

[Kamailio-Users] openser not distributing to all DNS SRV entries
by Brian 25 Feb '09

25 Feb '09

Hi, I have a SRV setup with three equal priority entries and equal weights. I expected openser (version 1.3.3) to distribute the INVITE requests equally to all 3 hosts, but it seems that all requests are forwarded to only one host. The SRV entries look like this: _sip._udp.mydomain.com. 8805 IN SRV 100 33 5060 proxy1.mydomain.com. _sip._udp.mydomain.com. 8805 IN SRV 100 34 5060 proxy1.mydomain.com. _sip._udp.mydomain.com. 8805 IN SRV 100 33 5060 proxy1.mydomain.com. And my routing for INVITE request looks like this: rewritehost("mydomain.com"); rewriteport(""); if (!t_relay()) { sl_reply_error(); }; exit; Am I wrong in expecting this to distribute equally to the three proxy hosts? I thought this would just "work". Thanks, Brian.

3 3

Re: [Kamailio-Users] event in kamailio
by Marion.DELEY＠fr.thalesgroup.com 25 Feb '09

25 Feb '09

Hi, Finally, I've used the send() function in kamailio.cfg with the @ of my UDP server to know when invite and register requests arrive. Thanks for your help. marion -----Message d'origine----- De : Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Envoyé : mardi 24 février 2009 21:32 À : Marion.DELEY(a)fr.thalesgroup.com Cc : users(a)lists.kamailio.org Objet : Re: [Kamailio-Users] event in kamailio Marion.DELEY(a)fr.thalesgroup.com wrote: > Hi everybody, > > I'm novice in using Kamailio and I need to know if it's possible to create > event from kamailio.cfg to another program. > Actually, I use MI command but I would find a way to force kamailio to send > me messages and not just responses. If the other programm supports SIP or HTTP you can send SIP requests from kamailio.cfg and generate HTTP requests. regards klaus

2 1

[Kamailio-Users] event in kamailio
by Marion.DELEY＠fr.thalesgroup.com 24 Feb '09

24 Feb '09

Hi everybody, I'm novice in using Kamailio and I need to know if it's possible to create event from kamailio.cfg to another program. Actually, I use MI command but I would find a way to force kamailio to send me messages and not just responses. Some help? thanks marion

3 2

[Kamailio-Users] Error compiling CARRIERROUTE module
by Asim Riaz 24 Feb '09

24 Feb '09

Hello, I am trying to compile CARRIERROUTE module, I am getting errors below; Kamailio 1.4 make prefix=/usr modules=modules/carrierroute modules make[1]: Entering directory `/usr/local/src/kamailio/sip-server/modules/carrierroute' Compiling carrierroute.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"carrierroute"' -DNAME='"kamailio"' -DVERSION='"1.4.3-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DSVNREVISION='"2:5452M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c carrierroute.c -o carrierroute.o In file included from carrierroute.c:30: ../../sr_module.h:48:19: error: dlfcn.h: No such file or directory In file included from /usr/lib/gcc/i486-linux-gnu/4.3.2/include-fixed/syslimits.h:7, from /usr/lib/gcc/i486-linux-gnu/4.3.2/include-fixed/limits.h:11, from ../../parser/../flags.h:32, from ../../parser/msg_parser.h:46, from ../../sr_module.h:50, from carrierroute.c:30: /usr/lib/gcc/i486-linux-gnu/4.3.2/include-fixed/limits.h:122:61: error: limits.h: No such file or directory In file included from ../../parser/msg_parser.h:47, from ../../sr_module.h:50, from carrierroute.c:30: ../../parser/../ip_addr.h:38:20: error: string.h: No such file or directory ../../parser/../ip_addr.h:39:23: error: sys/types.h: No such file or directory ../../parser/../ip_addr.h:40:24: error: sys/socket.h: No such file or directory ../../parser/../ip_addr.h:41:24: error: netinet/in.h: No such file or directory ../../parser/../ip_addr.h:42:19: error: netdb.h: No such file or directory In file included from ../../parser/../ip_addr.h:46, from ../../parser/msg_parser.h:47, from ../../sr_module.h:50, from carrierroute.c:30: ../../parser/../dprint.h:54:20: error: syslog.h: No such file or directory ../../parser/../dprint.h:55:18: error: time.h: No such file or directory In file included from ../../parser/../ip_addr.h:46, from ../../parser/msg_parser.h:47, from ../../sr_module.h:50, from carrierroute.c:30: any idea ? Thanks in advance. Asim

1 1

Re: [Kamailio-Users] help me in ldap kamailio
by Henning Westerholt 24 Feb '09

24 Feb '09

On Tuesday 24 February 2009, Husna Bte Mad Baguri wrote: > There is no error msg.just there is no file name ldap.cfg in > usr/local/etc/kamailio.lets me explain how i install. > > 1.i using ubuntu 8.10 > 2.i use kamailio 1.4.3 with tls > 3.i install open ldap --> libldap2-dev,ldap-utils,slapd > 4.then i exclude module =ldap Hi Husna, did i understand you correctly, you excluded the ldap module from compilation? This way it will not be compiled. > 5.then i install.no error but no file ldap.cfg > 6.can you explain what next. It seems that the ldap.cfg file is not installed per default. You find this file in modules/ldap/etc/ in the kamailio 1.4.3 directory. Just copy it to the location you want, and probably adapt it to your setup. Cheers, Henning

1 0

[Kamailio-Users] How can I use domain module?
by Ramu 24 Feb '09

24 Feb '09

Hi, I would like to use domain module to support multiple domain users. How can move forward on this? Thanks, Ramu

5 8

[Kamailio-Users] help me in ldap kamailio
by Husna Bte Mad Baguri 24 Feb '09

24 Feb '09

Can someone help me,i counter this error when try to install ldap in kamailio Compiling ldap_api_fn.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"ldap"' -DNAME='"kamailio"' -DVERSION='"1.4.3-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c ldap_api_fn.c -o ldap_api_fn.o ldap_api_fn.c:38:18: error: ldap.h: No such file or directory In file included from ldap_api_fn.c:40: Plzzzzz help me regards Husna This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

2 1

[Kamailio-Users] help me in ldap kamailio
by Husna Bte Mad Baguri 24 Feb '09

24 Feb '09

Can someone help me,i counter this error when try to install ldap in kamailio Compiling ldap_api_fn.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"ldap"' -DNAME='"kamailio"' -DVERSION='"1.4.3-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.3.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -c ldap_api_fn.c -o ldap_api_fn.o ldap_api_fn.c:38:18: error: ldap.h: No such file or directory In file included from ldap_api_fn.c:40: Plzzzzz help me regards Husna This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

1 1

[Serusers] Cleaning bad npdi=yes and rn=VALUE tags from INVITEs
by Frank Durda IV 24 Feb '09

24 Feb '09

I have a client who occasionally sends calls where there is a npdi=yes tag (and there may or may not be a rn=VALUE tag present with it), BUT the answer these two tags are providing is wrong! That is, the number in question has or hasn't been ported, but the information provided in the INVITE message doesn't match reality. Either a rn=VALUE should have had value X and it instead has value Y, OR there is no rn=VALUE at all, and there should have been one, OR the number hasn't actually been ported, and no rn=VALUE should have been sent at all but we got a junk one, that probably isn't the right LRN for the true destination of the call. Because of these wrong npdi=yes tags, our PSTN gateway switch ends up failing these incorrectly-marked call, because it takes the provided rn=VALUE or lack of one to be correct and so it doesn't do its own LNP query, and the one provided in the SIP INVITE is sometimes just flat out wrong. If the PSTN gateway switch had ignored these suspect npdi=yes and rn=VALUE tages and simply done the LNP query itself, in virtually all cases these otherwise doomed calls would have completed successfully. For this particular customer, the problem of bogus values appears to be a permanent artifact. They either don't know how to fix it, or don't think it is important, of they are spoofing it to avoid being billed for LNP queries. Whatever the reason, it causes trouble tickets. So, we would like to have SER strip the npdi=yes and any rn=VALUE tags from the INVITE when it is this customer (known by src ip), so that the PSTN switch will do the LNP query itself on the SS7 side and get the call routing right. (We have already determined that getting the outfit that is sending the junk to do the right thing isn't going to happen any time soon.) Does anybody have any suggestions as how to write the ser.cfg code to edit-out a npdi=yes and rn=VALUE if they are present in INVITEs? We just want to always throw npdi=yes and any rn=VALUE that shows up if the call is coming from this given customer, basically treating all npdi=yes and rn= tags coming from this customer as being untrustworthy. There has to be an easier way than stripping the thing down and reassembling a new one with repeated strip(1) type operations, which looks to be extremely inefficient for this task and probably really meant only for stripping country codes and such off the phone numbers, not for cleaning up bad tags. Thanks, in advance!

1 0

Re: [Kamailio-Users] LCR in Kamailio 1.4
by Ovidiu Sas 23 Feb '09

23 Feb '09

Make sure that from_uri is NULL and not a white space. Please copy the mailing list in your reply. Regards, Ovidiu Sas On Thu, Feb 19, 2009 at 11:25 AM, Asim Riaz <ariaz78(a)gmail.com> wrote: > > > On Thu, Feb 19, 2009 at 4:16 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: >> >> Is your R-URI starting with a '0'? >> You have defined a single route for all '0' prefix destinations. > > yes R-URI is starting with 0. >> >> >> Regards, >> Ovidiu Sas >> >> On Thu, Feb 19, 2009 at 11:06 AM, Asim Riaz <ariaz78(a)gmail.com> wrote: >> > >> > >> > On Thu, Feb 19, 2009 at 3:53 PM, Daniel-Constantin Mierla >> > <miconda(a)gmail.com> wrote: >> >> >> >> Hello, >> >> >> >> On 02/19/2009 05:34 PM, Asim Riaz wrote: >> >>> >> >>> Hello, >> >>> I am using LCR, for some reason load_gws() function is not finding the >> >>> gateway in DB, it was working in the start then added some more routes >> >>> and >> >>> after that everything stopped working, I am not sure where to start >> >>> troubleshooting, any idea ? >> >> >> >> run kamailio with higher debug mode (e.g., debug=5) in configuration >> >> file. >> >> >> >> You should get more debug messages in the syslog, that may give you >> >> some >> >> hints. look for lines containing 'lcr'. >> >> >> >> Cheers, >> >> Daniel >> > >> > Hi Daniel, >> > I tried higher debug level, not getting anything related to LCR all i >> > am >> > getting XLOG messages "unable to find Gateway". >> > >> > Thanks, >> > >> > Asim >> > >> >> >> >>> >> >>> kamctl lcr show >> >>> >> >>> lcr routes >> >>> +----+--------+----------+--------+----------+ >> >>> | id | prefix | from_uri | grp_id | priority | >> >>> +----+--------+----------+--------+----------+ >> >>> | 9 | 0 | | 1 | 1 | >> >>> +----+--------+----------+--------+----------+ >> >>> lcr gateways >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> | gw_name | ip_addr | port | uri_scheme | transport | grp_id >> >>> | >> >>> strip | tag | flags | >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> | XXXX | XXXX | 5060 | 1 | 1 | 1 | >> >>> 0 >> >>> | | 0 | >> >>> >> >>> >> >>> +------------+---------------+------+------------+-----------+--------+-------+------+-------+ >> >>> >> >>> >> >>> >> >>> my routing script >> >>> >> >>> if(!load_gws()) { >> >>> xlog("L_INFO", "[ROUTE-X] Unable To Find Gateway "); >> >>> sl_send_reply("503", "unable to find gateway"); >> >>> exit; >> >>> } >> >>> Thanks in advance. >> >>> >> >>> Asim >> >>> >> >>> ------------------------------------------------------------------------ >> >>> >> >>> _______________________________________________ >> >>> Kamailio (OpenSER) - Users mailing list >> >>> Users(a)lists.kamailio.org >> >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> >> Daniel-Constantin Mierla >> >> http://www.asipto.com >> >> >> > >> > >> > _______________________________________________ >> > Kamailio (OpenSER) - Users mailing list >> > Users(a)lists.kamailio.org >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> > > >

4 15

[Kamailio-Users] problem with 200ok
by BERGANZ François 23 Feb '09

23 Feb '09

Hello, I have Asterisk1---SER---Asterisk2. When I do INVITE from the left, --the asterisk2 send 200ok to the SER --the SER forward to the Asterisk1 --but the asterisk1 directly send the ACK to Asterisk2 Asterisk2 retransmit the 200ok… and error. I think that it need that the ACK come from the SER and not directly from the Asterisk1. So, how can I detect a 200ok and reply a ACK with my SER? Or, anyone have ever seen that problem? Thank you Next: my capture problem SIP/2.0 183 Session Progress Via: SIP/2.0/UDP IP_SER;branch=z9hG4bK815c.d8703d97.0;received=IP_SER Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653753 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 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 6§¢Iœÿ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653753 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 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 SIP/2.0 200 OK v: SIP/2.0/UDP IP_SER_TO_CLIENTS;branch=z9hG4bK6c33.6ae23971.0 v: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK598106c4;rport=5060 f: "francois berganz"<sip:170725014@IP_ASTERISK>;tag=as1551f6d3 t: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef i: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 INVITE Require: timer x: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,I NFO m: <sip:URI_TEST@192.168.1.82:5060;user=phone> Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold c: application/sdp l: 206 v=0 o=URI_TEST 4319348 4319348 IN IP4 192.168.1.82 s=- c=IN IP4 192.168.1.82 t=0 0 m=audio 41000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 200 OK v: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK598106c4;rport=5060 f: "francois berganz"<sip:170725014@IP_ASTERISK>;tag=as1551f6d3 t: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef i: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 INVITE Require: timer x: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,I NFO m: <sip:URI_TEST@IP_PHONE:5060;user=phone> Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold c: application/sdp l: 206 v=0 o=URI_TEST 4319348 4319348 IN IP4 192.168.1.82 s=- c=IN IP4 192.168.1.82 t=0 0 m=audio 41000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ACK sip:URI_TEST@IP_PHONE:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK0e97e7e8;rport Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Max-Forwards: 70 From: "francois berganz" <sip:170725014@IP_ASTERISK>;tag=as1551f6d3 To: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef Contact: <sip:170725014@IP_ASTERISK> Call-ID: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP IP_SER;branch=z9hG4bK815c.d8703d97.0;received=IP_SER Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653754 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 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 SIP/2.0 200 OK Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK369cf0be;rport=5060 From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:URI_TEST@IP_ASTERISK> Content-Type: application/sdp Content-Length: 287 v=0 o=root 1097653753 1097653754 IN IP4 IP_ASTERISK s=Asterisk PBX 1.6.0.1 c=IN IP4 IP_ASTERISK t=0 0 m=audio 16386 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 ACK sip:URI_TEST@IP_PHONE:5060;user=phone SIP/2.0 Record-Route: <sip:IP_SER_TO_CLIENTS;lr=on;ftag=as1551f6d3> Via: SIP/2.0/UDP IP_SER_TO_CLIENTS;branch=z9hG4bK6c33.6ae23971.2 Via: SIP/2.0/UDP IP_ASTERISK:5060;branch=z9hG4bK0e97e7e8;rport=5060 Max-Forwards: 70 From: "francois berganz" <sip:170725014@IP_ASTERISK>;tag=as1551f6d3 To: <sip:URI_TEST@IP_SER_TO_CLIENTS>;tag=c0a80101-41e5ef Contact: <sip:170725014@IP_ASTERISK> Call-ID: 1593ec3c0a617037319349245be7c5ab@IP_ASTERISK CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 ACK sip:URI_TEST@IP_ASTERISK SIP/2.0 Via: SIP/2.0/UDP IP_PROVIDER_ASTERISK:5060;branch=z9hG4bK7a6d7cf4;rport From: "francois berganz" <sip:170725014@IP_PROVIDER_ASTERISK>;tag=as4a6f8fc6 To: <sip:URI_TEST@IP_SER>;tag=as272dc8d3 Contact: <sip:170725014@IP_PROVIDER_ASTERISK> Call-ID: 5079515a2060a26155be2cbf6b73dc78@IP_PROVIDER_ASTERISK CSeq: 102 ACK User-Agent: MARS Max-Forwards: 70 Content-Length: 0 Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 2

[Kamailio-Users] NAT traversal
by BERGANZ François 23 Feb '09

23 Feb '09

Hello, Is it possible to do NAT traversal without RTP-PROXY ? If yes, have you an example? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

7 18

[Kamailio-Users] how to prepend area code into a local call phone number
by Jinsong Hu 23 Feb '09

23 Feb '09

Hi, Sir: In the US, assume a person from 1-408-1234567 is dialing phone number 1234568. how do I figure out the caller's country code and area code and prepend to the local number ? I tried prefix($(from.user{s.substr, 0, 4})); and openser reported as parser error in config file, bad argument, string expected. Jimmy

2 1

[Kamailio-Users] Using set_advertised_address with variable
by catalina oancea 23 Feb '09

23 Feb '09

Hi Does anybody know if there is a way to use the set_advertised_address function with an avp or variable parameter instead of plain string? I need to change the IP in the Via header with the value of a variable. Thanks Catalina

2 1

[Kamailio-Users] missing file ldap.h
by Husna Bte Mad Baguri 23 Feb '09

23 Feb '09

Dear Whom May Concern, When I try to compile ldap module in kamailio I get error,ldap.h :No such file or directory. Where can I get that file.please someone help me. Regards Husna ________________________________ This e-mail and the attachments are intended solely for the person to whom it has been addressed. It contains privileged and/or confidential information and is privileged or otherwise protected from disclosure. If you are not the person for whom this e-mail was intended, or the e-mail has reached you by mistake, please delete it immediately and inform us of the error. Our e-mail address is administrator(a)uniten.edu.my. All opinions, conclusions and other information in this message that do not relate to the official business of Universiti Tenaga Nasional (UNITEN) shall be understood as neither given nor endorsed by UNITEN. UNITEN shall not be responsible for any activity that may be considered as illegal and/or improper use of e-mail and UNITEN further is claims and shall not accept liability for any content of this e-mail, or for the consequences of any actions taken on the basis of the information provided, unless that information is subsequently confirmed in writing. WARNING Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, arrive late or contain viruses. As such, we do not accept liability for any errors or omissions in the content of this message which may arise as a result of internet transmission. UNITEN does not authorize any of its employees to make any defamatory or seditious statements or commit any offence which is contrary to the laws of Malaysia. Any such communications and/or actions by such employees are outside the scope of employment of the said individuals and UNITEN shall not be liable for such communications and/or actions.

2 1

Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Daniel-Constantin Mierla 20 Feb '09

20 Feb '09

Hello, On 02/13/2009 12:08 PM, Abdul Hakeem wrote: > Hi, > Are you able to put me in touch with the guys using the SDP parser ? > direct your emails to mailing list all the time. Those people are there, the developer of SDP parser and others using it. You will get much faster response and maybe better than writing private emails. Thanks, Daniel > Cheers, > Abdul Hakeem > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: 11 February 2009 15:20 > To: alhakeem(a)ipextelecom.net; kamailio > Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc > > Hello, > > On 02/11/2009 03:20 PM, Abdul Hakeem wrote: > >> Hi, >> >> Just wanted to touch base on the parser to find out if you are still >> pursuing this. >> >> > it is still on my todo list, however didn't make it in the upcoming 1.5.0. > Ovidiu Sas did some commits to the SDP parser and I know couple of others > using it with success. > > Please open a feature request at: > http://sourceforge.net/tracker/?atid=743023&group_id=139143&func=browse > > It is easier to track and not forgot about it for the next release. > > Thanks, > Daniel > > > >> I am in the mind of purchasing some Cisco equipment to do the Sip-MAP >> gateway from a GSM network. >> I have someone else from the OpenBTS with me on this. >> The object is to provide a full fledged sip based routing of voice, >> sms and multimedia calls from gsm handsets to BTS with a Sip server >> > backend. > >> Cheers, >> Abdul Hakeem >> >> -----Original Message----- >> From: users-bounces(a)lists.kamailio.org >> [mailto:users-bounces@lists.kamailio.org] On Behalf Of >> Daniel-Constantin Mierla >> Sent: 03 September 2008 17:45 >> To: Kristian Kielhofner >> Cc: users(a)lists.kamailio.org >> Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc >> >> Hello, >> >> On 09/03/08 19:10, Kristian Kielhofner wrote: >> >> >>> Hello everyone, >>> >>> Feel free to set $OSSPROXY to whatever you like = >>> Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). >>> >>> $OSSPROXY is my favorite SIP toolkit. In addition to being an >>> excellent proxy and registrar it serves as a platform to do SIP >>> mangling and wrangling. With that in mind let me propose a crazy >>> idea... >>> >>> How hard would it be to get $OSSPROXY to be able to parse what is >>> commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has >>> no business getting too involved in the SDP but I feel it makes sense >>> in some instances (like this one). What if I want to make a routing >>> decision based on an encapsulated ISUP parameter of some sort? Who >>> knows what it could be!?!? Maybe there are some other uses but that >>> makes the most sense and seems to be the most practical. It should >>> be possible to develop an $OSSPROXY module to: >>> >>> - Use the SDP parser to find the ISUP SDP part: >>> >>> http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing >>> >>> - Pass that part into a "real" ISUP parser (if necessary). I >>> couldn't seem to find the perfect library but perhaps something like >>> this could >>> work: >>> >>> http://www.openss7.org/isup.html >>> >>> - Make the ISUP parameters available via AVPs, pseudo variables, etc >>> for use in the $OSSPROXY script. >>> >>> I don't really have a need for this but it seems like it could be >>> cool. Am I nuts? >>> >>> >>> >>> >> I would expose the body as a dynamic structure, so components will be >> accessible in the script, e.g. (naming schema just for example), >> >> if($body(0=>content-type)=="application/sdp") >> { >> if($body(0=>m=>type) == "audio") >> { >> } >> } >> >> Based on the available content type parsers (sdp, isup), the structure >> has different attributes. >> >> Cheers, >> Daniel >> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

5 13

[Kamailio-Users] lookup("location") problem
by BERGANZ François 19 Feb '09

19 Feb '09

Hello, I do as the example in the module registrar doc: lookup("location"); switch ($?) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; And SER report me ERROR:core:pv_parse_spec: wrong char [$/36] in [$?] at [0 (0)] Know you why? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

3 3

[Kamailio-Users] call keepalived using rtimer?
by Noel R. Morais 19 Feb '09

19 Feb '09

Hi Guys, I would like to know if is possible to make a keepalive for active calls (dialog module)? Well, I think it is but is there a better way than making a selec into the dialog table and building the INVITEs? Thanks, Noel

6 8

[Kamailio-Users] SIP URI Extracting
by Ramu 19 Feb '09

19 Feb '09

Hi, I would like to extract uri part from SIP Request uri. Say I received a INVITE sip:1234567890@ip Now I would like to take 4567890 the uri and send it as a new INVITE. Ramu

3 2

[Kamailio-Users] LCR in Kamailio 1.4
by Asim Riaz 19 Feb '09

19 Feb '09

Hello, I am using LCR, for some reason load_gws() function is not finding the gateway in DB, it was working in the start then added some more routes and after that everything stopped working, I am not sure where to start troubleshooting, any idea ? kamctl lcr show lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 9 | 0 | | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateways +------------+---------------+------+------------+-----------+--------+-------+------+-------+ | gw_name | ip_addr | port | uri_scheme | transport | grp_id | strip | tag | flags | +------------+---------------+------+------------+-----------+--------+-------+------+-------+ | XXXX | XXXX | 5060 | 1 | 1 | 1 | 0 | | 0 | +------------+---------------+------+------------+-----------+--------+-------+------+-------+ my routing script if(!load_gws()) { xlog("L_INFO", "[ROUTE-X] Unable To Find Gateway "); sl_send_reply("503", "unable to find gateway"); exit; } Thanks in advance. Asim

3 6

[Kamailio-Users] related free voip providers
by Daniel-Constantin Mierla 19 Feb '09

19 Feb '09

Hello, I posted a summary of three free voip providers running sip-router.org related sip servers. It is based on my experience as user of all these (an other) services. The providers listed on the page are not ranked, just presented in alphabetic order, pretty summary about each one. I plan a longer version separately, one by one. It is a good starting point for those willing to see what can be done with Kamailio (OpenSER) and/or SIP Express Router (SER). Another benefit is for users looking to get an independent SIP address, not related to current job or location. I created the post after some private inquires, but I think it worth for public audience: http://miconda.wordpress.com/2009/02/17/top-three-free-voip-services/ In the same spirit, you can check the second edition of (my personal) Kamailio (OpenSER) awards: http://miconda.wordpress.com/2009/01/19/kamailio-openser-2008-awards/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] SIREMIS 0.9.2 released - Support for Carrierroute
by Elena-Ramona Modroiu 19 Feb '09

19 Feb '09

Hi all, a new version of SIREMIS, web management interface for Kamailio (OpenSER) and SIP-Router.org, is out as 0.9.2. Main improvements refer to addition of support to manage the carrierroute module: * management of carrierroute table * management of carrierfailureroute table * management of carrier_name table * management of carrier_domain table View screenshots: http://www.asipto.com/gallery/v/siremis/ Demo site (it works on a database with random data, username: admin, password: admin): http://siremis.asipto.com/demo/ Download: http://siremis.asipto.com/pub/downloads/siremis/ Install/Upgrade documentation: http://siremis.asipto.com/install/ Regards, Ramona

1 0

[Kamailio-Users] Config
by NYam 19 Feb '09

19 Feb '09

Dear Sir, I have problem . wenn i root:~ # kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed root:~ # kamailio -c Feb 17 11:01:49 [10767] ERROR:core:main: loading config file(/usr//etc/kamailio/kamailio.cfg): No such file or directory Feb 17 11:01:49 [10767] WARNING:core:fm_free: free(0) called Feb 17 11:01:49 [10767] WARNING:core:fm_free: free(0) called root:~ # kamailio -f /etc/kamailio/kamailio.cfg then i see udp: 127.0.0.1 [127.0.0.1]:5060 ********************* ********************** tcp: 127.0.0.1 [127.0.0.1]:5060 ok Where is Problem ? can you help me? with kind regards Nyam

3 2

[Kamailio-Users] Kamailio 1.4.3 Carrierroute BYE does not
by unknown＠1.org 19 Feb '09

19 Feb '09

Hello: I have this scenario: I already rad about the bye2bye, the dialog module, i was not able to see any light there. Here is the scenario A--GW--------Kamailio(carrierroute)---------GW1---B |--------------GW2 |--------------GW3 (etc) the capture |<-------------BYE---- -<----NOTHING HERE<<<-| the By does go back if the B user hangs it is not even attempted in the kamailio, per the traces. the BY generates records in the CRD from B side if i hang manually the bye generates CDR records from the A side. Again the Bye is understood by the kamailio and generates the CDR records, that is Not the problem, the Problem is i have the user A in silence, i am talking about the Signaling Only, not Media, no NAT. I even tried the mediaproxy (just trying) the config is like the: http://voipembedded.com/resources/openser_cr.cfg i added if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails exit; } route(10); } else { /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(200); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); xlog("L_WARN", "[$mi] discarding ACK\n"); exit; } this is the main difference for now, i changed to use as a basis example the Example 1.45. Configuration example - Routing to user tree. same results. i tried adding the record_route in the invite also, same result. At this point i have no clue what i am missing. Please let me know if this should work or not, and if yes, based in that basic example, what i should add.

2 8

[Kamailio-Users] A2billing Project <-> Openser
by BERGANZ François 18 Feb '09

18 Feb '09

Hello, I use Asterisk2billing to do prepaidit works fine! But, now, I need to auth my users in the openser and forward the call by loadbal to asterisk(s) which use Asterisk2billing. Anybody know how to do? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

2 14

[Kamailio-Users] SIP-T
by Kaeman Chris 18 Feb '09

18 Feb '09

Hello, Does opensips support SIP-T? Thank you! KC Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

3 3

[Kamailio-Users] SEMS (bugfix) release 1.0.1
by Stefan Sayer 17 Feb '09

17 Feb '09

Hello, The 1.0.1 release of SEMS, the SIP Express Media Server [0], is now available for download [1]. The 1.0.1 release contains all fixes applied to the 1.0 branch, and should be a drop-in replacement for anyone still using 1.0.0. Update is recommended to everyone who is still using 1.0.0-rcX or any later version from the 1.0 branch. Debian packages for etch/64 are also available [2]. Best Stefan Sayer [0] http://iptel.org/sems/ [1] http://ftp.iptel.org/pub/sems/sems-1.0.1.tar.gz [2] http://ftp.iptel.org/pub/sems/1.0/1.0.1/packages/debian Apologies if you receive multiple copies of this message. -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com IPTEGO GmbH Wittenbergplatz 1 10789 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

1 0

[Kamailio-Users] minutes for IRC meeting 17.02.2009, release dates
by Henning Westerholt 17 Feb '09

17 Feb '09

Hi all, the minutes for todays IRC meeting are now available in our wiki [1]. Among the discussed topics we decided about a date for the upcoming 1.5.0 release, this is scheduled for the 2th march 2009. The first release that will use the new common sip-router core [2] is scheduled for May. After 1.5.0 we will also do a bug fix release (1.4.4) for our latest stable branch. All dates can be found in our wiki as well at [3]. Cheers, Henning [1]http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-minutes… [2]http://sip-router.org/benefits/ [3]http://www.kamailio.org/dokuwiki/doku.php/development:release-policy#date… -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[Kamailio-Users] IRC meeting 17th februar, 15:00 UTC
by Henning Westerholt 17 Feb '09

17 Feb '09

Hi all, as the release cycle for 1.5.0 slowly come to an end, i think its time for a new IRC developer meeting, to focus on the upcoming development. I propose next Tuesday, 17th Feb 15:00 UTC. I've created a wiki page to collect an agenda for this meeting, and already added some content i would like to discuss. Please add your yours also and comment on already existing topics. Agenda: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-agenda-17… IRC meetings page: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meetings Cheers, Henning

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[Serusers] Problem with REGISTER
by buslik 17 Feb '09

17 Feb '09

Hi all, I have just default ser.cfg And it seems that if (uri==myself) never happens. SER trying to relay to <my domain name>:5060 and there my Gigaset C470IP just replies 405 Not Allowed. How to explain to the SER that it has <my domain name>? I have put alias=<my domain name> into my ser.cfg and SIP_DOMAIN=<my domain name> into env. It does no help. How to get any log out of it? -- Slava.

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[Kamailio-Users] Lost
by daniel medina 17 Feb '09

17 Feb '09

Hi everybody, I am trying to set up OpenSer as a Load balancing for Asterisk. I really need to know where to start. Please can anyone give me a hint where to star or if there is any tutorial??? Thanks in advance Daniel

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Re: [Kamailio-Users] ERROR:uac:replace_from
by Daniel-Constantin Mierla 16 Feb '09

16 Feb '09

Are you sure uac_replace_from() was called for initial invite? There has to be a new param (vsf) in the Recor-Route header and it is not there. Cheers, Daniel On 02/16/2009 04:50 PM, BERGANZ François wrote: > No. Time Source Destination Protocol Info > > [...] > > There is no 200OK because, the bye isn't good! > > > Thank you > > > Cordialement, > BERGANZ François > > > http://www.acropolistelecom.net >  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > > -----Message d'origine----- > De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Envoyé : lundi 16 février 2009 15:37 > À : BERGANZ François > Cc : users(a)lists.kamailio.org > Objet : Re: [Kamailio-Users] ERROR:uac:replace_from > > > > On 02/16/2009 04:07 PM, BERGANZ François wrote: > >> Now, I have modparam("uac","from_restore_mode","none") >> >> I control everything! Nothing in automatic! >> >> > :-) -- the power in your hands -- as said in the previous email, if you > send me the invite and appropriate 200ok, I can tell whose fault is. > > Cheers, > Daniel > > >> Cordialement, >> BERGANZ François >> >> >> http://www.acropolistelecom.net >>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >> >> -----Message d'origine----- >> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Envoyé : lundi 16 février 2009 14:07 >> À : BERGANZ François >> Cc : users(a)lists.kamailio.org >> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >> >> >> >> On 02/16/2009 03:03 PM, BERGANZ François wrote: >> >> >>> Then, if it is automatic: >>> >>> If I change with uac_replace_from() the INVITE, the bye have to be the same ? >>> In my case, it doesn't >>> >>> >>> >> Paste the BYE here, maybe the Route parameter used for auto-update is >> missing... >> >> Cheers, >> Daniel >> >> >> >>> Cordialement, >>> BERGANZ François >>> >>> >>> http://www.acropolistelecom.net >>>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >>> >>> >>> -----Message d'origine----- >>> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Envoyé : lundi 16 février 2009 12:39 >>> À : BERGANZ François >>> Cc : users(a)lists.kamailio.org >>> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >>> >>> >>> >>> On 02/16/2009 01:33 PM, BERGANZ François wrote: >>> >>> >>> >>>> It is for a bye request >>>> >>>> >>>> >>>> >>> BYE is withing dialog (practically ending the dialog). You do not need >>> to replace the From for it. It is done automatically -- see my other post. >>> >>> Cheers, >>> Daniel >>> >>> >>> >>> >>>> Cordialement, >>>> BERGANZ François >>>> >>>> >>>> http://www.acropolistelecom.net >>>>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >>>> >>>> >>>> -----Message d'origine----- >>>> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>>> Envoyé : lundi 16 février 2009 12:25 >>>> À : BERGANZ François >>>> Cc : users(a)lists.kamailio.org >>>> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >>>> >>>> Hello, >>>> >>>> On 02/16/2009 01:10 PM, BERGANZ François wrote: >>>> >>>> >>>> >>>> >>>>> Hello, >>>>> >>>>> I have an error with uac_replace_from() function. >>>>> >>>>> I want to do uac_replace_from(“$fn”,” sip:181812012@217.64.x.x”); >>>>> >>>>> I have that error: ERROR:uac:replace_from: decline FROM replacing in >>>>> sequential request in auto mode (has TO tag) >>>>> >>>>> Have you an idea? >>>>> >>>>> >>>>> >>>>> >>>>> >>>> you call uac_replace_from() for SIP messages within dialog and the >>>> replace mode is auto (which means is done automatically by kamailio) - >>>> therefore will be a duplicate. >>>> >>>> Either check for to-tag when you do the replacement or set replace mode >>>> to manual. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> >>>> >>>> >>> >>> >>> >> >> > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] ERROR:uac:replace_from
by BERGANZ François 16 Feb '09

16 Feb '09

Hello, I have an error with uac_replace_from() function. I want to do uac_replace_from($fn, sip:181812012@217.64.x.x); I have that error: ERROR:uac:replace_from: decline FROM replacing in sequential request in auto mode (has TO tag) Have you an idea? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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Re: [Kamailio-Users] ERROR:uac:replace_from
by Daniel-Constantin Mierla 16 Feb '09

16 Feb '09

On 02/16/2009 03:36 PM, BERGANZ François wrote: > the bye receive by the SER > The route parameter is missing, please send me the INVITE going out from kamailio and the 200ok for invite received by Kamailio. Should reveal whose fault is. Cheers, Daniel > thank you > > > > [...] > > > > Cordialement, > BERGANZ François > > > http://www.acropolistelecom.net >  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > > -----Message d'origine----- > De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Envoyé : lundi 16 février 2009 14:07 > À : BERGANZ François > Cc : users(a)lists.kamailio.org > Objet : Re: [Kamailio-Users] ERROR:uac:replace_from > > > > On 02/16/2009 03:03 PM, BERGANZ François wrote: > >> Then, if it is automatic: >> >> If I change with uac_replace_from() the INVITE, the bye have to be the same ? >> In my case, it doesn't >> >> > > Paste the BYE here, maybe the Route parameter used for auto-update is > missing... > > Cheers, > Daniel > > >> Cordialement, >> BERGANZ François >> >> >> http://www.acropolistelecom.net >>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >> >> >> -----Message d'origine----- >> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Envoyé : lundi 16 février 2009 12:39 >> À : BERGANZ François >> Cc : users(a)lists.kamailio.org >> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >> >> >> >> On 02/16/2009 01:33 PM, BERGANZ François wrote: >> >> >>> It is for a bye request >>> >>> >>> >> BYE is withing dialog (practically ending the dialog). You do not need >> to replace the From for it. It is done automatically -- see my other post. >> >> Cheers, >> Daniel >> >> >> >>> Cordialement, >>> BERGANZ François >>> >>> >>> http://www.acropolistelecom.net >>>  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. >>> >>> >>> -----Message d'origine----- >>> De : Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Envoyé : lundi 16 février 2009 12:25 >>> À : BERGANZ François >>> Cc : users(a)lists.kamailio.org >>> Objet : Re: [Kamailio-Users] ERROR:uac:replace_from >>> >>> Hello, >>> >>> On 02/16/2009 01:10 PM, BERGANZ François wrote: >>> >>> >>> >>>> Hello, >>>> >>>> I have an error with uac_replace_from() function. >>>> >>>> I want to do uac_replace_from(“$fn”,” sip:181812012@217.64.x.x”); >>>> >>>> I have that error: ERROR:uac:replace_from: decline FROM replacing in >>>> sequential request in auto mode (has TO tag) >>>> >>>> Have you an idea? >>>> >>>> >>>> >>>> >>> you call uac_replace_from() for SIP messages within dialog and the >>> replace mode is auto (which means is done automatically by kamailio) - >>> therefore will be a duplicate. >>> >>> Either check for to-tag when you do the replacement or set replace mode >>> to manual. >>> >>> Cheers, >>> Daniel >>> >>> >>> >>> >> >> > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] cfg scripts bank - inside group dial
by Daniel-Constantin Mierla 16 Feb '09

16 Feb '09

Hello, there were many useful hints and examples of how to solve specific needs or add new features just by config language. However, it is hard to search and find them back in the archive after a while. Therefore I created a new wiki page to collect them. If you see such example flowing on the mailing lists, take it and add it there or at least set the link to the mailing list post. Here is the page: http://www.kamailio.org/dokuwiki/doku.php/cfg-scripts-bank:main-index I added first example that adds inside group dial feature. You can group users by interest and they can call each other using short numbers instead of entire extension. Check it here: http://www.kamailio.org/dokuwiki/doku.php/cfg-scripts-bank:group-dial Practically is less than 10 lines, with loading additional modules and setting the parameters. Looking forward to your contributions there! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Serusers] Serusers Digest, Vol 70, Issue 4
by 郭灵犀 16 Feb '09

16 Feb '09

help 在2009-02-06，serusers-request(a)lists.iptel.org 写道： >Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers >or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > >You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Serusers digest..." > > >Today's Topics: > > 1. Re: configure SER and SEMS to work together (Stefan Sayer) > 2. preload_udomain error (caio) > 3. Re: preload_udomain error (solved) (caio) > 4. Re: Error loading "trusted" table (permissions.so) (caio) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 05 Feb 2009 13:25:00 +0100 >From: Stefan Sayer <stefan.sayer(a)iptego.com> >Subject: Re: [Serusers] configure SER and SEMS to work together >To: Martin Lane <martin.l.lane(a)gmail.com> >Cc: sems(a)iptel.org, Serusers(a)lists.iptel.org >Message-ID: <498ADA9C.4050302(a)iptego.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > > >o Martin Lane [02/04/09 20:17]: >> I am following the 'mini-Howto on how to configure SER and SEMS to work >> together' guide on iptel.org. So I have a separate directory for >as it says at the top of that howto, with a more recent version of SEMS >this is not required any more; especially if you are using 1.1.0 >version. Also note that there is the 1.1.0 release available >(http://ftp.iptel.org/pub/sems/sems-1.1.0.tar.gz). > >For SEMS 1.1.0 (recommended!), you can use the default configuration, >just configure 'application' and the server's IP address and port used >in sems.conf. Then just send the call to the SIP IP:port, e.g. using ser >with t_relay_to_udp("<sip_ip>","<sip_port>"); > >for completeness, the following comments: > >> integrating the SER and SEMS functionality. For development purposes is >> there a switch to change the path were SER looks for the modules? As >no, you need to change every loadmodule line. > >> well control which version of SEMS is being used? I've installed Sip >> Express Media Server (1.1.0-rc1 (i386/linux)) as well as the version >> from the howto for testing. >> >> My basic problem is I currently get a version mismatch error: >> >> 0(9590) ERROR: module version mismatch for >> /usr/local/lib/ser/modules/sl.so; core: ser 0.9.6-sems (i386/linux); >> module: ser 0.9.6 (i386/linux) >> > >Your specific error comes from that the module you are trying to load >was not compiled with the same version, e.g. you have one ser installed >into /usr/local and you have another one in a local directory. > >Regards >Stefan > >> The appropriate modules are located in under my ser-sems test dir and I >> have not been successful in redirecting SER to look there. >> >> Also, I have modified ser.cfg and sems.cfg for my system but I did not >> find a default ser_sems.cfg. Is this file necessary and where is the >> default location? >> >> Thanks for any help or advice. >> >> mll >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > >-- >Stefan Sayer >VoIP Services > >stefan.sayer(a)iptego.com >www.iptego.com > >IPTEGO GmbH >Am Borsigturm 40 >13507 Berlin >Germany > >Amtsgericht Charlottenburg, HRB 101010 >Geschaeftsfuehrer: Alexander Hoffmann > > >------------------------------ > >Message: 2 >Date: Thu, 05 Feb 2009 12:18:40 -0200 >From: caio <elcaio(a)gmail.com> >Subject: [Serusers] preload_udomain error >To: serusers(a)lists.iptel.org >Message-ID: <498AF540.4030102(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hi all, > >Running ser0.9.7 on Solaris 10 (mysql 5.0.51 + serweb 0.9.6) > >Have this problem when I stop ser service, and then start it again (ser >-E -D).. > > 0(0) preload_udomain: ERROR: bad callid record in table aliases > 0(0) preload_udomain: ERROR: for username 579981, contact >sip:579981@mydomain.com > 0(0) preload_udomain: ERROR: skipping... > >And finally it dies with: > > 0(0) get_connection(): Connection not found in the pool > Fallo de segmentaci??n (core dumped) > >It happens with many records on aliases table and happens too with only >one record: > >mysql> select * from aliases\G >*************************** 1. row *************************** > username: 579981 > domain: 10.10.1.1 > contact: sip:579981@mydomain.com > received: NULL > expires: 2009-02-05 11:45:21 > q: 1.00 > callid: >The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything > cseq: 42 >last_modified: 2009-02-05 11:45:46 > replicate: 0 > state: 0 > flags: 128 > user_agent: SIP Express Router FIFO >1 row in set (0.00 sec) > > >SER start normally if I delete all the records on this table. But I need >create the aliases again, and this is not the normal behaviour..I think. > >Here the description of aliases table: > >mysql> desc aliases; >+---------------+---------------------+------+-----+---------------------+-------+ >| Field | Type | Null | Key | Default > | Extra | >+---------------+---------------------+------+-----+---------------------+-------+ >| username | varchar(64) | NO | PRI | > | | >| domain | varchar(128) | NO | PRI | > | | >| contact | varchar(255) | NO | PRI | > | | >| received | varchar(255) | YES | | NULL > | | >| expires | datetime | NO | | 2020-05-28 21:32:15 >| | >| q | float(10,2) | NO | | 1.00 > | | >| callid | varchar(255) | NO | | Default-Call-ID > | | >| cseq | int(11) | NO | | 42 > | | >| last_modified | timestamp | NO | | CURRENT_TIMESTAMP > | | >| replicate | int(10) unsigned | NO | | 0 > | | >| state | tinyint(1) unsigned | NO | | 0 > | | >| flags | int(11) | NO | | 0 > | | >| user_agent | varchar(50) | NO | | > | | >+---------------+---------------------+------+-----+---------------------+-------+ >13 rows in set (0.01 sec) > > >Important: when "location" table keeps some records, ser doesn't start too: > >mysql> select * from location\G >*************************** 1. row *************************** > username: 579981 > domain: mydomain.com > contact: sip:579981@10.10.60.115:5060 > received: NULL > expires: 2009-02-05 11:46:16 > q: -1.00 > callid: 1f46c06d5e034425286f095b11cf4a1a(a)10.10.60.115 > cseq: 6 >last_modified: 2009-02-05 11:45:46 > replicate: 0 > state: 0 > flags: 0 > user_agent: CM5K-TA2S (807100.02) >1 row in set (0.00 sec) > >Here too I must to delete all the records to let SER start successfully. > >Does anyone knows about this issue? > >Regards, >Claudio > > >------------------------------ > >Message: 3 >Date: Thu, 05 Feb 2009 17:13:09 -0200 >From: caio <elcaio(a)gmail.com> >Subject: Re: [Serusers] preload_udomain error (solved) >To: serusers(a)lists.iptel.org >Message-ID: <498B3A45.5070501(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >caio escribi?: >> Hi all, >> >> Running ser0.9.7 on Solaris 10 (mysql 5.0.51 + serweb 0.9.6) >> >> Have this problem when I stop ser service, and then start it again (ser >> -E -D).. >> >> 0(0) preload_udomain: ERROR: bad callid record in table aliases >> 0(0) preload_udomain: ERROR: for username 579981, contact >> sip:579981@mydomain.com >> 0(0) preload_udomain: ERROR: skipping... >> >[cut] > >solved. >there was an old installation of mysql libs (in /usr/sfw/), then I >removed it, and finally recompile mysql.so module including correct >mysql.h (/opt/csw/). > >cheers, >caio > > >------------------------------ > >Message: 4 >Date: Thu, 05 Feb 2009 17:18:03 -0200 >From: caio <elcaio(a)gmail.com> >Subject: Re: [Serusers] Error loading "trusted" table (permissions.so) >To: serusers(a)lists.iptel.org >Message-ID: <498B3B6B.20606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Miklos Tirpak escribi?: > >>> Seem that VAL_TYPE of "val + 2" isn't STRING, read it as DOUBLE >>> (DB_DOUBLE = 1). >>> Do not know why. >>> And because of this the 'if' conditions never are satisfy on trusted.c >>> (line #396): >>> >>> if ((ROW_N(row + i) == 3) && >>> (VAL_TYPE(val) == DB_STRING) && !VAL_NULL(val) && >>> (VAL_TYPE(val + 1) == DB_STRING) && !VAL_NULL(val >>> + 1) && >>> (VAL_TYPE(val + 2) == DB_STRING) && !VAL_NULL(val >>> + 2)) { >> >> >> Thank you very much for the debug lines! I think SER simply uses the >> field types returned by the mysql server, so there might be a problem >> with the server or library. You can check what the field types are along >> with their names returned by the server in modules/mysql/res.c, search >> for the mysql_fetch_fields function. >> >> Miklos >> > >Miklos, > >I've resolved this issue and a similar one ("preload_udomain error"). >There was an old installation of mysql libs (in /usr/sfw/), then I >removed it, and finally recompile mysql.so module including correct >mysql.h (/opt/csw/). > >Now fields returned are with their correct type. > >Thank you.. > >Caio > > >------------------------------ > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > >End of Serusers Digest, Vol 70, Issue 4 >***************************************

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[Kamailio-Users] mediaproxy
by NYam 14 Feb '09

14 Feb '09

Dear Sir, As it was stated in your website, KAMAILIO (OpenSER) is available in binary format for several embedded platforms, including Linksys NSLU. Does it mean that mediaproxy is already installed into the platform with KAMAILIO? If not, how mediaproxy will be installed? Will it be hard to do so? with kind regards Nyam Sharibu

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[Kamailio-Users] SIREMIS feedback
by Elena-Ramona Modroiu 14 Feb '09

14 Feb '09

Hi, it seems that meeting at FOSDEM brought lot of ideas around. Stefan already had a good suggestion - integration of SIREMIS with SEMS giving the possibility to monitor SEMS servers in the same nice manner from a web page. I would like to collect suggestions about what would be good to have in SIREMIS, what should be improved. My goal up to Kamailio 1.5 release is to add support for carrierroute, a cdr rating engine and cleanup the source tree in order to upload the sourceforge svn - this would make easier for others to contribute. However, if it is something we find very useful then I will prioritize. There is time to add new features after the release, as well. Please shoot here any crazy idea you have for a web management interface of Kamailio and later SIP Router. Regards, Ramona

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[Kamailio-Users] UDP packet loss
by Alex R.S.M 13 Feb '09

13 Feb '09

Hi, I am performing SIP to SIP performance testing using Kamailio proxy (UAC-Kamailio-UAC). After increasing the load to 300 calls/sec Kamailio begins dropping UDP packets. (250 cps works fine). CPU utilization and network bandwidth looks fine. I think the bottleneck is the Linux UDP buffer size of the proxy server. Here is the proxy server information: $ uname -a Linux lab-linux01 2.6.24-19-server #1 SMP Wed Jun 18 14:44:47 UTC 2008 x86_64 GNU/Linux $ sudo sysctl -a | grep net.core.rmem_max net.core.rmem_max = 131071 The question is if I increase the linux UDP buffer size do I need to change any configuartion parameter (like MAX_RECV_BUFFER_SIZE) in Kamailio? Thanks, Alex R _________________________________________________________________ Windows Live™: Keep your life in sync. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_022…

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[Kamailio-Users] Trying to compile openser 1.3.3 without PKG_MALLOC support
by toly 13 Feb '09

13 Feb '09

Greetings, Trying to compile openser 1.3.3 without PKG_MALLOC, just using OS malloc Here's me Makefile.defs DEFS+= $(extra_defs) \ -DNAME='"$(MAIN_NAME)"' -DVERSION='"$(RELEASE)"' -DARCH='"$(ARCH)"' \ -DOS='"$(OS)"' -DCOMPILER='"$(CC_VER)"' -D__CPU_$(ARCH) -D__OS_$(OS) \ -D__SMP_$(ISSMP) -DCFG_DIR='"$(cfg-target)"'\ -DSHM_MEM -DSHM_MMAP \ #-DPKG_MALLOC \ -DUSE_IPV6 \ -DUSE_MCAST \ -DUSE_TCP \ -DDISABLE_NAGLE \ -DHAVE_RESOLV_RES \ -DSTATISTICS \ -DCHANGEABLE_DEBUG_LEVEL \ #-DF_MALLOC \ -DQM_JOIN_FREE \ #-DDBG_QM_MALLOC \ #-DDBG_F_MALLOC \ #-DNO_DEBUG \ #-DNO_LOG \ #-DVQ_MALLOC \ #-DDBG_LOCK \ #-DNOSMP \ #-DEXTRA_DEBUG \ #-DUSE_SHM_MEM It's bailing out compiling blacklists.c blacklists.c: In function `rm_dups': blacklists.c:435: error: syntax error before "else" blacklists.c:440: error: syntax error before "else" blacklists.c: At top level: blacklists.c:443: error: syntax error before "else" blacklists.c:445: warning: type defaults to `int' in declaration of `q' blacklists.c:445: error: invalid type argument of `->' blacklists.c:445: warning: data definition has no type or storage class blacklists.c:446: error: syntax error before '}' token make: *** [blacklists.o] Error 1 Did anybody had a problem compiling openser without pkg_malloc support? I would appreciate any help. TIA Toly. -- View this message in context: http://www.nabble.com/Trying-to-compile-openser-1.3.3-without-PKG_MALLOC-su… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] Kernel Message
by Brian Chamberlain 12 Feb '09

12 Feb '09

Hi All, Is this something I should worry about? Feb 12 20:27:22 sw1 kernel: [3273100.087148] INFO: task kamailio:1845 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.087206] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.087291] kamailio D 8ce9265a 0 1845 1829 Feb 12 20:27:22 sw1 kernel: [3273100.087346] f6a98da0 00000082 23f7e426 8ce9265a 000701ae f6a98f2c c302bfa0 00000002 Feb 12 20:27:22 sw1 kernel: [3273100.087427] c01344f6 001e17ce 00000001 00000000 c01190e6 f6a98da0 f6a98da0 c02c4dc0 Feb 12 20:27:22 sw1 kernel: [3273100.087538] f74eea84 f74eea8c f74eea88 f6a98da0 c02b8846 f68c3d68 f74eea8c f6a98da0 Feb 12 20:27:22 sw1 kernel: [3273100.087648] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.087730] [<c01344f6>] ktime_get +0xd/0x21 Feb 12 20:27:22 sw1 kernel: [3273100.087784] [<c01190e6>] hrtick_start_fair+0xeb/0x12c Feb 12 20:27:22 sw1 kernel: [3273100.087840] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.087896] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.087945] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.087998] [<c01318e9>] autoremove_wake_function+0xd/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.088055] [<c011845d>] __wake_up_common+0x2e/0x58 Feb 12 20:27:22 sw1 kernel: [3273100.088109] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.088165] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.088223] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.088273] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.088327] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.088387] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.088438] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.088497] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.088554] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.088610] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.091119] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.091174] ======================= Feb 12 20:27:22 sw1 kernel: [3273100.091218] INFO: task kamailio:1846 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.091274] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.091358] kamailio D 00000002 0 1846 1829 Feb 12 20:27:22 sw1 kernel: [3273100.091413] f6a995e0 00000086 00000000 00000002 00000000 f6a9976c c3017fa0 00000000 Feb 12 20:27:22 sw1 kernel: [3273100.091495] ffffffff 1d621a39 00000000 00000000 00000000 00000000 00000000 000000ff Feb 12 20:27:22 sw1 kernel: [3273100.091605] f74eea84 f74eea8c f74eea88 f6a995e0 c02b8846 f74eea8c f6b21d68 f6a995e0 Feb 12 20:27:22 sw1 kernel: [3273100.091715] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.091790] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.091847] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.091895] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.091949] [<c0129ab0>] lock_timer_base+0x19/0x35 Feb 12 20:27:22 sw1 kernel: [3273100.092002] [<c0129b10>] try_to_del_timer_sync+0x44/0x4a Feb 12 20:27:22 sw1 kernel: [3273100.092057] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.092108] [<c01297ec>] process_timeout+0x0/0x5 Feb 12 20:27:22 sw1 kernel: [3273100.092163] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.092219] [<c015a0d0>] __rmqueue +0x13/0x174 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.092221] ======================= Feb 12 20:27:22 sw1 kernel: [3273100.092221] INFO: task kamailio:22820 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.092221] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.092221] kamailio D 854aa937 0 22820 22803 Feb 12 20:27:22 sw1 kernel: [3273100.092221] f762fa40 00000086 2603ada4 854aa937 000767a5 f762fbcc c3021fa0 00000001 Feb 12 20:27:22 sw1 kernel: [3273100.092221] c01344f6 001fcdbd 00000001 00000000 c01190e6 f762fa40 f762fa40 c02c4dc0 Feb 12 20:27:22 sw1 kernel: [3273100.092221] f6b38544 f6b3854c f6b38548 f762fa40 c02b8846 f68a3d68 f6b3854c f762fa40 Feb 12 20:27:22 sw1 kernel: [3273100.092221] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c01344f6>] ktime_get +0xd/0x21 Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c01190e6>] hrtick_start_fair+0xeb/0x12c Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.092221] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.092274] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.092328] [<c01318e9>] autoremove_wake_function+0xd/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.092383] [<c015a05d>] __rmqueue_smallest+0x83/0xe3 Feb 12 20:27:22 sw1 kernel: [3273100.092438] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.092493] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.092551] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092601] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.092653] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.092712] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.092764] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.092822] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.092881] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.092937] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.092991] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.093046] ======================= Feb 12 20:27:22 sw1 kernel: [3273100.093090] INFO: task kamailio:22821 blocked for more than 120 seconds. Feb 12 20:27:22 sw1 kernel: [3273100.093146] "echo 0 > /proc/sys/ kernel/hung_task_timeout_secs" disables this message. Feb 12 20:27:22 sw1 kernel: [3273100.093230] kamailio D 8543066c 0 22821 22803 Feb 12 20:27:22 sw1 kernel: [3273100.110119] f762ede0 00000082 2603ada4 8543066c 000767a5 f762ef6c c3017fa0 00000000 Feb 12 20:27:22 sw1 kernel: [3273100.110201] c01344f6 001fcdbd 00000002 00000000 c01190e6 f762ede0 f762ede0 c02c4dc0 Feb 12 20:27:22 sw1 kernel: [3273100.110312] f6b38544 f6b3854c f6b38548 f762ede0 c02b8846 f6b3854c f6bfdd68 f762ede0 Feb 12 20:27:22 sw1 kernel: [3273100.110422] Call Trace: Feb 12 20:27:22 sw1 kernel: [3273100.110493] [<c01344f6>] ktime_get +0xd/0x21 Feb 12 20:27:22 sw1 kernel: [3273100.110543] [<c01190e6>] hrtick_start_fair+0xeb/0x12c Feb 12 20:27:22 sw1 kernel: [3273100.110598] [<c02b8846>] __mutex_lock_slowpath+0x50/0x7b Feb 12 20:27:22 sw1 kernel: [3273100.110653] [<c02b86dc>] mutex_lock +0xa/0xb Feb 12 20:27:22 sw1 kernel: [3273100.110702] [<c02a7ea9>] unix_dgram_recvmsg+0x3e/0x245 Feb 12 20:27:22 sw1 kernel: [3273100.110755] [<c01318e9>] autoremove_wake_function+0xd/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.110810] [<c011845d>] __wake_up_common+0x2e/0x58 Feb 12 20:27:22 sw1 kernel: [3273100.110864] [<c024ef76>] sock_recvmsg +0xde/0xf9 Feb 12 20:27:22 sw1 kernel: [3273100.110919] [<c01318dc>] autoremove_wake_function+0x0/0x2d Feb 12 20:27:22 sw1 kernel: [3273100.110976] [<c0117c67>] kunmap_atomic+0x55/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.111027] [<c0117c79>] kunmap_atomic+0x67/0x8a Feb 12 20:27:22 sw1 kernel: [3273100.111079] [<c015ae91>] get_page_from_freelist+0x34f/0x3e3 Feb 12 20:27:22 sw1 kernel: [3273100.111138] [<c024fd9a>] sys_recvfrom +0xb4/0x116 Feb 12 20:27:22 sw1 kernel: [3273100.111189] [<c015b0fb>] __alloc_pages_internal+0xb5/0x34e Feb 12 20:27:22 sw1 kernel: [3273100.111248] [<c0163cd3>] handle_mm_fault+0x257/0x690 Feb 12 20:27:22 sw1 kernel: [3273100.111307] [<c025024d>] sys_socketcall+0x135/0x19e Feb 12 20:27:22 sw1 kernel: [3273100.111362] [<c0103853>] sysenter_past_esp+0x78/0xb1 Feb 12 20:27:22 sw1 kernel: [3273100.111417] [<c02b0000>] acpi_pci_root_add+0x165/0x296 Feb 12 20:27:22 sw1 kernel: [3273100.111471] =======================

3 12

[Kamailio-Users] SIREMIS v0.9.1 Released
by Elena-Ramona Modroiu 12 Feb '09

12 Feb '09

Hi everybody, a new version, 0.9.1, of SIREMIS - the web management interface for Kamailio (OpenSER) and SIP-Router.org - is released. New features and improvements: * send MI commands to Kamailio (OpenSER) via UDP * display the output of a MI command * ability to predefine MI commands that can be selected from a drop-down menu * create graphic charts from statistic data * one or more charts per page * one or more graphics per chart * now new coding required, only configuration changes required to add new charts * drop-down selection for user AVP type Goals for next version: * carrierroute management * rating engine for CDRs View screenshots of the 0.9.1: http://www.asipto.com/gallery/v/siremis/ Relevant screenshots for new features: - charts: http://www.asipto.com/gallery/v/siremis/siremis_12.jpg.html - MI: http://www.asipto.com/gallery/v/siremis/siremis_15.jpg.html Demo site (it works on a database with random data, no MI; username: admin, password: admin): http://siremis.asipto.com/demo/ Download: http://siremis.asipto.com/pub/downloads/siremis/ Install/Upgrade documentation: http://siremis.asipto.com/install/ Regards, Ramona

3 3

[Kamailio-Users] how to work with Remote-Party-ID ?
by BERGANZ François 12 Feb '09

12 Feb '09

hello, i need to get the Remote-Party-ID in the header. I have that: Remote-Party-ID: "francois bgz" <sip:170725014@217.64.x.x:5067>;screen=yes;privacy=off;party=calling;-call-i nitiator=5016;-call-initiator-location=int;-redirected-by;-int-ext=5016;-ent -name=Acro;-direction=ext;-call-id=55665. I need to have -ent-name=Acro but, when I use $re, I have sip:170725014@217.64.x.x:5067 how can I do ? thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

3 2

Re: [Kamailio-Users] [Kamailio-Devel] SIREMIS v0.9.1 Released
by Elena-Ramona Modroiu 12 Feb '09

12 Feb '09

Hi Stefan, Stefan Sayer wrote: > Hi, > > o Elena-Ramona Modroiu [02/11/09 11:48]: > >> Hi everybody, >> >> a new version, 0.9.1, of SIREMIS - the web management interface for >> Kamailio (OpenSER) and SIP-Router.org - is released. >> > this looks very nice. > thanks! > Just recently I started with some monitoring support for SEMS > (http://lists.iptel.org/pipermail/semsdev/2009-February/003517.html) > The way this works is to push some call related info to the monitoring > module, and access it from the outside via DI functions, e.g. via XMLRPC > (http://svn.berlios.de/wsvn/sems/trunk/doc/Readme.monitoring). So you > can get e.g. a list of callids by executing the XMLRPC function > server.listActive(), and get the info for a call with server.get(callid). > > At the moment it logs ruri/from/to/application/direction in the core, > for conference application additionally the conference room. To > illustrate, you would get this from the server: > >>> s.list() > ['3c2d4bde0d60-gds81g00uz1n'] > >>> s.get(s.list()[0]) > [{'from': '"2(a)stefan-lap.office.iptego.de" > <sip:2@stefan-lap.office.iptego.de>', 'app': 'conference', 'to': > '<sip:35@stefan-lap.office.iptego.de;user=phone>', 'conf_id': '35', > 'ruri': 'sip:35@stefan-lap.office.iptego.de;user=phone', 'dir': 'in'}] > > This is of course extensible, e.g. codec used, call start/end time etc. > > Maybe, if we would add a way to monitor SEMS servers from within > SIREMIS, it would give a good opportunity to monitor e.g. conference > bridge, voicemail server, prepaid app, or b2bua (like sbc). > > What do you think? > I think it would be very useful. It looks to me more like an extension to communicate via XMLRPC with SEMS - send command and display the result. What is the XMLRPC you are using in SEMS? The C library used in Kamailio proved quite unstable, therefore SIREMIS uses MI UDP - simpler protocol and easy to troubleshoot. There should be XMLRPC PHP lib, therefore easy to extend SIREMIS. Is it possible to get statistic information out of SEMS? Like number of active calls? Such data can be used to build charts, which is nice to monitor the service health and evolution. Regards, Ramona

2 2

[Serusers] Forking Proxy
by Amanda Mattiuz 12 Feb '09

12 Feb '09

Hello all, Is is possible to configure SER to act as a forking proxy, where the INVITE messages that arrive at the server are forwarded to multiple clients ins parallel? I saw something about call hunt, but it seems that the INVITES are sent sequentially, could I make it become concurrently? Thanks, Amanda

2 1

[Kamailio-Users] IP SEC tunel
by ALAEDDINE abbech 11 Feb '09

11 Feb '09

hi, How can i do a tunel (use IP SEC)whith my serveur openser. is there a code to write to the configuration file to do this. thanks __________________________________________________________________________________________________ Ne pleurez pas si votre Webmail ferme ! Récupérez votre historique sur Yahoo! Mail ! http://fr.docs.yahoo.com/mail/transfert_mails.html

2 1

[Kamailio-Users] Kamilio Failover Options
by Stagg Shelton 11 Feb '09

11 Feb '09

I am looking for some ideas on how to create a more redundant setup in my operation. About 60 days ago our openser 1.3.2 system started sending out 500 errors. Looking through the log file I found alot of ERROR:tm:new_t: out of mem that corresponded to the INVITES. A simple restart of openser freed the memory and calls processed normally as expected. Since then we have upgraded to openser 1.3.3 and haven't experienced the problem since. Currently I am testing kamilio 1.4.3 and planning my migration into kamilio from openser. What I would like to create is a system that has two kamailio systems where one can fail to the other in the event of an application error that prevents calls from processing. Does anyone have any suggestions on implementing a redundant kamilio system, or can point me in the direction of where to start reading and researching. Thanks Stagg Shelton www.3pconnect.com

3 2

Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Daniel-Constantin Mierla 11 Feb '09

11 Feb '09

Hello, On 02/11/2009 03:20 PM, Abdul Hakeem wrote: > Hi, > > Just wanted to touch base on the parser to find out if you are still > pursuing this. > it is still on my todo list, however didn't make it in the upcoming 1.5.0. Ovidiu Sas did some commits to the SDP parser and I know couple of others using it with success. Please open a feature request at: http://sourceforge.net/tracker/?atid=743023&group_id=139143&func=browse It is easier to track and not forgot about it for the next release. Thanks, Daniel > I am in the mind of purchasing some Cisco equipment to do the Sip-MAP > gateway from a GSM network. > I have someone else from the OpenBTS with me on this. > The object is to provide a full fledged sip based routing of voice, sms and > multimedia calls from gsm handsets to BTS with a Sip server backend. > > Cheers, > Abdul Hakeem > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin > Mierla > Sent: 03 September 2008 17:45 > To: Kristian Kielhofner > Cc: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc > > Hello, > > On 09/03/08 19:10, Kristian Kielhofner wrote: > >> Hello everyone, >> >> Feel free to set $OSSPROXY to whatever you like = >> Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). >> >> $OSSPROXY is my favorite SIP toolkit. In addition to being an >> excellent proxy and registrar it serves as a platform to do SIP >> mangling and wrangling. With that in mind let me propose a crazy >> idea... >> >> How hard would it be to get $OSSPROXY to be able to parse what is >> commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has >> no business getting too involved in the SDP but I feel it makes sense >> in some instances (like this one). What if I want to make a routing >> decision based on an encapsulated ISUP parameter of some sort? Who >> knows what it could be!?!? Maybe there are some other uses but that >> makes the most sense and seems to be the most practical. It should be >> possible to develop an $OSSPROXY module to: >> >> - Use the SDP parser to find the ISUP SDP part: >> >> http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing >> >> - Pass that part into a "real" ISUP parser (if necessary). I couldn't >> seem to find the perfect library but perhaps something like this could >> work: >> >> http://www.openss7.org/isup.html >> >> - Make the ISUP parameters available via AVPs, pseudo variables, etc >> for use in the $OSSPROXY script. >> >> I don't really have a need for this but it seems like it could be >> cool. Am I nuts? >> >> >> > I would expose the body as a dynamic structure, so components will be > accessible in the script, e.g. (naming schema just for example), > > if($body(0=>content-type)=="application/sdp") > { > if($body(0=>m=>type) == "audio") > { > } > } > > Based on the available content type parsers (sdp, isup), the structure has > different attributes. > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Looking for a highly available / load balanced setup
by John D 11 Feb '09

11 Feb '09

I currently have three Asterisk servers in my environment. None of them speak to each other just yet. The idea is to install Kamailio and be able to load balance/distribute SIP calls to any of the three asterisk boxes. If an entire box plug gets pulled, the operation continues. If a Kamailio box gets the plug pulled (and due to limited resources, I plan at first to run kamailio on all three of the asterisk boxes), the operation will continue. Does anyone have good documentation or insight in order to facilitate this sort of setup? Thanks.

4 6

[Kamailio-Users] New developer: Don Jackson
by Henning Westerholt 10 Feb '09

10 Feb '09

Hi all, its my pleasure to announce a new developer: Don Jackson. He'll maintain in the future the OpenBSD port of our favorite SIP server, and make sure that Kamailio will work there as good as on other operating systems. His sf.org id is 'dcjackson'. Thank you Don for the packaging you've done, i'm looking forward to work further with you. Best regards, Henning

1 0

[Kamailio-Users] OpenBSD port for Kamailio 1.4.3
by Don Jackson 10 Feb '09

10 Feb '09

Hello, I have updated my OpenBSD port to support Kamailio 1.4.3 I sent the updated port to the ports@openbsd mailing list: http://archives.neohapsis.com/archives/openbsd/2009-01/1107.html In addition, the port is available here: http://www.kamailio.org/pub/kamailio/1.4.3/packages/openbsd Thanks to Henning Westerholt for putting the files up on kamailio.org

1 0

Re: [Kamailio-Users] [Kamailio-Devel] FOSDEM2009 - first remarks/photos
by Daniel-Constantin Mierla 10 Feb '09

10 Feb '09

On 02/10/2009 04:44 PM, Iñaki Baz Castillo wrote: > 2009/2/10 Johansson Olle E <oej(a)edvina.net>: > >>> I have been to XMPP interop, although they focused on Voice/Jingle, I >>> tried the new purple module -- still some things to sort out and I will >>> post a resume about this particular case and entire event. >>> >> I spent monday on the XMPP dev meeting a got a few ideas for improvements. >> > > Nooo, forget XMPP!! > SIP SIMPLE already has enough cool specifications, the problem is that > nobody implements them! > :-) -- maybe SIP extension writers can learn from xmpp where they go first for prototype implementation and then to standardization. You are right, there are cool specification, but many do not map properly to reality. > Looking for interoperability with XMPP will not help, and I don't want > a future in which my softphone implements SIP for voice and XMPP for > presence/IM, that's a dirty workaround! > I think a competing open protocol is good all the time, makes things improve in both sides. Apart of that, sip and xmpp does not really match, SIP is more a routing-like protocol, while xmpp is a content-like protocol. XMPP network exist and cannot be ignored. A good usability of xmpp protocol will be communication between various points of your environment, sip servers, web servers, etc... Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 2

[Kamailio-Users] FOSDEM2009 - first remarks/photos
by Daniel-Constantin Mierla 10 Feb '09

10 Feb '09

Hello, thanks to Alexandr Dubovikov, a selection of photos from FOSDEM2009 and Kamailio/SIP Router Meeting are now available at: http://www.kamailio.org/events/2009-FOSDEM/FOSDEM2009-Album/ Some movies should become available as well. As for the FOSDEM itself, there were a lot of sessions focusing on scalability for backends, mainly from MySQL track. There are some goodies that will help improving performance with 0-lines of code in the sip servers. I have been to XMPP interop, although they focused on Voice/Jingle, I tried the new purple module -- still some things to sort out and I will post a resume about this particular case and entire event. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

[Kamailio-Users] dialog enhancements #3
by Daniel-Constantin Mierla 09 Feb '09

09 Feb '09

Hello, the dialog module for upcoming release kamailio/openser 1.5.0 (http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x) has the ability to execute a script route when a dialog timeouts. Just set the $dlg_ctx(timeout_route) before creating the dialog. route { ... $dlg_ctx(timeout_route) = 11; dlg_manage(); ... } route[11] { ... } Before the execution of timeout route the $dlg(...) psudo-variable is set to the expired dialog. One particular use case is to write full cdr, via sqlops and using $dlg(...) pseudo-variable. All functions that can be executed in a request route are safe here - there is a faked SIP message given as parameter. There are couple of other new properties that can be set before dialog creation: - send BYE at timeout - to make that happen you have to set: $dlg_ctx(timeout_bye) = 1; - set an integer that can be used as a bitmask flag checking later: $dlg_ctx(flags) = ...; Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] errors installing on centos 5 64 bits
by Fabian Borot 06 Feb '09

06 Feb '09

Hello I am trying to install it on Centos 5 64 bits using this command: NICER=1 make prefix=/usr/local/kamailio-install-dir all and it fails with this message: make[1]: Entering directory `/usr/src/kamailio-1.4.3-notls/modules/db_mysql' Linking db_mysql.so /usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl /usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching for -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make[1]: *** [db_mysql.so] Error 1 make[1]: Leaving directory `/usr/src/kamailio-1.4.3-notls/modules/db_mysql' make: *** [modules] Error 2 I think that somehow it is trying to use the lib for 32 bits or that my specific lib file is not what it should be or it is not on my system. I have the folders /usr/lib and usr/lib64 with its respective library files. Is this "libssl" related to openssl? I have these installed on my system already: [root@proxy1 lib64]# yum install openssl Setting up Install Process Parsing package install arguments Package openssl - 0.9.8b-10.el5_2.1.x86_64 is already installed. Package openssl - 0.9.8b-10.el5_2.1.i686 is already installed. Nothing to do [root@proxy1 lib64]# I also looked for "libssl" and this is what I see, it was only found in the folder for 32 bits libs if I am not mistaken. if that is the case how can i get the 64 bits equivalent? [root@proxy1 lib64]# find / -name "libssl.so" /usr/lib/libssl.so [root@proxy1 lib64]# thank you very much Fabian _________________________________________________________________ Windows Live™: E-mail. Chat. Share. Get more ways to connect. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_AE_Faster_022009

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Re: [Kamailio-Users] nathelper module: function rtpproxy_stream2uac/uas
by Rash 06 Feb '09

06 Feb '09

Hi Daniel, thanks for the quick repsonse, you delivered the magic hint! Putting the count paramaeter into double quotes works. Thanks for your effort! Rash

2 1

[Kamailio-Users] nathelper module: function rtpproxy_stream2uac/uas
by Rash 06 Feb '09

06 Feb '09

Hello, I'm using latest version of Kamailio (OpenSER) from daily snapshot, as i wanted to test the rtpproxy_stream2uac function of the nathelper module in order to make announcements with rtpproxy. Integration of Kamailio and RTPProxy was successfull and I added the following line to the kamailio.cfg: rtpproxy_stream2uas("/var/rtpproxy/prompts/music_on_hold", -1); as described in the nathelper documentation ( http://www.kamailio.org/docs/modules/devel/nathelper.html#rtpproxy_stream2x… ) After adding this line to the appropriate section (onreply_route in my case) kamctl refuses to start.: kamailio: CRITICAL:core:yyerror: parse error in config file, line 414, column 29-30: syntax error kamailio: CRITICAL:core:yyerror: parse error in config file, line 414, column 30-31: bad arguments where line 414 corresponds to the line shown above. I can't find the error as the line is copied from nathelper module. Playing around with the parameters, didn't lead to anything. It would be nice if anyone could help or tryout the example from nathelper documentation and feedback. Best regards Rash

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[Kamailio-Users] dialog enhancements #2
by Daniel-Constantin Mierla 06 Feb '09

06 Feb '09

Hello, another addition to the dialog module is the new pseudo-variable class $dlg(...) http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:devel#dlg_attr With it you can get access to all attributes of current processed dialog, e.g., caller and callee URI, start time, flags... A typical use case is to write full cdr when the BYE comes over - the dialog stores the start time and you can get the end time via $Ts. You can either write the acc record at dialog ending or update the start acc event written when INVITE was processed - you have access to the call-id, from tag and to tag in the right order and one shot of sql_query() will do it. Cheers, Daniel

3 2

[Kamailio-Users] location table
by joy yue 05 Feb '09

05 Feb '09

Hi, Is there a limitation on the number of rows in location table? In my rig, whatever number of users I use (>2million users), I notice the number of users in location table is 343707, which is far less than the number of users in my test. I thought previously that location table saves all the users in my test. Also when I use large number of users (>2million), openser pops up an error: ERROR: registrar:update_contacts: invalid cseq for aor <xxxx>. Has anyone saw this before? I am using openser1.3.2. Thanks, -Joy

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[Serusers] Error loading "trusted" table (permissions.so)
by caio 05 Feb '09

05 Feb '09

Hello all, Does anyone knows about this "permission.so" module error when try to load "trusted" table (mysql)? If "trusted" table is empty, there is no error, but whan at least there is 1 record, the error appears. for example: mysql> select * from trusted; +---------------+-------+--------------+ | src_ip | proto | from_pattern | +---------------+-------+--------------+ | 10.124.35.253 | any | ^sip:.*$ | +---------------+-------+--------------+ and when I try lo restart ser: Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 966737 daemon.info] permissions - initializing Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 414293 daemon.warning] Default allow file (/usr/local/etc/ser/permissions.allow) not found => empty rule set Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 844440 daemon.warning] Default deny file (/usr/local/etc/ser/permissions.deny) not found => empty rule set Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 895822 daemon.error] ERROR: permissions: trusted_reload(): Database problem Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 650101 daemon.crit] init_trusted(): Reload of trusted table failed Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 345573 daemon.error] Error while initializing allow_trusted function Jan 15 15:34:45 delcolimba /usr/local/sbin/ser[703]: [ID 660978 daemon.error] init_mod(): Error while initializing module permissions or when executing serctl fifo cmd: root@delcolimba /usr/local # sbin/serctl fifo trusted_dump 200 OK root@delcolimba /usr/local # sbin/serctl fifo trusted_reload 400 Trusted table reload failed root@delcolimba /usr/local # tail /var/adm/messages Jan 15 15:45:37 delcolimba /usr/local/sbin/ser[717]: [ID 895822 daemon.error] ERROR: permissions: trusted_reload(): Database problem Jan 15 15:45:37 delcolimba /usr/local/sbin/ser[717]: [ID 450600 daemon.error] ERROR: fifo_server: command (trusted_reload) processing failed Thank you, Caio

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[Serusers] preload_udomain error
by caio 05 Feb '09

05 Feb '09

Hi all, Running ser0.9.7 on Solaris 10 (mysql 5.0.51 + serweb 0.9.6) Have this problem when I stop ser service, and then start it again (ser -E -D).. 0(0) preload_udomain: ERROR: bad callid record in table aliases 0(0) preload_udomain: ERROR: for username 579981, contact sip:579981@mydomain.com 0(0) preload_udomain: ERROR: skipping... And finally it dies with: 0(0) get_connection(): Connection not found in the pool Fallo de segmentaciÃ³n (core dumped) It happens with many records on aliases table and happens too with only one record: mysql> select * from aliases\G *************************** 1. row *************************** username: 579981 domain: 10.10.1.1 contact: sip:579981@mydomain.com received: NULL expires: 2009-02-05 11:45:21 q: 1.00 callid: The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything cseq: 42 last_modified: 2009-02-05 11:45:46 replicate: 0 state: 0 flags: 128 user_agent: SIP Express Router FIFO 1 row in set (0.00 sec) SER start normally if I delete all the records on this table. But I need create the aliases again, and this is not the normal behaviour..I think. Here the description of aliases table: mysql> desc aliases; +---------------+---------------------+------+-----+---------------------+-------+ | Field | Type | Null | Key | Default | Extra | +---------------+---------------------+------+-----+---------------------+-------+ | username | varchar(64) | NO | PRI | | | | domain | varchar(128) | NO | PRI | | | | contact | varchar(255) | NO | PRI | | | | received | varchar(255) | YES | | NULL | | | expires | datetime | NO | | 2020-05-28 21:32:15 | | | q | float(10,2) | NO | | 1.00 | | | callid | varchar(255) | NO | | Default-Call-ID | | | cseq | int(11) | NO | | 42 | | | last_modified | timestamp | NO | | CURRENT_TIMESTAMP | | | replicate | int(10) unsigned | NO | | 0 | | | state | tinyint(1) unsigned | NO | | 0 | | | flags | int(11) | NO | | 0 | | | user_agent | varchar(50) | NO | | | | +---------------+---------------------+------+-----+---------------------+-------+ 13 rows in set (0.01 sec) Important: when "location" table keeps some records, ser doesn't start too: mysql> select * from location\G *************************** 1. row *************************** username: 579981 domain: mydomain.com contact: sip:579981@10.10.60.115:5060 received: NULL expires: 2009-02-05 11:46:16 q: -1.00 callid: 1f46c06d5e034425286f095b11cf4a1a(a)10.10.60.115 cseq: 6 last_modified: 2009-02-05 11:45:46 replicate: 0 state: 0 flags: 0 user_agent: CM5K-TA2S (807100.02) 1 row in set (0.00 sec) Here too I must to delete all the records to let SER start successfully. Does anyone knows about this issue? Regards, Claudio

1 1

[Serusers] configure SER and SEMS to work together
by Martin Lane 05 Feb '09

05 Feb '09

I am following the 'mini-Howto on how to configure SER and SEMS to work together' guide on iptel.org. So I have a separate directory for integrating the SER and SEMS functionality. For development purposes is there a switch to change the path were SER looks for the modules? As well control which version of SEMS is being used? I've installed Sip Express Media Server (1.1.0-rc1 (i386/linux)) as well as the version from the howto for testing. My basic problem is I currently get a version mismatch error: 0(9590) ERROR: module version mismatch for /usr/local/lib/ser/modules/sl.so; core: ser 0.9.6-sems (i386/linux); module: ser 0.9.6 (i386/linux) The appropriate modules are located in under my ser-sems test dir and I have not been successful in redirecting SER to look there. Also, I have modified ser.cfg and sems.cfg for my system but I did not find a default ser_sems.cfg. Is this file necessary and where is the default location? Thanks for any help or advice. mll

2 1

[Kamailio-Users] dialog enhancements #1
by Daniel-Constantin Mierla 04 Feb '09

04 Feb '09

Hello, during the week before freezing the code for release kamailio (openser) 1.5.0, I have added several enhancements to the dialog module. I am going to present them in several emails, as I get the documentation properly updated. There is now a new function called dlg_bridge(From, To, outbound_proxy). It can be used to connect From and To users. The mechanism is a simple call transfer scenario. kamailio => INVITE (hold) =>From kamailio <= 200ok <= From kamailio => REFER (To) =>From kamailio <= 200ok <= From ... From => INVITE ... =>To ... kamailio => BYE =>From kamailio <= 200ok <= From The function can be called from configuration file or via MI interface. It opens the door for several cute features you can do with the sip server. Here are couple of usage examples: - easy integration with web/external applications for Click-To-Dial. Doing it with kamctl will look loke: kamctl fifo dlg_bridge sip:from@kamailio.org sip:to@kamailio.org - wake up/reminder/scheduled call - use rtimer module to run scheduled calls - you can connect your user to a media server playing a message, to a conference room or where the user sets its preference - call again - when a user is calling a busy destination, when that destination becomes available, the server can initiate the call to connect the caller with the destination Documentation is available at: http://kamailio.org/docs/modules/1.5.x/dialog.html Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 4

[Kamailio-Users] Reg. SIP conferencing support for RFC 4579
by Padmaja R 04 Feb '09

04 Feb '09

Hi all, Is it possible to use Openser as the SIP conference server? I understand that Openser cannot do the media mixing required of a centralized media server but if it can do the centralized signaling by acting as a focus and let a media proxy do the media mixing, that would be good. I need ad hoc conferencing capability. Is there any media proxy that can achieve this together with openser? My scenario is as follows:- Say A and B are in a SIP call, and A and B suddenly decide to have a conference with C. A puts B on hold and dials C' s number. After call between A and C is up, if A presses hook flash, this sends an invite to the conference application namely: conf@openseratlocalhost:5060 (which is the ip and port number at which my openser proxy would be listening). The server should then send back 200 Ok with the contact header containing the isfocus" parameter of the form -Contact: <sip:conference1@mediaproxyip;isfocus> The isfocus is an RFC 4579 based feature. The 200 Ok responses to other invites (normal calls that do not create conference) should not include this isfocus in the contact header of 200 Ok. Finally, the openser proxy should be able to contact the media proxy on receiving an invite to create the conference and send that ip and focus information in the 200 Ok. The UA -A would then send a REFER to B and C to refer to the media server and they disconnect with A. Then all three UAs would individually send invite to the uri returned in the contact header and join the conference. Thanks, Padmaja

4 4

[Serusers] SER and media server
by Chatrapathi. H. Rahul 04 Feb '09

04 Feb '09

Hi I am using solaris 9v. I tried to use SEMS as media server with SER. SEMS is not compatible with Solaris 9v. Gives compilation errors. My query is whether any other opensource Media Server is available that uses SER and compatible with solaris 9v.This would help me a lot. Thanks in advance. -- Regards Rahul

1 0

[Kamailio-Users] FOSDEM Meeting - Feb 7, 2009, 19:30, Le Mess
by Daniel-Constantin Mierla 04 Feb '09

04 Feb '09

Hello, the place for the social networking event at FOSDEM 2009 has been selected: Le Mess http://www2.resto.be/mess/home-en.htm Louis Schmidtlaan - Av. Louis Schmidt 1 1040 BRUXELLES (ETTERBEEK) It is in walking distance from FOSDEM site. More details at: http://www.kamailio.org/mos/view/News/NewsItem/FOSDEM-2009---Social-Network… If you haven't confirmed your participation, you can still join, subject of available free places in the restaurant. See you Saturday! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] New module : PURPLE - IM/Presence multi-protocol gateway
by Eric PTAK 04 Feb '09

04 Feb '09

Hello everyone, Let me introduce to you a new module, available in trunk since yesterday, so it will be included in 1.5.0 release : >From http://kamailio.org/docs/modules/devel/purple.html: *Purple module is a multi-protocol instant-messaging gateway module which use libpurple, the library on bottom of Pidgin (ex-gaim).* *This plugin was wrote in order to proof the concept of gateway between SIP and MSN/Live networks.* *[...]* *It's no really adapted to a production environment with a huge user count but can be used in order to demonstrate the concept or to use it in SOHO environment.* I hope you'll enjoy purple module, and I'll welcome all feedbacks. Eric.

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Re: [Kamailio-Users] sip trace & duplicate_uri
by Daniel-Constantin Mierla 03 Feb '09

03 Feb '09

On 02/02/2009 09:35 PM, kokoska rokoska wrote: > > [...] >>> Hello, >>> >>> i have increased debug to 6 and I can not see errors in syslog. >>> Messages are traced, stored into DB, but not duplicated. >>> I'm at 1.4.3 on CentOS 5 - if it makes sense... >>> >>> Thank you very much, Daniel, for your interest! >>> >>> >>> >> ok, checked the source code. For some reason I do not recall now, when I >> wrote the module the duplication is done only when traced_user_avp is set. >> http://kamailio.org/docs/modules/devel/siptrace.html#id2452997 >> >> Set that avp and try again. If works I will try to fix and make it work >> without avp. >> >> Thanks, >> Daniel >> >> >> > > > Thank you very much, Daniel, for your help! > > It works :-) > > BTW: My idea is to offload proxy from (blocking ?) DB operatinos and > still have all possible logs => I think I can duplicate all messages to > another kamailio server and store them there to another DB. Is it > doable? With sip trace you still get them in a DB, because it first stores to DB and then sends to another address. However, what you want to achieve is something good, so for the future I will have in mind the option just to replicate to another address and skip writing to db. > Or, better to say, can I trace and duplicate messages on one > server and not log them? > If you want to duplicate the messages that come to kamailio, you can use the send() function in route and onreply_route block. http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:devel#send_destinat… If you want the siptrace module to do it, you have to hack it a bit. Practically is to introduce a new parameter, e.g., skip_db_storage, and put the code that writes in db inside a if not skip_db_storage. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 4

Re: [Kamailio-Users] [Kamailio-Devel] New module : PURPLE - IM/Presence multi-protocol gateway
by Eric PTAK 03 Feb '09

03 Feb '09

I agree with you, the module needs improvements on that point. It's a really old point I missed when we decided to reuse and to reverse that module. I think about two manners in order to solve it : * implement a return code for purple_send_message * sending a reply directly in the module Regarding the design of the module, I think both possibilities needs a lot of mod. I'm not sure how to consider that mod as it's not a bug but a limitation. Is it a new feature I'll code when the trunk will be unfreezed ? If yes, we may add that limitation in the doc for now. Eric. 2009/1/29 Iñaki Baz Castillo ibc(a)aliax.net > Why is needed to reply "200"? doesn't this module act as a proxy > waiting for a response from downstread (i.e. MSN network) and mapping > that response to the appropiate SIP code? > > This is: if I send a MESSAGE to a MSN account not online (or not > existing) I will receive "200" anyway, shouldn't I receive a 480 or > 404? > > Regards. > > > -- > Iñaki Baz Castillo > <ibc(a)aliax.net> > > _______________________________________________ > Kamailio (OpenSER) - Devel mailing list > Devel(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel > http://lists.openser-project.org/cgi-bin/mailman/listinfo/devel

4 7

[Kamailio-Users] wiki documentation index for kamailio
by Henning Westerholt 03 Feb '09

03 Feb '09

Hello all, thanks to Frederick Bullik we've now a nice documentation index for the Kamailio 1.4.x and 1.5.x releases available. You can easily lookup all module exports like functions [1] or module parameters [2]. The index is sorted alphabetically and provides convinient links to jump directly to the respective place in the module documentation. You'll find the new index in our wiki at: http://kamailio.org/dokuwiki/, as well as the old ones for past OpenSER releases. Best regards, Henning Westerholt [1] http://www.kamailio.org/dokuwiki/doku.php/modules:1.5.x:index-functions [2] http://www.kamailio.org/dokuwiki/doku.php/modules:1.4.x:index-parameters -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany I'm going to FOSDEM

2 1

[Kamailio-Users] [Fwd: Re: sip trace & duplicate_uri]
by kokoska rokoska 02 Feb '09

02 Feb '09

Daniel-Constantin Mierla napsal(a): > Hello, > > On 02/02/2009 08:08 PM, kokoska rokoska wrote: >> >> Daniel-Constantin Mierla napsal(a): >> >>> Hello, >>> >>> do you get any error in the syslog? >>> >>> Cheers, >>> Daniel >>> >>> >> >> >> Hello, >> >> i have increased debug to 6 and I can not see errors in syslog. >> Messages are traced, stored into DB, but not duplicated. >> I'm at 1.4.3 on CentOS 5 - if it makes sense... >> >> Thank you very much, Daniel, for your interest! >> >> > > ok, checked the source code. For some reason I do not recall now, when I > wrote the module the duplication is done only when traced_user_avp is set. > http://kamailio.org/docs/modules/devel/siptrace.html#id2452997 > > Set that avp and try again. If works I will try to fix and make it work > without avp. > > Thanks, > Daniel > > Thank you very much, Daniel, for your help! It works :-) BTW: My idea is to offload proxy from (blocking ?) DB operatinos and still have all possible logs => I think I can duplicate all messages to another kamailio server and store them there to another DB. Is it doable? Or, better to say, can I trace and duplicate messages on one server and not log them? Best regards, kokoska.rokoska

1 0

[Kamailio-Users] sip trace & duplicate_uri
by kokoska rokoska 02 Feb '09

02 Feb '09

Hi all, could someone be so kind and explain me how to use (or how exactly works) duplicate_uri in siptrace module? I think, if I setup modparam("siptrace","duplicate_uri","sip:1.2.3.4:5060") that all (and what is it "all"? :-) traced messages will be copied to defined sip uri (1.2.3.4:5060). I see nothing "on the wire" with ngrep, but all messages are succesufly inserted into local DB so I thing the siptrace module works. But doesn't copy messages to 1.2.3.4:5060 :-) Any hitns are really welcome! Best regards, kokoska.rokoska

2 3

[Kamailio-Users] store a double in avp
by ALAEDDINE abbech 02 Feb '09

02 Feb '09

hi all, can i store a double in avp? i try to store a double from my DB in the avp, so i have alwyse null. who have a solution for this pb. thanks

2 2

[Kamailio-Users] FOSDEM 2009 - Kamailio (OpenSER)/SIP-Router Social Networking Event
by Daniel-Constantin Mierla 02 Feb '09

02 Feb '09

Hello, a bunch of us will be at FOSDEM 2009: http://fosdem.org/2009/ Therefore is the opportunity for first Social Networking Event in 2009. Everyone interested in Kamailio (OpenSER), SIP Router, SER, SIP and not only are invited to join -- participation is free, everybody pays for its expenses. The event is very likely to be a dinner on Saturday evening, Feb 7, 2009. So far, from Kamailio (OpenSER) community you can meet there: - Henning Westerholt - http://www.kamailio.org/mos/view/Management/ - Elena-Ramona Modroiu - http://www.kamailio.org/mos/view/Management/ - Daniel-Constantin Mierla - http://www.kamailio.org/mos/view/Management/ - Dan Bogos - http://itsyscom.com/ In order to know properly the size of the place we need, please drop me an email if you intend to attend. People knowing Brussels are welcome to suggest a nice place to meet (good beer and food are a must). Looking forward to meeting you in Brussels. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 1

[Kamailio-Users] No UDP Address already in use?
by mayamatakeshi 31 Jan '09

31 Jan '09

Hello, If I start a second instance of kamailio by mistake, if the cfg file instructs it to listen to tcp ports, the second instance will exit with something like this: tcp_init: bind(7, 0x81b28ec, 16) on 127.0.0.1:5060 : Address already in use But if the cfg file only instructs kamailio to bind to udp ports, the second instance will not exit. Is this a bug? I'm curious because if we start kamailio and a udp port is in use by another application (not kamailio), kamailio will exit with udp_init: bind(4, 0x819f274, 16) on 192.168.88.29: Address already in use Why is that so? regards, mayama

3 4

[Serusers] Modified "To" header on REGISTER
by caio 30 Jan '09

30 Jan '09

Hello, Does save() method of registrar module use original sip message or can it use a modified one like for example by the subst() method of textops module? For example: A coming "To" header of the a REGISTER is modified and then the save() is called: Code: # REGISTER subst_uri('/^(.*)sip:(ip_address)(.*)$/\1sip:mydomain\3/i'); subst('/^To:(.*)sip:(.*)@(ip_address)(.*)$/To:\1sip:\2\@mydomain\4/i'); save("location"); Log: Jan 12 10:09:34 [/usr/sbin/ser] subst_run: matched (0, 17): [sip:200.2.127.187]_ Jan 12 10:09:34 [/usr/sbin/ser] textops: subst_uri_f: match - old uri= [sip:ip_address], new uri= [sip:mydomain]_ Jan 12 10:09:34 [/usr/sbin/ser] subst_run: running. r=0_ Jan 12 10:09:34 [/usr/sbin/ser] subst_run: matched (172, 40): [To: "579997" <sip:579997@ip_address>_]_ Jan 12 10:09:34 [/usr/sbin/ser] textops: subst_f: replacing at offset 208 [To: "579997" <sip:579997@ip_address>_] with [To: "579997" <sip:579997@mydomain>_]_ Cheers.. Caio

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