- sr-users - kamailio.org

Support
by Murali Krish 18 Dec '12

18 Dec '12

Hi this is Murali krishnan i am final student of master degree in Dalhousie University, I am Working on my project, i m need of a sip server which can support H.264 codec video calls...I created an account in iptel sip account when i used the account it doesn't support video calls. If i use SER will this support video calls? -- *Regards Murali Krish*

2 1

From sip phone to provider trunk
by andre second 18 Dec '12

18 Dec '12

Hi,I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running. What I want to do is to put some of the calls directly from the phones to SIP Provider without involving asterisk. I think I need to use Dispatcher module - what is the best way of doing that?Thank you!

2 4

p_usrloc registration update problem
by Hans Goossens 18 Dec '12

18 Dec '12

Hi, I'm testing the p_usrloc module (Kamailio version 3.3.2, installed via ubuntu package) and it looks like it's working but for some reason a re-registration is causing a new entry being added to the distributed location table instead of updating the existing one. The kamailio configuration looks like this: #!define DBURL "mysql://openser:openserrw@localhost/openser" loadmodule "p_usrloc.so" modparam("p_usrloc", "db_mode", 3) modparam("p_usrloc", "write_db_url", DBURL) modparam("p_usrloc", "read_db_url", DBURL) modparam("p_usrloc", "write_on_db", 1) modparam("p_usrloc", "alg_location", 1) modparam("p_usrloc", "reg_db_table", "locdb") mysql> select * from locdb; +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ | id | no | url | status | errors | failover | spare | rg | +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ | 1 | 1 | mysql://root:secret@172.16.10.131/openser | 1 | 0 | 1900-01-01 00:00:01 | 0 | 0 | | 1 | 2 | mysql://root:secret@172.16.10.132/openser | 1 | 0 | 1900-01-01 00:00:01 | 0 | 0 | +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ When using the default usrloc functionality everything is working fine. The only difference I noticed is that the default usrloc is using MYSQL UPDATE to update the location entry and that p_usrloc is using MYSQL REPLACE. Regards Hans

1 0

variable in rewritehost ?`
by Ali Jawad 17 Dec '12

17 Dec '12

Hi I am using slqops to get some user related info from location table, I want to use that info in rewritehost, however when I do use a variable in rewritehost I get an error "Unable to resolve variable". Is it possible to use a variable in rewritehost ? If not what is the workaround ? Thanks

3 9

Dialog module not decrementing profile size
by Grant Bagdasarian 17 Dec '12

17 Dec '12

Hello Community, I've recently started to use the Dialog module to manage the amount of incoming calls from our SIP provider allowed to pass through by using profiles_with_values and setting the size of these profiles to a certain amount. Most of the time this works as it should, but sometimes the 'slots' available for a profile aren't cleared even though a dialog has finished. This seems to happen when an high amount of regular traffic is hitting kamailio. Has anyone else experienced this? Are there perhaps any other ways to control the slots available for certain incoming SIP traffic? Below parts of my configuration file. Perhaps I'm doing something wrong which causes this to happen. In my route[WITHINDLG] -> loose_route -> is_method("BYE") I'm not calling dlg_manage(), could this be the problem? Other comments are welcome too, as I'm fairly new to Kamailio. modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "hash_size", 4096) modparam("dialog", "profiles_with_value", "318005004"); modparam("dialog", "default_timeout", 43200) modparam("dialog", "dlg_match_mode",2) modparam("dialog", "detect_spirals", 1) request_route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE")) { xlog("L_INFO", "Incoming request for $rU from $fU"); if ($rU == "318005004") { route(DIALOG); } } # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); route(RELAY); } route[DIALOG] { dlg_manage(); record_route(); $var(SIZE) = 0; sql_query("vf", "exec dbo.SelectAvailableSlots '00$rU'", "ra"); $avp(s:limit) = $dbr(ra=>[0,7]); xlog("L_INFO", "Maximum simultaneous calls is configured at $avp(s:limit) ...."); sql_result_free("ra"); if ($rU == "318005004") { get_profile_size("318005004", "$rU", "$var(SIZE)"); } xlog("L_INFO", "There are currently $var(SIZE) calls for $rU, excluding the current call"); if($var(SIZE) >= $avp(s:limit)) { xlog("L_INFO", "Simultaneous calls limit of $avp(s:limit) reached for $rU: $var(SIZE)\n"); sl_send_reply("603", "Simultaneous calls limit reached"); exit; } if ($rU == "318005004") { set_dlg_profile("318005004", "$rU"); } if ($rU == "318005004") { if(get_profile_size("318005004", "$var(SIZE)")) { xlog("L_INFO", "There are currently $var(SIZE) calls for $rU, including the current call"); } } } route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #Used for call transfer if($fU == "318005004") { if (is_method("UPDATE")) { sl_send_reply("405","Method not allowed"); xlog("L_INFO", "Dropping UPDATE message...\n"); exit; } if (is_method("INVITE")) { record_route(); } if (is_method("BYE")) { dlg_manage(); } } if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } Regards, Grant

2 8

status of kamailio stun server?
by Juha Heinanen 16 Dec '12

16 Dec '12

i tried to use kamailio built-in stun server, but failed, because it supports only binding requests. my ua tried to send it "allocate request lifetime" request, which failed with 600 global failure response. what is the status of kamailio's stun server? is anyone interested in developing it further or should we obsolete the server and start using some external server? -- juha

2 5

Writing User-preferences Username into Radacct Table on Call Forward
by Jonathan Hunter 15 Dec '12

15 Dec '12

Hi All, I was wondering if anyone would be kind enough to let me know if they have implemented anything like as follows in Kamailio? We are looking to implement call fowarding, activating it as usual in usr_preferences table. I was wondering if on the outbound leg its possible to write the username from usr_preferences with CFW active into the radacct table easily? This is with Radius and opensips DBs in Mysql. Many thanks Jon

2 2

NAT Traversal :: RTPProxy and Jitsi client
by Raj Roy Ghandhi 14 Dec '12

14 Dec '12

Hi, I am trying to communicate 2 Jitsi clients in 2 separate private networks. Both clients are behind NAT. Kamailio Server is on Public IP. Both Jitsi clients does register and presence works fine. And also text messages works well. But unable to call each other. Had some one had the same issue before. I tried with many soft clients and all failed with same result. Same scenario works with IP Phones. Can anybody point out what I miss there. Regards, Roy.

2 2

OT: MSRP over WebSocket Javascript stack
by Peter Dunkley 14 Dec '12

14 Dec '12

Hi, Crocodile has just open-sourced our MSRP over WebSocket (see http://tools.ietf.org/html/draft-pd-msrp-websocket) Javascript stack. The project is hosted on Google Code: http://code.google.com/p/crocodile-msrp/ The stack is distributed using the MIT License and was developed and tested along side the Kamailio MSRP over WebSocket implementation. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 2

RTP Proxy Interaction
by Austin Einter 14 Dec '12

14 Dec '12

Dear All I am looking at Kamailio's module, file, function that receives rtp and rtcp port from rtp proxy. Can somebody please let me know which file/function I need to check. Thanks Austin

2 1

Re: [SR-Users] Help
by Peter Dunkley 14 Dec '12

14 Dec '12

Hi Roy, Please keep emails on list. Regards, Peter On Fri, 2012-12-14 at 09:31 -0800, Raj Roy Ghandhi wrote: > Dear Peter, > > Thanks for all the support that you gave me during the Kamalio > configuration with RTPProxy. > Currently it works fine with IP Phones but not soft clients. (Jitsi) > > > If I give you my server information would you be able to check that > out for me. > > > I was lost for that issue. > > > Best Regards, > Roy. -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Local call routing accross many kamailio servers
by Ali Jawad 14 Dec '12

14 Dec '12

Hi I do have 3 kamailio servers sharing the same db, calls go to Asterisk and that works great, I did purchase a DID and point it to one of the kamailio servers, it works fine so far for that kamailio server as the call is forwarded to the softphone ..but ONLY if the user is registered on the kamailio server the DID is forwarded to..if the kamailio user is on another kamailio server it does not work..although the database and location tables are shared. Simple diagram DID Provider <=== DID Kamailio ===> Kamailio Servers A, B, C If a softphone registers on DID Kamailio and I call the DID it works, if a softphone registers on Kamailio A or B or C, I never get the call, looking at ngrep the call just stays on DID Kamailio and it does not try to send it to any other server, how can I make the Kamailio DID lookup the destination from the location table and send the call there ? Regards

2 9

Hardware requirements voip servers
by Grant Bagdasarian 14 Dec '12

14 Dec '12

Hello, We are considering upgrading our VoIP platform and I'm wondering what kind of hardware we need for our requirements. The architecture will consist of a SIP Proxy Server (Kamailio) in front and a cluster of Media/Application Servers behind it. The requirements are based on a single server. SIP Proxy Server: - Up to 100 - 300 calls per second. - Up to 5000 - 10000 concurrent calls. - RTP Proxy - Record-Route Media/Application Server: - Up to 100 - 300 calls per second. - Up to 5000 - 10000 concurrent calls. What do we need in terms of CPU, Memory, HDD when looking at the following specific requirements: - 100 CPS - 200 CPS - 300 CPS Which hardware components are critical for handling this many calls, is it CPU, Memory, Disk IO bound? Also which open source media/application server software is suitable for these requirements? We are currently using Asterisk, which works great, but perhaps there are others available which are more suitable for these requirements. Thanks in advance! Regards, Grant Bagdasarian

1 0

Kamailio <-> Asterisk MWI
by Daniel Tryba 13 Dec '12

13 Dec '12

I'm trying to get the MWI indicator on endpoints to work by redirecting the SUBSCRIBE to the voicemail server (asterisk) using usc_replace_(to|from). This works fine for the initial request but the next subscribe will fail since the to/from don't get rewritten, which is indicated by the error: ERROR: uac [replace.c:250]: decline FROM replacing in sequential request in auto mode (has TO tag) This occurs when the modparam "uac","restore_mode" is manual or auto. When set to none the redirect will be correct but all responses have to be rewritten manually (I'm a beginner and I don't want to do this just yet). And since uac has knowledge about this having being rewritten, sending the original from/to looks like like a bug to me (more probable a misconfiguration/omission of something crucial withing dialog responses :) Changes to the (default debian package) config, in route route[PRESENCE] in if(is_method("SUBSCRIBE")) if( is_method("SUBSCRIBE")) { if(search_hf("Event", "message-summary", "a")) { if (!www_authenticate("$fd", "subscriber")) { www_challenge("$fd", "1"); exit; } if(avp_db_query("SELECT value FROM usr_preferences WHERE username='$au' and attribute='voicemail' ORDER BY value limit 1")) { $avp(voicemail)=$avp(i:1); $avp(subscrfrom)=$fu; avp_subst("$avp(subscrfrom)", "/sip:.*@/sip:$avp(voicemail)@/"); uac_replace_from("$avp(subscrfrom)"); uac_replace_to("","sip:$avp(voicemail)@$sel(cfg_get.voicemail.srv_ip): $sel(cfg_get.voicemail.srv_port)"); remove_hf("Authorization"); $ru = "sip:" + $avp(voicemail) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); } } else { handle_subscribe(); t_release(); } } exit; 192.168.34.226 is the sip device, 192.168.32.40 kamailio, 192.168.32.41 the voicemailserver. Initial request: U 192.168.34.226:5852 -> 192.168.32.40:5060 SUBSCRIBE sip:grandstream@sip.local SIP/2.0. From: "Daniel" <sip:grandstream@sip.local>;tag=67d8f67065827fe2. To: <sip:grandstream@sip.local>. Gets rewritten to: U 192.168.32.40:5060 -> 192.168.32.41:5060 SUBSCRIBE sip:0880100782@192.168.32.41:5060 SIP/2.0. From: "Daniel" <sip:0880100782@sip.local>;tag=67d8f67065827fe2. To: <sip:0880100782@192.168.32.41:5060>. The response gets relayed to the endpoint and MWI will light up since there is a message. But next subscribe update from the endpoint triggers the "decline FROM replacing in sequential request" error and thus the subscribe send to voicemail is the original to/from: U 192.168.32.40:5060 -> 192.168.32.41:5060 SUBSCRIBE sip:0880100782@192.168.32.41:5060 SIP/2.0. From: "Daniel" <sip:grandstream@sip.local>;tag=67d8f67065827fe2. To: <sip:grandstream@sip.local>;tag=as324de8f3. Resulting in a 404 since the to/from are not the real peername on the voicemail server. How do I force Kamailio to rewrite the from/to without doing this manually for a requests/responses (restore_mode none)? The easy way out of this problem would be to have a 1 to 1 relation for the Kamailio subscribers and voicemail peers. But uac_replace should work (somehow). Attached is a full capture of the subscribe/notify. Running 3.3.2+squeeze from http://deb.kamailio.org/kamailio/. Machine is also running topoh module (192.168.0.1). -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

2 6

NAT for Load Balancer and Proxies
by Nguyen Anh Tuan 13 Dec '12

13 Dec '12

Hi everyone, I have 1 Load Balancer and 2 SIP Servers, all of them are Kamailio. Does anyone suggest how to add NAT for them? Best Regard, Nguyen Anh Tuan

2 1

SIP proxy changing Contact header of presence server requests
by Owen Lynch 13 Dec '12

13 Dec '12

Hi, we have 2 separate instances of kamailio running as a SIP proxy and presence server. The proxy uses the nathelper module to correctly change the Contact headers of client requests. However it also does the same for notify requests from the presence server, which results in incorrect RURI from subsequent client subscribes. The NATDETECT route is shown below, I plan to leave the existing nat_uac_test but also explicitly skip the fix_nated...() calls for requests from the presence server. Does this seem reasonable? # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } Thanks, Owen Lynch

2 1

Dispatching to Asterisk cluster
by Grant Bagdasarian 12 Dec '12

12 Dec '12

Hello, We have a cluster of three Asterisk machines. Each machine answers an incoming call and transfers it to unique remote destination. So, Asterisk01 transfers to Destination01, Asterisk02 to Destination02, Asterisk03 to Destination03. The Asterisk machines are allowed to receive and answer only 50 calls each. I have a Kamailio in front of these three machine. It is currently used to dispatch calls to the three machines using round-robin. But I need some way to equally dispatch a maximum of 50 calls to each device in the Asterisk dispatcher set. I was thinking of using the dispatcher module and using the 10 algorithm, to load balance based on call load. But I'm not sure if that will guarantee a maximum of 50 calls being dispatched to each Asterisk. Theoretically, if I set the ds_hash_size to 7, which is 128 slots, at some point two of them should have 43 calls and one 42 calls. Also the value of ds_hash_size should be the power of two. Are decimal values allowed, like 7,229 which equals to almost 150? Perhaps I should use the ratelimit module for this? I'd appreciate it if someone could give me some advice on this. Regards, Grant

2 2

Syntax of xavp
by Andreas Granig 12 Dec '12

12 Dec '12

Hi, I'm playing around with xavp, but there are some things I can't wrap my head around. What basically works is this: $xavp(a=>foo) = 'foo'; $xavp(a[0]=>bar) = 'bar'; Getting the value like this: xlog("L_INFO", "a-foo='$xavp(a=>foo)' and a-bar='$xavp(a=>bar)'"); as well as xlog("L_INFO", "a-foo='$xavp(a[0]=>foo)' and a-bar='$xavp(a[0]=>bar)'"); ... works, because when omitting the index, it assumes [0]. I really like to avoid such implicit assumptions though, to make the config really clear, so I prefer the second approach. What doesn't work for me then is explicit assignment of the very first value, like this: $xavp(a[0]=>foo) = 'foo'; $xavp(a[0]=>bar) = 'bar'; If I do this, the "a" xavp is never created. Wouldn't it be good practice to create the xavp on first assignment? That way, I don't have to take care in the config file whether the xavp is used for the first time, where I have to omit the index. What I'm also wondering is whether it's possible to directly access nested xavps, like this: $xavp(a=>foo) = 'afoo'; $xavp(b=>foo) = 'bfoo'; $xavp(c=>a) = $xavp(a); $xavp(c=>b) = $xavp(b); xlog("L_INFO", "a-foo='$xavp(c=>a[0]=>foo)'"); or xlog("L_INFO", "a-foo='$xavp(c=>a=>foo)'"); Both ways give me "a-foo='<<xavp:0x7f0d387fe178>>'". Is this even intended? Would be really cool if that's possible! Thanks, Andreas

3 4

NAPTR, SRV and sips vs. transport=tls
by Andreas Granig 12 Dec '12

12 Dec '12

Hi, Hope to get some guidance here over the usage of "sips" and "sip plus transport=tls" when following RFC3263. The RFC3263 says that a NATPR record could return something like this for a query like "host -t NAPTR example.com": ; order pref flags service regexp replacement IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.example.com. IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.example.com IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.example.com. This means that the client should use sips if possible when contacting example.com. Concluding from that, I suppose it should perform an SRV lookup "host -t SRV _sips._tcp.example.com", and the result would be: ;; Priority Weight Port Target IN SRV 0 1 5061 server1.example.com IN SRV 0 2 5061 server2.example.com What I'm curious about is how requests towards one of these servers should look like, and how they are being handled by kamailio. A lot of clients and servers are sending "sip:user@example.com;transport=tls" in request URIs and Contact headers and Record-Route headers, and you can check with uri_param("transport","tls") which transport socket to use. This is pretty useful as you can determine hop-by-hop whether or not to use TLS. This approach has been obsoleted by RFC3261 though, and there doesn't seem to be a mechanism in RFC3263 to indicate "use schema sip, but use transport=tls". On encounter of a NAPTR record like the one above, how does kamailio act? Does it set a "sips" schema for the next hop? And what's the general take on this "sips" schema? As far as I understand RFC3261, it means that if a client sends a request to a sips-URI, the request is sent to the domain via TLS, and from there "the request is sent securely to the callee, but with security mechanisms that depend on the policy of the domain of the callee." (RFC3261, Chapter 4). What does this really mean in practice? Are you allowed to rewrite the schema to "sip" and pass it on for example via UDP to the callee if the callee didn't indicate transport=tls (deprecated anyways) or "sips:" in the Contact of the registration? Or should you keep "sips" as schema, but still send it via UDP, because you know based on local policy or based on client registration that the next hop is not supporting TLS? How would widespread clients react when getting a call to a "sips" URI, especially if they receive it via UDP? Looking forward hearing your input on that, Andreas

5 14

MySql error
by Hermes F. 11 Dec '12

11 Dec '12

I'm trying to create MySql database using the following /usr/local/sbin/kamdbctl create I followed the instructions on http://kb.asipto.com/kamailio:install:3.3.x-from-git-centos5x but I receive the following error message: "ERROR: database engine not specified, please setup one in the config script" Can someone help me please ?

2 1

Regarding Kamailio Registrar
by AMIT SHAH 11 Dec '12

11 Dec '12

Dear all, We are using Kamailio SIP Server in our application. We want to know 1 thing: We are using Kamailio SIP server as registrar and proxy server. For registration database, we are using dbtext module. We maintain record of SIP Extensions who have registered in Our proprietary file too. Now when we boot the Kamailio SIP server, At that time our file should be empty i.e. previously registered Extension entries should be deleted or it must be kept as it is. But if during power off, any extension had sent REGISTER request to Unregister then it was not executed as kamailio SIP server was not ON. But at boot time, as its entry was present, Kamailio SIP server will Consider it a registered extension. Please guide us for this situation. Thank you, Amit Shah.

2 1

Unable to set custom RPID
by Mino Haluz 11 Dec '12

11 Dec '12

Hi, what things should I set in order to set my custom RPID. What I am doing: remove_hf("Remote-Party-ID"); append_rpid_hf("", ";party=calling;id-type=subscriber;screen=no"); but RPID is only removed, second command does not append any rpid. Thanks Mino

2 1

New developer: Konstantin Mosesov
by Daniel-Constantin Mierla 10 Dec '12

10 Dec '12

Hello, I want to announce that a new developer got GIT write access to repository: Konstantin Mosesov - he has contributed patches to app_python and joins the team to help maintaining and developing this module. His git commit id is: ez My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

How to use LIMIT 1 in db_query api
by Krishna Kurapati 10 Dec '12

10 Dec '12

Hi, I need a way to limit the results to just one row. The only way I know I could do is to call db_query followed by fetch_result of 1 row. However this is not optimal as all the rows are fetched to dynamic memory with db_query. Is there an alternative approach? Thanks Krish Kura

2 1

Help
by ameneh pourmoghadas 07 Dec '12

07 Dec '12

Dear Admin, My name is Ameneh from India. I am new in SIP. I have installed asterisk 1.8 on my ubuntu system and could have registered the internal users in my LAN to talk over x-lite. Now my question is how can I make a call to a mobile in India or a PSTN number in another country that would require a public SIP I think. how can I make my SIP server as public? would you please help me? thanks and regards, Ameneh

7 6

CTL module and alternatives for remote execution mi commands.
by Pavel Zalunin 07 Dec '12

07 Dec '12

Hi. In our project we need to query current users, dialogs and ability to end dialog using some rpc. Requirements are - do this remotely. I already tried to use xmlrpc module. When we had 42 users online it returned 17kb broken xml file. Then I tried to use binrpc protocol from CTL module. Usage ends with returning broken message too (in case, when message is biger than 21kb). Are there any other more stable solutions to query mi information using some tcp protocol? P.S. When I discovered sources of CTL module I found usage of writev - here is the problem about partial packets, when buffer is longer, than 21kb it don't pushes all vector buffers into socket. And in general, this module is using async sockets, AFAIK writev don't gives any guarantees, that it will write all vectors in async sockets. P.P.S. Of couse, last resort solution will be writing proxy for using mi_fifo on kamailio side to forward all records to other servers, but if there are more pretty solutions, please give me know:) Thanks. Pavel.

2 2

OFFTOPIC - Realtime testers needed - Asterisk Path header support (Oolong branch)
by Olle E. Johansson 07 Dec '12

07 Dec '12

While this is not Kamailio - it's related. If you want to use Kamailio in front of Asterisk, you do need SIP Path header support in Asterisk to do it right. Especially if you have one Kamailio handling Internet-facing communication and one handling internal communication. Please test! /O Vidarebefordrat brev: > Från: "Olle E. Johansson" <oej(a)edvina.net> > Ämne: Realtime testers needed - Path header support (Oolong branch) > Datum: 7 december 2012 11:07:09 CET > Till: Asterisk Developers Mailing List <asterisk-dev(a)lists.digium.com> > Kopia: "Olle E. Johansson" <oej(a)edvina.net> > > Friends, > > I have taken Klaus Darillion's patch for Path header support in Asterisk and created branches for 1.4, 1.8 and trunk. I need help testing it with realtime support by someone who are more used to the realtime databases than I am. I also need help completing the LDAP schemas with the new attribute. > > General testing is also encouraged. > > I have been using this patch without realtime for a couple of months. So far no Squirrels where hurt in the progress. And no bugs hit me in the back. > > Read more about the patch here: > http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/R… > > Klaus' original review: > https://reviewboard.asterisk.org/r/991/ > > The SIP Path header support is documented in RFC 3327 (that has a strange title). > > Basically, this is used when you have a proxy between a phone and your Asterisk. Asterisk needs to remember the path to the device when sending an INVITE, so we save both the contact and the path (basically a route) to the phone when Asterisk receives a REGISTER. When sending an INVITE (or anything based on the registry) a route header is added and the message finds the proper way to the device. > > Path header support exists in the path module in Kamailio if you want to test. > http://kamailio.org/docs/modules/3.3.x/modules_k/path.html > > Looking forward to your feedback! > > /O

1 0

Kamailio\ Message 300 Contacts
by Fatima Chahrour~Vanrise Support 06 Dec '12

06 Dec '12

Dears, In the redirect message 300, is there any way to control the maximum number of contacts(IP's) returned in the message? (am only getting 5 IP's) Also, if I have one gateway with 5 IP's and trying the same call 3 times; the order of the IP's returned in the message 300 is different in each time. Why is that? Thanks in advance, F Chahrour

2 2

Kamailio 3.X.X + BLF configuration trouble
by Hemanshu Patel 06 Dec '12

06 Dec '12

Hello everyone, Till now we were working with asterisk to handle BLF. But then we decided to move everything to kamailio and hence i started working on BLF with kamailio. I have already configured kamailio for Call handling and presence information. >From different pages on Internet, I have found some information about configuration of kamailio+blf. And i did so, but somehow things are not working as they should. I have done following changes to kamailio.cfg file loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_mwi.so" loadmodule "dialog.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.so" modparam("presence", "fallback2db", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", "mysql://xxxxxxxxx:xxxxxxxxxxxxxx@localhost/openser") modparam("dialog", "db_mode", 1) modparam("pua", "db_url", "mysql://xxxxxxxxx:xxxxxxxxxxxxxx@localhost/openser") Now when my phone(yealink VP530), sends subscribe request it looks as below: SUBSCRIBE sip:1001@172.16.27.66 SIP/2.0 Via: SIP/2.0/UDP 172.16.27.61:5063;branch=z9hG4bK10464096 From: "1000" <sip:1000@172.16.27.66>;tag=393923564 To: <sip:1001@172.16.27.66> Call-ID: 1572901975(a)172.16.27.61 CSeq: 1 SUBSCRIBE Contact: <sip:1000@172.16.27.61:5063> Accept: application/dialog-info+xml Max-Forwards: 70 User-Agent: VP530P 23.70.0.40 Expires: 120 Event: dialog Content-Length: 0 In response to this message, Kamailio returns 202 with following information SIP/2.0 202 OK Via: SIP/2.0/UDP 172.16.27.61:5063;branch=z9hG4bK245273058 From: "1000" <sip:1000@172.16.27.66>;tag=393923564 To: <sip:1001@172.16.27.66>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8ee2 Call-ID: 1572901975(a)172.16.27.61 CSeq: 2 SUBSCRIBE Expires: 120 Contact: <sip:172.16.27.66:5060> Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Followed by a NOTIFY NOTIFY sip:1000@172.16.27.61:5063 SIP/2.0 Via: SIP/2.0/UDP 172.16.27.66;branch=z9hG4bK3ccd.adfbc416.0 To: sip:1000@172.16.27.66;tag=2058189864 From: sip:1001@172.16.27.66;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6cd8 CSeq: 1 NOTIFY Call-ID: 2040123403(a)172.16.27.61 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:172.16.27.66:5060> Subscription-State: active;expires=170 During subscribe message asked for XML based reply, yet NOTIFY doesnt send any XML. and hence there is no impact of this message on device. Nothing is changed.Also there's nothing when i call from this device to another, ideally i should get some NOTIFY messages So, i think i am doing something wrong in Configuration. Can anyone please help me in it. If you need any other info, i'll provide you. -- Regards, Hemanshu Patel Senior Software Engg P Help the environment – please don't print this email unless you really need to!

3 7

Internal/External SIP setup
by Zero Aggression 06 Dec '12

06 Dec '12

Hello, I am a bit new to VoIP, and I just need a quick yes/no (and maybe pointers) answer. Is the following possible (and smart) to do with Kamailio? Kamailio server. Dual-homed (internal network, Internet). Internal users connect to Kamailio to make calls to internal and Internet SIP peers. Users have NO internet connection themselves (they can only reach the server). Can Kamaillio (when dual-homed) transport all the data/voice between Internet and intranet? Can it also be used to carry all traffic between intranet users (if there is no direct route between them)? Thanks so much. Zane

3 3

db_cluster configuration
by Juan José Ivars 06 Dec '12

06 Dec '12

Hello, i am configuring db_cluster module in kamailio 3.3 I have configured in roun-robin read between two MySQL servers but the load is not distributed equally My configuration is that: modparam("db_cluster", "connection", "con1=>mysql://USER1:PASS1@1.1.1.1/DB") modparam("db_cluster", "connection", "con2=>mysql://USER2:PASS2@2.2.2.2/DB") modparam("db_cluster", "cluster", "cls1=>con1=9r9p;con2=9r9p") And the auth module with: modparam("auth_db", "db_url", "cluster://cls1") But i have enabled the mysql logs and the major part of the querys are targeted to the second server Is there any cache by peer? What am i doing wrong? Thanks in advance.

2 1

Preparing for the next major release
by Olle E. Johansson 06 Dec '12

06 Dec '12

Hi! Daniel has set a code freeze date for the next major release and the Kamailio community needs to work together to prepare the release. That means you as well! There will be a lot of work in the coming weeks to merge code from SIP router into the Kamailio core to complete the merger process. We also have a large set of new modules and functions to test. The creation of a new release is much more than code - it's testing, documentation proof-reading and feedback. With a large and complex piece of software like Kamailio we can't test all functions with all possible combinations of configurations and connections to third party products - this is where YOU can help out! Read more on http://www.kamailio.org/w/2012/12/preparations-for-new-release/ and join us in creating the new release! /O

1 0

Kamailio Load Balancing, problems ACK and BYE.
by Nguyen Anh Tuan 06 Dec '12

06 Dec '12

Hi everyone, I am a beginner in using Kamailio. I am currently building a load balancer for multiple SIP Servers, all of them are Kamailio. The Load balancer can help end-users REGISTER and INVITE normally, but has problems with ACK and BYE. Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK failed, is transmitted to the SIP Server then stop here, doesn't go back to Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because doesn't receive ACK, the Called continuously sent 200 call OK to Caller (sent successfully) and wait for confirmation ACK (after arrival to the SIP Server, ACK doesn't come back to Load Balancing). Because the Called party doesn't receive ACK, the call will be automatically disconnected after 30 seconds. With BYE have rather, if the Caller hangs up first, BYE is transferred over SIP Server and returned to Load balancer, but Load Balancing can't sent it to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side of called). Caller had picked off the phone, but the Caller is not received BYE and after 30 seconds the call will automatically disconnect (by ACK errors). If the called party hangs up first, HUNG UP button of the called even useless, and I don't see any packet of the Called party's BYE message in Wireshark. I use X-Lite and EyeBeam for two end users, the configuration of the SIP Server is the default configuration of Kamailio. DISPATHCHER.LIST: I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy 2. But for capturing packets more convenient, I turn off the Proxy 2 and 3. # group sip addresses of your * units # Kamailio SIP Proxies/Registrars 1 sip:192.168.3.11:5060 #1 sip:192.168.3.12:5060 #1 sip:192.168.3.13:5060 Previously I used the Stateless Load Balancing configuration by default, but could not INVITE: # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); record_route(); forward(); # t_relay(); } Then I changed the following, sent INVITE and setup call successfully, but met ACK and BYE error above. if ( is_method("INVITE")) { # Not Dispatch INVITE from SIP Proxies if ( ds_is_from_list("1") ) { t_relay(); } else if ( !ds_is_from_list("1") ) { route(DISPATCH); } } else { # If not INVITE route(DISPATCH); } } route[DISPATCH] { # dispatch destinations ds_select_dst("1", "0"); record_route(); t_relay(); #forward(); } Please tell me How to modify in the configuration of Proxy and Load Balancing? Thank so much for helping! Best Regard Nguyen Anh Tuan, -------------------------------------------------------------------- The latest 200 OK from Called after Ringing : LB->Caller Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55 cf46-1--d87543-;rport=9040 Record-Route: <sip:192.168.3.11;lr;nat=yes> Record-Route: <sip:192.168.3.10;lr=on> Contact: <sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 435 Caller's ACK (1001(a)192.168.3.88) to Load Balancer (192.168.3.10): Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-6f1c8659392a0773-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 ACK 192.168.3.10:5060 (Load Balance) to 192.168.3.11:5060 (Proxy) then STOP HERE (Not transfer further) Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-6f1c8659392a 0773-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 Caller's BYE (1001(a)192.168.3.88) to Load Balance (192.168.3.10) Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-616e232f86191a37-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE was transferred from Load Balance(192.168.3.10) to Proxy (192.168.3.11), repeated THREE TIMES, possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 69 Route: <sip:192.168.3.10;lr> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.1 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Send back BYE from SIP Proxy (192.168.3.11) to Load Balance (192.168.3.10) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE LOOPs Back from Load Balance(192.168.3.10) to Proxy (192.168.3.11), possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.33f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.23f49894.0 Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 66 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0

1 0

Kamailio Load Balancing, problems ACK and BYE.
by Nguyen Anh Tuan 06 Dec '12

06 Dec '12

Hi everyone, I am a beginner in using Kamailio. I am currently building a load balancer for multiple SIP Servers, all of them are Kamailio. The Load balancer can help end-users REGISTER and INVITE normally, but has problems with ACK and BYE. Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK failed, is transmitted to the SIP Server then stop here, doesn't go back to Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because doesn't receive ACK, the Called continuously sent 200 call OK to Caller (sent successfully) and wait for confirmation ACK (after arrival to the SIP Server, ACK doesn't come back to Load Balancing). Because the Called party doesn't receive ACK, the call will be automatically disconnected after 30 seconds. With BYE have rather, if the Caller hangs up first, BYE is transferred over SIP Server and returned to Load balancer, but Load Balancing can't sent it to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side of called). Caller had picked off the phone, but the Caller is not received BYE and after 30 seconds the call will automatically disconnect (by ACK errors). If the called party hangs up first, HUNG UP button of the called even useless, and I don't see any packet of the Called party's BYE message in Wireshark. I use X-Lite and EyeBeam for two end users, the configuration of the SIP Server is the default configuration of Kamailio. DISPATHCHER.LIST: I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy 2. But for capturing packets more convenient, I turn off the Proxy 2 and 3. # group sip addresses of your * units # Kamailio SIP Proxies/Registrars 1 sip:192.168.3.11:5060 #1 sip:192.168.3.12:5060 #1 sip:192.168.3.13:5060 Previously I used the Stateless Load Balancing configuration by default, but could not INVITE: # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); record_route(); forward(); # t_relay(); } Then I changed the following, sent INVITE and setup call successfully, but met ACK and BYE error above. if ( is_method("INVITE")) { # Not Dispatch INVITE from SIP Proxies if ( ds_is_from_list("1") ) { t_relay(); } else if ( !ds_is_from_list("1") ) { route(DISPATCH); } } else { # If not INVITE route(DISPATCH); } } route[DISPATCH] { # dispatch destinations ds_select_dst("1", "0"); record_route(); t_relay(); #forward(); } Please tell me How to modify in the configuration of Proxy and Load Balancing? Thank so much for helping! Best Regard Nguyen Anh Tuan, -------------------------------------------------------------------- The latest 200 OK from Called after Ringing : LB->Caller Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55 cf46-1--d87543-;rport=9040 Record-Route: <sip:192.168.3.11;lr;nat=yes> Record-Route: <sip:192.168.3.10;lr=on> Contact: <sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 435 Caller's ACK (1001(a)192.168.3.88) to Load Balancer (192.168.3.10): Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-6f1c8659392a0773-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 ACK 192.168.3.10:5060 (Load Balance) to 192.168.3.11:5060 (Proxy) then STOP HERE (Not transfer further) Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-6f1c8659392a 0773-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 Caller's BYE (1001(a)192.168.3.88) to Load Balance (192.168.3.10) Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-616e232f86191a37-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE was transferred from Load Balance(192.168.3.10) to Proxy (192.168.3.11), repeated THREE TIMES, possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 69 Route: <sip:192.168.3.10;lr> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.1 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Send back BYE from SIP Proxy (192.168.3.11) to Load Balance (192.168.3.10) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE LOOPs Back from Load Balance(192.168.3.10) to Proxy (192.168.3.11), possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.33f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.23f49894.0 Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 66 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0

1 0

Segmentation fault in sip-capture
by Owen Lynch 05 Dec '12

05 Dec '12

Hi, I am running Kamailio 3.3 as a Homer capture agent, the sip-capture module receives HEP packets from a sip-trace module in a remote kamailio instance. Periodically the capture agent dies. The log and core dump suggest the initial problem is an invalid HEP msg received, but kamailio crashes when trying to send an error reply. Can anyone shed any light on this? Thanks, Owen Lynch Back trace: Program terminated with signal 11, Segmentation fault. #0 0x0813c916 in send2child (tcpconn=0xb55c1c60) at tcp_main.c:3967 3967 tcp_main.c: No such file or directory. in tcp_main.c Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.80.el6.i686 keyutils-libs-1.4-4.el6.i686 krb5-libs-1.9-33.el6.i686 libcom_err-1.41.12-12.el6.i686 libselinux-2.0.94-5.3.el6.i686 mysql-libs-5.1.61-4.el6.i686 nss-softokn-freebl-3.12.9-11.el6.i686 openssl-1.0.0-20.el6_2.5.i686 zlib-1.2.3-27.el6.i686 (gdb) bt #0 0x0813c916 in send2child (tcpconn=0xb55c1c60) at tcp_main.c:3967 #1 0x08136e78 in handle_tcpconn_ev (fm=<value optimized out>, ev=<value optimized out>, idx=<value optimized out>) at tcp_main.c:4310 #2 handle_io (fm=<value optimized out>, ev=<value optimized out>, idx=<value optimized out>) at tcp_main.c:4362 #3 0x0813ea3c in io_wait_loop_epoll () at io_wait.h:1092 #4 tcp_main_loop () at tcp_main.c:4656 #5 0x080a502b in main_loop () at main.c:1727 #6 0x080a6c00 in main (argc=11, argv=0xbfeafd14) at main.c:2546 Log: Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: : <core> [pass_fd.c:288]: ERROR: receive_fd: recvmsg on 10 failed: Connection reset by peer Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: <core> [tcp_main.c:2347]: BUG: tcp_send: failed to get fd(receive_fd): Connection reset by peer (104) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: sl [../../forward.h:193]: msg_send: ERROR: tcp_send failed Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=704 a=3 n=exit Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: sipcapture [sipcapture.c:682]: ERROR: sipcapture:hep_msg_received: not supported version or bad length: v:[10] l:[10] vs [40] Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:634]: SIP Reply (status): Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:638]: status: <500> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:640]: reason: <Invalid CSeq> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 234, <received> = <yadayadayada>; state=6 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK2348.d1620f62.0>; state=16 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: ALERT: <core> [main.c:785]: child process 19576 exited by a signal 11 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: ALERT: <core> [main.c:788]: core was generated Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19575]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19574]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19573]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19572]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tls [tls_init.c:674]: tls module final tls destroy Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [mem/shm_mem.c:242]: shm_mem_destroy Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [mem/shm_mem.c:245]: destroying the shared memory lock Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [main.c:804]: terminating due to SIGCHLD

2 5

modules_s/eval - survey of usage
by Daniel-Constantin Mierla 05 Dec '12

05 Dec '12

Hello, is anyone using the modules_s/eval module? In the context of merging modules_s/* to modules/*, this one lacks the docbook xml files for readme. It has a text readme file, from where I understand it does script operations using a stack and polish notation expressions. Probably it was developed before the expressions support in config file language, at this moment all its operations can be done directly by the interpreter. I am considering moving the module to obsolete/ folder. If anyone is using it and wants to keep it in modules/*, then he/she has to convert the readme text into docbook format (which is not complex at all, just takes a bit of time, probably like 60 min or even less). If there is no answer in one week about usage, will be sent to obsolete/, from where it can be rescued in case of need (just to be clear for everybody that code is not lost). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

dns resolver issue (RFC3263)
by MÉSZÁROS Mihály 05 Dec '12

05 Dec '12

Hi All, I am experiencing an issue when i try to contact xy(a)cisco.com. I found that kamailio/sip-router can't resolve by default resolving way a TCP + SRV records from domain cisco.com. e.g. cisco.com misi@alma:~$ host -t NAPTR cisco.com cisco.com has no NAPTR record misi@alma:~$ host -t SRV _sip._udp.cisco.com Host _sip._udp.cisco.com not found: 3(NXDOMAIN) misi@alma:~$ host -t SRV _sip._tcp.cisco.com _sip._tcp.cisco.com has SRV record 1 0 5060 vcsgw.cisco.com. misi@alma:~$ host -t SRV _sips._tcp.cisco.com _sips._tcp.cisco.com has SRV record 1 0 5061 vcsgw.cisco.com. I can't call xy(a)cisco.com because it does not exists an NAPTR record in domain cisco.com, and furthermore no SRV with udp. But it exists sip+tcp record and also exists an secure sip SRV, so sips+tcp record! I read rfc3263 I find kamailio dns resolver is not working as it should according RFC3263. http://tools.ietf.org/html/rfc3263 _If no NAPTR records are found, the client constructs SRV queries for those transport protocols it supports, and does a query for each._ So kamailio should query all (udp, tcp, tls, sctp whatever) protocols. Can anyone help/guide me to create a fix to this issue? My plan is to create a patch to kamailio resolver, to correct and behave according RFC3263. Any help or guidance appreciated! Thanks, Misi

4 14

syslog message about invalid domain name
by Juha Heinanen 05 Dec '12

05 Dec '12

when i t_relay in-dialog bye: Request-Line: BYE sip:0x8eb94d0@[2002:c062:660a::1]:5050;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP [2002:C062:660A:0:0:0:0:1];branch=z9hG4bKde53.6aaed77ea908953bbb133e5021d74036.0 Via: SIP/2.0/UDP [::1]:5090;received=0:0:0:0:0:0:0:1;branch=z9hG4bKq~ouJaGs;rport=5090 From: <sip:jh@as.test.fi>;tag=5A29C828-50BEFE55000C7118-B4FF9B70 To: "" <sip:test@test.fi>;tag=92e26ffbfbc16e53 CSeq: 10 BYE Call-ID: 28d1e4872cc5f83e User-Agent: Sip Express Media Server (1.6.0 (x86/linux)) Max-Forwards: 69 Content-Length: 0 with $du set to sip:[2002:C062:660A:0:0:0:0:1]:59933;transport=tcp, i get to syslog: Dec 5 09:53:48 sip /usr/sbin/sip-proxy[1947]: INFO: <core> [resolve.c:729]: invalid domain name '[2002:C062:660A:0:0:0:0:1]' since everything is and works as expected, why is the info message produced and how can i get rid of it? -- juha

2 2

Xcap server chooice among openSER, mobicents and openXCAP
by zhi sun 05 Dec '12

05 Dec '12

hi guys, it is my first time to use XCAP, i want to implement XCAP server to support maybe huge amount of user accesses (as cloud solution), so far, i can find out three XCAP server solution: 1. openSER with XCAP Server Module http://kamailio.org/docs/modules/devel/modules_k/xcap_server.html 2. mobicents with XCAP component/server build-in http://code.google.com/p/mobicents/wiki/Welcome?tm=6 http://www.mobicents.org/diameter/index.html http://hudson.jboss.org/hudson/view/Mobicents/job/Mobicents-Books/lastSucce… 3. openXCAP http://www.ag-projects.com/projects-products-96/101-openxcap but i am not clear to choose with solution to best match my situation (high volumn, high scale, and quick response). - does openSER support the cloud solution with high volum, high scale? - is it easy to use/integrate the XCAP server independently from mobicents? - is there performance issue for openXCAP, because it is implemented using python? any suggestions would be appreciated! best regards, -iamsy

1 0

io_wait errors
by Juha Heinanen 05 Dec '12

05 Dec '12

any idea what could be the cause for lots of these kind syslog messages: Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 56 Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 56 in the hash(-1, 0, 0xb2f28adc) flags 0) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 59 in the hash(-1, 0, 0xb725a9d8) flags 0) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 58 on 59: Broken pipe (32) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [tcp_main.c:4003]: ERROR: send2child: send_fd failed for 0xb3024a48 (flags 0x4018), fd 58 -- juha

2 2

[Kamailio-Users] BYE message not relayed to caller
by Vikram Ragukumar 04 Dec '12

04 Dec '12

Hello, My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy. I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like : No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route: <sip:IPaddr_proxy:7160;lr=on;nat=yes> CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" <sip:1234@IPaddr_VoipSwitch>;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0 The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up. Any suggestions on how i can get around this problem ? Thanks and Regards, Vikram.

3 9

openser_receiver
by Uri Shacked 04 Dec '12

04 Dec '12

Hi, I notice that under my /tmp folder there are openser_receiver filles as well (kamailio_fifo and kamailio_ctl are know....) Version is 3.3.2 latest branch. Any ideas? Thanks, Uri

1 0

Problem with Stateless Dispatcher Call Load Distribution (Algorithm 10)
by SIP Mailing-list 04 Dec '12

04 Dec '12

Hi guys, I wonder if someone could sanity check something for me. I currently have a simple load balancer set up using algorithm 10 (call load distribution) from the dispatcher module. I'm using Record-Routing so that I can clear down the call load record at the end of the call. Everything looked like it was working okay until I just spotted a problem that I think might be a bug. It's to do with ACKs to non-200 responses. In a normal call, the flow goes: UAC Proxy UAS 1 INVITE --> 2 INVITE--> 3 <-- 200 OK --- ds_load_update() 4 <-- 200 OK 5 ACK --> 6 ACK --> Now between step 3 and 4 the proxy runs ds_load_update which according to the documentation "set internal state to confirmed for the call load" entry in the internal call state store. This means the proxy knows where to send the ACK at step 5 because it is safely in the call state store so you can use ds_select_dst again on the ACK to get it to the right place. Works fine! Now consider the non-200 case: UAC Proxy UAS 1 INVITE --> 2 INVITE--> 3 <-- 603 Decline --- ds_load_unset() 4 <-- 603 Decline 5 ACK --> 6 ACK --> ??? Between step 3 and 4 the proxy is supposed to run ds_load_unset() which unconditionally removes the call from the call state store. Now when the ACK comes in at step 5, the proxy has no record of the call and so doesn't know where to send the ACK. This results in retried Declines and ACKs. Broken :-( If the proxy doesn't run ds_load_unset() on the 603 response, is there some kind of timer that will cause the call to be removed from the call state store on unconfirmed calls? I don't think I can run ds_load_unset on the ACK because there's nothing in the ACK that tells me it is because of a 603, rather than a 200. Edit: actually there is! The Route header is present on the 200 ACK, but * not* on the 3xx to 6xx ACK. Maybe I can use loose_route... Anyway, any thoughts would be appreciated! Cheers, Richard

2 4

default flavour set to kamailio
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, based on community interaction, development, package distributions and documentation maintenance in the recent years, the default flavour has been switched now to 'kamailio'. As written in my previous email, we started the last phase to complete the integration between kamailio-ser, aiming to get to a unique database schema and set of modules in maximum 6 months. Kamailio is the most spread and popular flavour, presented in many linux/unix distributions, being the one released and packaged periodically. So it will stay as main flavour, the one officially released by the project. Those that want can take the sources and build their own flavour, as the support in Makefile system is kept. Once the integration is finished, the flavour system will just set the application name, nothing else. From the open source project perspective, the resources will be allocated only on releasing under the main/default flavour (Kamailio). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Last phase of integration for kamailio - ser
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, we plan to start the last phase of integration between kamailio and ser. After about 4 years with same source code tree, there is still some overhead caused by several duplicated modules. Following the decision to go on with the most developed components and absorb the missing features from the duplicates, kamailio modules are those that will stay (move to modules/ eventually) and get updates when needed. We used same decision to keep the core and makefile system from ser, adding to it what was extra in kamailio. I present next a summary of what is planned to happen: 1) Modules from modules_s/ that will be simply moved to modules/ because there is no conflict at all: - avp, avp_db, db_ops, eval, mangler, print, print_lib 2) Modules from modules_s/ that will be renamed and moved modules/ because it is mainly only conflict of name, but not in the backend (e.g., database): - auth_db => auth_dbattrs - because it uses (attribute,value) style to store subscriber profiles, which can be a preferred alternative for many out there - ldap => db2_ldap - it is a DB APIv2 connector to ldap server - bdb => db2_bdb - it is a DB APIv2 connector to berkeley db (to be double-checked) - xlog => xprintf - it uses %spec to print log messages with all the code inside the module - the only reason not discarding it and stick only to modules_k/xlog for the moment is because it is used from other modules, otherwise modules_k/xlog has more features (e.g., syslog facility, color printing, a.s.o) 3) Modules from modules_s/ that will be removed, after absorbing some extra features: - usrloc/registrar => get the ability to store AVPs per contact (modules_k is more advanced in features, like path/gruu/partial outbound support, publish presence states, etc.) - several modules must be reviewed: acc*, radius*, permissions, pike, gflags, options, timer 4) Modules from modules_s/ that will be removed, by moving to obsolete/ directory - the rest of them from modules_s/, like dialog, xcap, cpl-c, presence_b2b, dispatcher, dbtext, domain, etc. The categories and their content are not nailed down, it is the purpose of this discussion to see if something was overlooked. We hope to end this process in max 6 months time - after that will be a unique database schema and a unique set of modules. Flavour system in the Makefile will stay just for letting people choose easier the name of binary, otherwise everything will be the same (e.g., what is compiled, installed, etc). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v3.3.2 Released
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, Kamailio SIP Server v3.3.2 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.1. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0 or v3.3.1. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.2. For more details about version 3.3.2 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/10/kamailio-v3-3-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

4 4

sample SCA kamailio.cfg added
by Andrew Mortensen 04 Dec '12

04 Dec '12

I've added a basic working kamailio.cfg with Shared Call Appearances to the SCA docs. It should help clarify where and when it's necessary to have the sca module inspect traffic. Those of you simply wanting to test SCA can just copy and paste the configuration, updating the listen address as necessary. Until the sca module doc on kamailio.org gets updated, you can find the sample configuration in the git master branch, or view it through gitweb at the link below. Best, andrew <http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=mo…>

1 0

kamailio vs.ser
by Ovidiu Sas 03 Dec '12

03 Dec '12

Hello all, By inspecting the source code, the only difference that I could see between kamailio and ser flavours is that kamailio has support for the "tm:local-request". Are there any constrains in having the tm:local-request present for ser flavour? Does it make sense to continue to build two flavours? My suggestion would be to focus one one single flavour and go forward with it for the next release. Comments? Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

2 1

health check
by Alex Solt 03 Dec '12

03 Dec '12

Hi, Is there a way to perform health check on kamailio application? AS

2 1

LCR Failover
by Daniel W. Graham 03 Dec '12

03 Dec '12

I am having issues with LCR failure route. I have two gateways specified in LCR table, if I change the address of the gateway with highest priority to a bogus IP, the failure route never seems to takes place and call is never routed to second gateway. Also, I have tested successful calls to both gateways by switching the priorities around. Here is what I receive after the first gateway times out: U kamailio:5060 -> asterisk:5060 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK633b44d8;rport=5060. From: "user" <sip:user@asterisk>;tag=as67c0037f. To: <sip:dialednumber@domain>;tag=d7177f97ed67076dddc3163c85ab0656-f2e5. Call-ID: 248d69ed6495d0a65315458e2ba6c528@asterisk:5060. CSeq: 102 INVITE. Server: SIP Proxy. Content-Length: 0. Config: modparam("tm", "fr_timer", 10000) modparam("tm", "fr_inv_timer", 120000) modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") route[LCR] { if (!load_gws(1)) { sl_send_reply("503", "Internal Server Error, Unable to load gateways"); exit; } else { if (next_gw()) { t_on_failure("LCR_FAIL"); route(RELAY); } else { sl_send_reply("503", "Service Unavailable, No available gateways"); exit; }; }; exit; } failure_route[LCR_FAIL] { if (next_gw()) { t_on_failure("LCR_FAIL"); route(RELAY); } else { t_reply("503", "Service Unavailable, No gateways"); exit; }; exit; }

2 7

Kamailio World - Conference & Exhibition - April 16-17, 2013, Berlin, Germany
by Daniel-Constantin Mierla 03 Dec '12

03 Dec '12

Kamailio project is announcing its first dedicated event - Kamailio World - to take place in Berlin, Germany, during April 16-17, 2013. The event is organized by Asipto (http://www.asipto.com) in collaboration with Fraunhofer Fokus Institute (http://www.fokus.fraunhofer.de). The conference is targeting to present commercial products and services build around Kamailio and related projects, as well as technical workshops of how to use Kamailio for various purposes. Along with the conference will be space for exhibition, that will allow companies to make demos and show cases for participants. For more details, visit the web site of the event: http://conference.kamailio.com Besides Kamailio, you will have the chance to interact with other open source VoIP projects such Asterisk, FreeSwitch, SIP Express Media Server, Homer SIP Capture, Siremis, a.s.o., as well as with commercial vendors in real time communications market. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Jitsi to Jitsi call
by Raj Roy Ghandhi 03 Dec '12

03 Dec '12

Hi Peter, Still I am unable to make call to Jitsi to Jitsi. Call established But no media. But text message works. NAT traversal work with RTPProxy and I can do IPPhone to IPPhone call and voice works very well. Tried with many soft phones and voice did not go through. Did I miss something in the configuration ? Please advice me. Best Regards, Roy.

2 2

Call pickup through a NAT
by Mark Boyce 03 Dec '12

03 Dec '12

Hi All, I've hit my next stumbling block on my mission to get BLF/Call pickup working on Linksys/Cisco phones which are behind a NAT working with Kamailio. I'm using RFC 3265 and pua_dialoginfo to handle subscribes from phones and generate notify packets on status change. The phones are on two different sites behind a couple of NATs. Kamailio is on public IP in the data centre. BLFs work fine. However the <target uri="x.x.x.x."> given in the XML of the notify packet is the that of the Public IP of the NAT so that call pickup fails. I'm guessing that this actually needs to be the address of the Kamailio server? I cannot see where to correct this without breaking tables the server needs to know how to contact the phone when it needs to. Has anyone seen this before? Cheers Mark

2 1

Kamailio SCA Module
by Robert Boisvert 01 Dec '12

01 Dec '12

Andrew, Congratulations on your new position as a Kamailio developer! When I saw your module (SCA) I began testing immediately since it may be a stepping stone to help me get the Polycom barge-in feature working with Kamailio. However, I ran into a few difficulties. Can you assist me? To make your existing code work as a K module I had to change a few lines of code and recompile under modules_k. Specifically, since the K definition of the ucontact structure defines ucontact->aor as str* and the S definition as str**, I changed “&c->aor” to “c->aor” in the following lines of sca_usrloc_cb.c: 73: if ( !sca_uri_is_shared_appearance( sca, c->aor )) { 74: LM_DBG( "%.*s is not a shared appearance line", STR_FMT( c->aor )); 79: &SCA_EVENT_NAME_CALL_INFO, c->aor ) < 0 ) { Since I use PostgreSQL as my SQL engine I created the SCA entries using this code: INSERT INTO version (table_name, table_version) values ('sca_subscriptions', '0'); CREATE TABLE sca_subscriptions ( id SERIAL PRIMARY KEY NOT NULL, subscriber VARCHAR(255) NOT NULL, aor VARCHAR(255) NOT NULL, event INTEGER NOT NULL, expires INTEGER NOT NULL, state INTEGER NOT NULL, app_idx INTEGER NOT NULL, call_id VARCHAR(255) NOT NULL, from_tag VARCHAR(128) NOT NULL, to_tag VARCHAR(128) NOT NULL, notify_cseq INTEGER NOT NULL, subscribe_cseq INTEGER NOT NULL, CONSTRAINT sca_subscriptions_idx UNIQUE (subscriber, call_id, from_tag, to_tag)); I applied the recommendations from your documentation on configuring Kamailio (3.3.1) and inserted the call to “sca_handle_subscribe ()” in kamailio.cfg. Next, I connected two Polycom phones (SP335 and SP650) to my network and configured them to enable presence and use a shared line. Four lines were configured on the phones as shown below: SP335 Line 1 (p11) = private line 6013 SP335 Line 1 (p12) = shared line 1519 SP650 Line 1 (p21) = private line 6014 SP650 Line 2 (p22) = shared line 1519 My tests get inconsistent results on the line LEDs and it seems that they do not work as a shared line. When I ran a tcpdump of the SIP communication (see attached pcap files) I found several anomalies: - Some packets (sca-reboot1 #22, 44, 113, 120 and sca-reboot2 #17, 38) contain spurious NUL characters in the tags section. In all tests it seems to occur in the Expire tag when Kamailio accepts the subscription. - When Kamailio sends call-info NOTIFY the Content-Length tag is missing. - Sometimes the phone does not respond to a call-info NOTIFY event ( sca-reboot1 #23, 45 and sca-reboot2 #18, 39). - I don't have an example here but sometimes the phones respond to the call-info NOTIFY with "481 Call Leg/Transaction Does Not Exist". I think some of my problems are related to these anomalies. Could you help me to determine if this is a problem with the code itself, my failure to properly port it to the K module, or some other issue? Once again, thank you for making this very useful module for Kamailio and for the documentation to make it work. I hope that resolving my problems may help the code to become a part of Kamailio. Sincerely yours, Bob Boisvert

2 1

Linphone presence
by Moacir Ferreira 30 Nov '12

30 Nov '12

I am trying to get Linphone to show presence information. I did my Kamailio setup using just the template from a stardard install and it is working properly with Bria (X-Ten). However, when using Linphone it does not show buddies presence. Any hint? Cheers! Moacir

2 1

Call forking
by Moacir Ferreira 30 Nov '12

30 Nov '12

All the topics/examples I found on the net is not what I am looking for, so here is the question: How do I achieve parallel forking (call all devices) if I have a single user registered using the same account from different devices (i.e.: table phone, iPad and PC)? What I want is to ring all devices a user may be using. Cheers! Moacir

2 1

t_fork_to?
by Juha Heinanen 30 Nov '12

30 Nov '12

core cookbook reads: append_branch Similarly to t_fork_to, it extends destination set by a new entry. The difference is that current URI is taken as new entry. what is t_fork_to? i could not find it in tm README. -- juha

3 3

WITH_PRESENCE charsh Kamailio
by Raj Roy Ghandhi 30 Nov '12

30 Nov '12

Hi, I am running Kamailio 3.4 with web-socket and everything works fine with NAT traversal for IP Phones. But when I enable the PRESENCE it crash with "(3) errors in config file" I have copied the config.cfg to paste-bin at http://pastebin.ca/2257817<http://pastebin.ca/2257815> Please guide me on this issue. Best Regards, Roy.

2 1

Enforced TLS, still some packets are through UDP
by Kamal Palei 30 Nov '12

30 Nov '12

Dear All I ma trying to have all SIP signalling through TLS. Using Kamailio 3.1.5. In a typical setup, endpoints initiates call, messages reach Kamailio proxy, Kamailio proxy forwards these packets to main proxy. In Kamailio.cfg, I have added below code. route { # per request initial checks route(REQINIT); setflag(FLT_NATS); if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.19:443"); exit(); } else { xlog("incoming request\n"); route(WITHINDLG); route(RTPPROXY); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); t_relay_to("tls:115.114.48.19:443"); exit(); } .............. I am finding most of the messages are getting transmitted in TLS. But few ACK and REFER / REFER response packets are still being transported in UDP. Can somebody tell me whats the wrong I am doing here. By any chance route(WITHINDLG); route(RTPPROXY); before t_relay_to("tls:115.114.48.19:443"); can cause this ???? And one more doubt.., 115.114.48.19 I get in route header. How can I extract the value of route header, so that I do not have to hard-code Many thanks in advance. -kamal

3 2

Lua env not initialized in event_route
by Vicente Hernando 30 Nov '12

30 Nov '12

Hello I'm trying to execute a Lua script on kamailio startup. I'm calling the script inside event_route[hable:mod-init], but it fails to be executed with this error: [app_lua_mod.c:167]: Lua env not initialized It happens both with lua_run and lua_dofile. I've played with the order in wich the modules are loaded, but I haven't been able to get rid of that message. Kamailio version is 3.3.0. Do you think this is feasible or should I try a different solution? Thanks in advance Regards, Vicente.

2 1

Presence on Cisco/Linksys handsets
by Mark Boyce 29 Nov '12

29 Nov '12

Hi All, I've been trying to get some Cisco/Linksys SPA phones working with Kamailio (current stable). All seems to be ok apart from getting Presence/BLF/SCA working. The phones are set to Server_Type : RFC3265_4235 Line Key 2 Extension : Disabled Line Key 2 Share Call Appearance : Private Line Key 2 Extended Function : fnc=blf+sd+cp;sub=1001@ourserver.com;ext=1001@ourserver.com Where 1001 is the other phone. The phone appears to be sending event type 'call-info' rather than dialog etc. Phone is being sent '489 Bad Event' and the error in the logs is : presence [subscribe.c:1007]: Unsupported event header field value call-info Whatever I set the Server Type to on the phones it still tries to send call-info events on subscribes Has anyone got Presence working with Cisco SPA handset? The module needed seems to be available on OpenSIPs http://www.opensips.org/html/docs/modules/1.8.x/presence_callinfo.html Thanks! Mark

3 8

Kamailio getting started
by Scott, Matt 29 Nov '12

29 Nov '12

Anybody got a good tutorial or starter project for Kamailio? We have Siremis 3 installed, but there is 0 help on that? Trying to solve 2 scenarios: -asterisk to softphone through kamailio -softphone to asterisk through kamailio. I ultimately want to use Kamailio as an edge router Any help is appreciated. Matt Scott

3 3

Kamailio HA with Virtual IP and multiple Asterisk servers for dispatching
by Rumen Mihailov 29 Nov '12

29 Nov '12

Hi all, Please bear with me as this is my first post to a mailing list ever so something might be quite wrong. My question is: How do we make Kamailio HA ? For the moment I’ve setup MySQL cluster with MySQL proxies that provide 1 Virtual IP that I use to connect Kamailio to (for usrloc and dispatcher) I have a couple of asterisk servers that Kamailio servers are supposed to send calls to. I have two kamailio servers that share the same IP address with keepalived. Both Kamailio servers are bind to the Virtual IP. The problem is: When Kamailio 1 is active calls are correctly dispatched to both the asterisk servers correctly. When Kamailio 2 is active calls are only sent to one of the asterisk servers. ‘kamctl dispatcher show’ shows the correct servers on both machines. The two Kamailio servers share EXACTLY the same config files. Do you guys have any thoughts on this ? p.s. Thank you for creating such a wonderful product. Thank you !!! Best regards, Rumen

2 13

Call from SipML5 -> PSTN
by Raj Roy Ghandhi 28 Nov '12

28 Nov '12

Hi, >From all of your support now I can call from 1. IP Phone <--> IP Phone 2. Web Page <--> Web Page 3. IP Phone -> PSTN without any issue But when I try to call from Web Page to PSTN then it tries to call sip:00xxxxxx89078@mysipdomain.com and that time out. Trying to figure out how to get this work ? Can anybody guide me on this please. Best Regards, Roy.

6 7

Re: [SR-Users] CRLF from Kamailio server
by Klaus Darilion 28 Nov '12

28 Nov '12

forwarded to mailing-list .... -------- Original Message -------- Subject: Re: [SR-Users] CRLF from Kamailio server Date: Mon, 26 Nov 2012 16:34:26 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: miconda(a)gmail.com To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Hello, I haven't checked actually the parts for tcp, but recently I added the feature of removing the location records if there is no reply to OPTIONS keepalives in nathelper, and it was only for UDP. Also, I think the nat ping is 4 bytes (zeros) or so. Might be another option from tcp, like: http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tcp_keepalive Cheers, Daniel On 11/26/12 3:54 PM, Klaus Darilion wrote: > IIRC tcp_crlf_ping is only about the Ping-Pong mechanism: client sends > PING=CRLF, server responds with PONG=CRLFCRLF. > > Thus, server should only response on client CRLF. Further the PONG is > always double-CRLF. > > So I suspected it is NAT keepalive - maybe someone added > NAT-keep-alive for TCP :-) > > regards > Klaus > > On 26.11.2012 11:05, Daniel-Constantin Mierla wrote: >> Hello, >> >> this might be actually related to tcp connection keepalive: >> >> http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tcp_crlf_ping >> >> IIRC, nathelper pings only udp contacts. >> >> Cheers, >> Daniel >> >> On 11/26/12 10:54 AM, Klaus Darilion wrote: >>> >>> >>> On 26.11.2012 05:13, Dmytro Bogovych wrote: >>>> Greetings. >>>> May anyone give me advice? >>>> I have working Kamailio with TLS. >>>> It is used by iOS softphone - i developed own one based on >>>> resiprocate. >>>> All signalling is TLS. >>>> Almost everything is fine. >>>> >>>> But looks kamailio sends CRLF messages sometimes. >>>> If it happens when softphone is in background mode - it is killed >>>> by iOS >>>> (as iOS prohibits too often awakes from background mode). >>>> >>>> How I can disable these messages? >>>> I see natping_interval = 30 and ping_nated_only=0 >>>> Should I set natping_interval to zero ? >>> >>> Either set the interval to 0 or set ping_nated_only=1 and make sure to >>> not set the NAT flag for the iphone clients. >>> >>> A problem may be firewalls/NAT between the iPhone and Kamailio, as >>> they may drop the TCP connection is there is no activity for too long >>> time. >>> >>> regards >>> Klaus >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Call from SipML5 -> PSTN
by Peter Dunkley 28 Nov '12

28 Nov '12

You need some sort of media gateway or server. I don't know of any that currently support this. Peter Raj Roy Ghandhi <roy.gandhi(a)gmail.com> wrote: >_______________________________________________ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

memcached loading problem
by Alex Balashov 28 Nov '12

28 Nov '12

Hello, So, I built the memcached module against the latest Git production branch of memcached and libmemcache-1.4.0.rc2 (the latest available), on Debian i386. But when I try to load the Kamailio memcached module: Nov 28 03:47:44 diminuendo-1 kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/memcached.so>: /usr/lib/libmemcache.so.0: undefined symbol: mcm_buf_len I find that very odd, because the symbol is quite clearly exported in the library: diminuendo-1:~# nm /usr/lib/libmemcache.so.0 | grep -i mcm_buf_len U mcm_buf_len Does anyone have any idea what the underlying linking issue is here? -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 4

Websocket with Kamailio 3.2.2 release
by Raj Roy Ghandhi 28 Nov '12

28 Nov '12

Hi, I am trying to integrate the websocket module into release 3.2.2. But I am unable to do that. Please guide me to get it done. Regards, Roy.

6 17

Eavesdropping SRTP sessions
by Mino Haluz 28 Nov '12

28 Nov '12

Hi, maybe it is not that kamailio related question, but I dont know any other place with such good voip professionals ;) I have kamailio and mediaproxy. Clients are BudgetTone 200 (Grandstream) and CSipSimple. I am forcing clients to use SRTP but it does not support adding any certificate on both sides. SRTP call is working fine. The question is, in this case, is man-in-the-middle attack possible? Maybe I should study SRTP more, but basically, if there are no certificates, there is no method how to be 100% sure that the media goes directly between clients. Is it true? Thanks for response, Mino

2 4

Managing expiration values with memcached
by Alex Balashov 28 Nov '12

28 Nov '12

According to the memcached module docs, setting $mctex(key) can be used to modify the expiration value of a key. Should this be done after or before setting the key itself with $mct(...)? I have tried it before and the assignment didn't work, so clearly the key must first exist in memcache: $mctex(bgn=>$avp(s:bgn)) = $sel(cfg_get.bgn_expiry); $mct(bgn=>$avp(s:bgn)) = $dbr(ra=>[0,0]); Nov 28 05:23:41 diminuendo-1 /usr/local/sbin/kamailio[32089]: ERROR: <core> [lvalue.c:354]: setting pvar failed Nov 28 05:23:41 diminuendo-1 /usr/local/sbin/kamailio[32089]: ERROR: <core> [lvalue.c:407]: assignment failed at pos: (164,37-164,69) OK, so that makes sense. I tried it the other way: $mct(bgn=>$avp(s:bgn)) = $dbr(ra=>[0,0]); $mctex(bgn=>$avp(s:bgn)) = $sel(cfg_get.bgn_expiry); But when I look on my memcached server, the expiration value is not being modified. It's still the default 'expire' value set in the memcached modparam. I ran the memcached server with maximum verbosity and verified that no TOUCH command is taking place, and that the SET command for the key contains the default expiration value, not the one I set with $mctex(...). So, it seems to me that the module is not committing this new expiration value when it is written to $mctex(...). Is this a bug, or am I missing something? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 2

problem with kamailio and libradiusclient-ng/freeradius-xs
by Rabary 27 Nov '12

27 Nov '12

Hi all, When I have this line in kamailio.cfg: modparam("acc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") Kamailio failed to start: kamailio error, failed to start (/var/run/kamailio/kamailio.pid exists). I already try to stop kamailio, an remove this file but don't work In log,I have: Nov 27 20:13:39 voip-core-1 kernel: [14318.988658] kamailio[8370]: segfault at 0 ip 00007f0b0b94ca5d sp 00007fffa08f9460 error 4 in libradiusclient-ng.so.2.0.0[7f0b0b946000+c000] Nov 27 20:13:39 voip-core-1 kamailio: ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe When I remove just this line and restart kamailio, it works, I don't know what is the problem because before I can do accounting with freeradius with the same config I have kamailio 3.3.2, kamailio-radius-modules 3.3.2,libradiusclient-ng2 0.5.6-1.1 and on 2nd server freeradius-xs and freeradius-xs-mysql 1.1.7-4 I put in attached file a few config of kamailio and radiusclient Regards -- Rabary Teddy Inutile d'imprimer ce mail

1 0

Disconnection from Kamailio
by Dmytro Bogovych 27 Nov '12

27 Nov '12

Greetings. I observe strange behavior with Kamailio. My iphone 3gs run softphone based on resiprocate. First 10-20 minutes everything works fine. After it the connection is closed from server (select() loop returns read notification, read attempt returns zero bytes). Ok, softphone closes old socket and create new one by new registration. But this connection is terminated from server in 4-5 seconds after registration finish (200 OK). Softphone makes next attempt to register - connection is closed after registration again. So it is infinite loop. After several attempts iOS kills application as it awakes too often in background mode. I have not logs from server yet; but maybe anyone had similar issue? Thank you :)

1 0

send 100 provisioning delayed
by pars3c 27 Nov '12

27 Nov '12

Hi , The proxy send the "automatic" provisioning message 100 "Trying" , near 1-2 seconds later the INVITE message, so the client send 3 times the same invite (correct , 500 ms timer). Is there any parameter to set , for the time between the INVITE and "100" message ? Is correct to disable the automatic send, and set after the invite, sl_send_reply("100", "Trying"); ? Thanks

2 1

Kamailio HA with Virtual IP and multiple Asterisk servers for dispatching
by Rumen Mihailov 27 Nov '12

27 Nov '12

Hi all, Please bear with me as this is my first post to a mailing list ever so something might be quite wrong. My question is: How do we make Kamailio HA ? For the moment I've setup MySQL cluster with MySQL proxies that provide 1 Virtual IP that I use to connect Kamailio to (for usrloc and dispatcher) I have a couple of asterisk servers that Kamailio servers are supposed to send calls to. I have two kamailio servers that share the same IP address with keepalived. Both Kamailio servers are bind to the Virtual IP. The problem is: When Kamailio 1 is active calls are correctly dispatched to both the asterisk servers correctly. When Kamailio 2 is active calls are only sent to one of the asterisk servers. 'kamctl dispatcher show' shows the correct servers on both machines. The two Kamailio servers share EXACTLY the same config files. Do you guys have any thoughts on this ? p.s. Thank you for creating such a wonderful product. Thank you !!! Best regards, Rumen

1 0

Kamailio crash
by Ricardo Martinez 27 Nov '12

27 Nov '12

Hello In the last three days my kamailio process has crashed two times. I obtained the core file and see it, this is the output. Can someone check what is the problem? Core was generated by `/usr/local/sbin/kamailio -m 1024'. Program terminated with signal 11, Segmentation fault. #0 core_hash (cmd_tree=<value optimized out>, param=<value optimized out>) at ../../lib/kcore/hash_func.h:56 56 v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3]; Missing separate debuginfos, use: debuginfo-install filesystem-2.4.30-2.1.el6.x86_64 glibc-2.12-1.25.el6_1.3.x86_64 keyutils-libs-1.4-1.el6.x86_64 krb5-libs-1.9-9.el6_1.2.x86_64 libcom_err-1.41.12-7.el6.x86_64 libselinux-2.0.94-5.el6.x86_64 mysql-libs-5.1.52-1.el6_0.1.x86_64 nss-softokn-freebl-3.12.7-1.1.el6.x86_64 openssl-1.0.0-20.el6_2.1.x86_64 pcre-7.8-3.1.el6.x86_64 zlib-1.2.3-26.el6.x86_64 (gdb) bt #0 core_hash (cmd_tree=<value optimized out>, param=<value optimized out>) at ../../lib/kcore/hash_func.h:56 #1 process_mi_params (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_hash.c:1057 #2 mi_terminate_dlgs (cmd_tree=<value optimized out>, param=<value optimized out>) at dlg_hash.c:1165 #3 0x00007fc789996370 in run_mi_cmd (rx_sock=11, tx_sock=12) at ../../lib/kmi/mi.h:77 #4 mi_datagram_server (rx_sock=11, tx_sock=12) at datagram_fnc.c:525 #5 0x00007fc789998509 in datagram_process (rank=<value optimized out>) at mi_datagram.c:346 #6 0x00007fc78999a09c in mi_child_init (rank=<value optimized out>) at mi_datagram.c:296 #7 0x00000000004dbd61 in ?? () #8 0x00007fc78d99aab8 in ?? () #9 0x00007fc78d0181cd in pkg_proc_update_real_used (data=<value optimized out>) at pkg_stats.c:141 #10 0x000000000052b423 in ?? () #11 0x00007fc78d8a2830 in ?? () #12 0x0000000000000000 in ?? () (gdb) I’m using kamailio [root@pxh ~]# kamailio -V version: kamailio 3.2.3 (x86_64/linux) 2a00c6-dirty with dialog module, hash tables, radius authorization. Hope someone could help me. Regards, Ricardo.-

2 8

plans for next major release
by Daniel-Constantin Mierla 27 Nov '12

27 Nov '12

Hello, following the devel IRC meeting from last week, I am starting the discussion about the next major release planning. There were two relevant options: 1) freeze by end of January (2013) and release by March 2) freeze by end of December (2012) and release by February Many devels announced quite some work ahead, so the 1st option seemed not to be enough for finishing intended work, which may add some extra delays (several estimated to finish by end of February or March). So the 2nd came into debate, to do an early release as many people seemed very interested in getting websockets support out in a stable release. Both of the options fit in our releasing policy (8 to 10 months), first one with potential to delays. We do have now lot of new stuff in devel branch, with other ongoing work planned to be pushed before end of year. A suggestion was to do the following major release a bit shorter, if it proves to collect a lot of additions quickly -- maybe by summer 2013. As not all devels were in the meeting, but also because it impacts the user community, I am looking to get some feedback on the mailing lists as well. That will allow to make the decision quicker based on showed interest. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

One active registration per sip instance.
by Krishna Kurapati 26 Nov '12

26 Nov '12

Hi, I have 3.x Kamailio with GRUU support. I see SIP Instance in the contact. However, I see a problem. When I kill the SIP client on a device and restart, it sends registration on a different port. The server instead of updating the contact in user location table, is creating a new contact for the same instance. Do I need to turn on something to get the expected behavior? Here is the output from kamctl ul show. Thanks Krish Kura AOR:: 100227074 Contact:: sip:100227074@10.132.60.70:50845;transport=TCP;ob Q= Expires:: 3132 Callid:: ATZJcr.HqXxwWdsPmanjERetcdeRVeZy Cseq:: 4399 User-agent:: n/a Received:: sip:166.137.120.37:48873;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b3602b-50e-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50839;transport=TCP;ob Q= Expires:: 3029 Callid:: oYsIYUyP3N77MKSoQ3EDChEph.lD4fjB Cseq:: 37489 User-agent:: n/a Received:: sip:166.137.120.37:61763;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b3602b-507-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50803;transport=TCP;ob Q= Expires:: 1527 Callid:: b4IeJXybn6Yr.jB8sc-Kz8CJ0kt0hNbO Cseq:: 30014 User-agent:: n/a Received:: sip:166.137.120.37:36879;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f5c-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50818;transport=TCP;ob Q= Expires:: 2199 Callid:: DM0Z1d-ZenGHknjmT.7Cle5cos7Uhs7H Cseq:: 22942 User-agent:: n/a Received:: sip:166.137.120.37:39471;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f6f-1 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1 Contact:: sip:100227074@10.132.60.70:50815;transport=TCP;ob Q= Expires:: 1833 Callid:: VJy3SNE66U1U971u5BEeqbYJi6a-puT- Cseq:: 64347 User-agent:: n/a Received:: sip:166.137.120.37:35872;transport=TCP State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Ruid:: uloc-50b35a12-7f5a-2 Instance:: <urn:uuid:237BF2F6-2FDE-47E8-A9E2-32D46D43F174-319-00000121AFA29D2E> Reg-Id:: 1

2 4

new developer: Andrew Mortensen
by Daniel-Constantin Mierla 26 Nov '12

26 Nov '12

Hello, I want to announce that a new developer got GIT write access to repository: Andrew Mortensen from University of Pennsylvania, USA - he has contributed a new module: sca (Shared Call Appearances for Broadsoft extensions). The module is right now in a personal branch, soon to me merged in master branch. His git commit id is: admorten My warm welcome and looking forward to future work within the project. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

ul.dump Problem
by Spencer Thomason 26 Nov '12

26 Nov '12

Hello, I'm attempting to dump the contents of the usrloc cache and I'm receiving the following error: sercmd> ul.dump error: 500 - Internal error creating aor struct sercmd> mi ul_dump error: 500 - Internal server error processing 's': buffer too small (overflow) (-2) It seems to work fine when there are only a small number of registrations (about 20 or so) but then fails when the number increases. It there a setting I need to increase somewhere? Thanks, Spencer

2 4

problem with $time Pseudo-Variables on 3.3.2
by Uri Shacked 26 Nov '12

26 Nov '12

Hi, I upgraded to 3.3.2 On 3.2 i used the variable $time() to get the current time for some process. Every 2 minutes a RTIMER process started, checked the $time(hour) and $time(min) and on a certain time executed something. This process stopped working on 3.3.2 When i debug, i see that the $time() variable keeps on giving me the same hour and minute that was given on the first run. Meaning the $time() variable does not give the current time. I notice also that the $Ts that on 3.2 gave a current time stopped doing it and $TS does it now.... Any help here? Is it OK and i need to check time another way? Or something in the $time() has changed...? Thanks, Uri

2 6

CRLF from Kamailio server
by Dmytro Bogovych 26 Nov '12

26 Nov '12

Greetings. May anyone give me advice? I have working Kamailio with TLS. It is used by iOS softphone - i developed own one based on resiprocate. All signalling is TLS. Almost everything is fine. But looks kamailio sends CRLF messages sometimes. If it happens when softphone is in background mode - it is killed by iOS (as iOS prohibits too often awakes from background mode). How I can disable these messages? I see natping_interval = 30 and ping_nated_only=0 Should I set natping_interval to zero ? Thank you :)

3 4

Re: [SR-Users] Forwarding Relaying of REGISTER Messages
by kellon＠sis-grenada.com 25 Nov '12

25 Nov '12

Hi All Alex responded to my question saying the following: "You can relay SIP messages of any type just the same way as you do with INVITEs -- e.g. t_relay(), forward(). But, proxying registration messages can have complications." can anyone give me some pointers or a list of some of these complications so that I can further research to see how its is done. I will attempt using the t_relay to view the issues, just need some idea more specifically what to look for! Thanks

2 1

Re: [SR-Users] Re-Directing of REGISTER Messages
by kellon＠sis-grenada.com 25 Nov '12

25 Nov '12

Hi Can anyone point to a good example of Kamailio performing the function of Re-direct server. We would like to forward messages onto another domain for authentication.

4 3

Difficulties with with loose_route() when trying to handle non-compliant ACK
by Richard Brady 24 Nov '12

24 Nov '12

Hi guys I have a multihomed Kamailio proxy sitting between two B2BUAs on separate networks and record-routing all dialogs. The problem I have is that when one of these devices receives a 200 OK, it does not populate the RURI of the ACK correctly. Instead of taking it from the Contact header on the 200 OK, it uses the user part from the Contact header and sets the domain to the proxy IP. It then also populates the Route headers. The ACKs are below and the IPs are: 10.152.1.92:5060: UAC on outside 10.200.70.100:5060: proxy outside interface 192.168.242.100: proxy inside interface 192.168.242.102: UAS on inside This the ACK message going into and coming out of the proxy. # U 10.152.1.92:5060 -> 10.200.70.100:5060 ACK sip:natted_ua*9197**192.168.242.102*5080*udp@10.200.70.100 SIP/2.0. Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK-2280-1-5. Route: <sip:10.200.70.100;r2=on;lr=on;nat=yes>,<sip:192.168.242.100;r2=on;lr=on;nat=yes>. From: sipp <sip:sipp@127.0.1.1:5060>;tag=2280SIPpTag001. To: sut <sip:9197@10.200.70.100:5060>;tag=p0FaeQ1QUS9ae. Call-ID: 1-2280(a)127.0.1.1. CSeq: 1 ACK. Contact: sip:sipp@127.0.1.1:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 10.200.70.100:5060 -> 192.168.242.100:5060 ACK sip:192.168.242.100;r2=on;lr=on;nat=yes SIP/2.0. Via: SIP/2.0/UDP 10.200.70.100;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=10.152.1.92;branch=z9hG4bK-2280-1-5. Route: . From: sipp <sip:sipp@127.0.1.1:5060>;tag=2280SIPpTag001. To: sut <sip:9197@10.200.70.100:5060>;tag=p0FaeQ1QUS9ae. Call-ID: 1-2280(a)127.0.1.1. CSeq: 1 ACK. Contact: <sip:sipp@127.0.1.1:5060;alias=10.152.1.92~5060~1>. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. So I am trying to understand why it is trying to relay to itself (I have mhomed=1) and why it is rewriting the RURI as if it is a strict router. Currently decode_contact() is disabled but enabling doesn't seem to help. Unless there is a very specific place where it belongs. Any advice would be hugely appreciated. I can always paste logs / configs / traces. Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

2 3

can not make calls with kamailio 3.3 and asterisk-10.7.0
by James zhu 24 Nov '12

24 Nov '12

hello, guys:i follow the linkhttp://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-a… can registered users, kamctl online shows those status. but i can not make calls. this is debug message when i make a call from 102 to SIP 103--------------------------------------------------------------- 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=486 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=3 n=return 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=489 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=636 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=601 a=25 n=has_totag 8(1286) DEBUG: siputils [checks.c:103]: no totag 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=501 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=26 n=is_method 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=496 a=25 n=t_check_trans 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [tm.c:1015]: lookup_original: t_lookupOriginalT returned: 0xb4d36320 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=497 a=25 n=t_relay 8(1286) DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=29 , global msg id=28 , T on entrance=(nil) 8(1286) DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=60763, isACK=0 8(1286) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(1286) DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found 8(1286) DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb4d3a698, callback type 1, id 0 entered 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [t_fwd.c:1243]: DEBUG: e2e_cancel: e2e cancel proceeding 8(1286) DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.1.103, 192.168.1.103, 0) 8(1286) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 8(1286) DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 8(1286) DEBUG: tm [t_reply.c:703]: DEBUG: reply sent out. buf=0xb721ee38: SIP/2.0 200 cancelin..., shmem=0xb4d38c08: SIP/2.0 200 cancelin 8(1286) DEBUG: tm [t_reply.c:713]: DEBUG: _reply_light: finished 8(1286) DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=3 n=exit 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list=================================anyone can give a hint for that? thanks! Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com

1 0

Using uac_req_send function
by Freddy C 22 Nov '12

22 Nov '12

Hello, I'm using uac_req_send () to send a request generated in Kamailio (eg Example 1.21. UAC module) $ uac_req (method) = "OPTIONS"; $ uac_req (ruri) = "sip: kamailio.org"; $ uac_req (furi) = "sip: kamailio.org"; $ uac_req (turi) = "sip: kamailio.org"; uac_req_send (); However, I need to capture the responses of this request with failure_route TM module. Is there a way to capture these responses when I'm using uac_req_send() function? I appreciate your comments. Freddy

2 1

Next IRC devel meeting
by Daniel-Constantin Mierla 22 Nov '12

22 Nov '12

Hello, I am thinking of having our next IRC devel meeting soon, to plan the next major release and review current stable releases and the environment around the project (e.g., if you can add anything else to the project to make life easier for developers and users). I'm proposing next week, Thursday, at 15:00GMT, on Freenode IRC server, channel #sip-router. Anyone able to join? If it is not a convenient date for you, just propose alternatives and we will select the one that meets the constraints of the most developers and users willing to participate. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 10

How does SIP transfer work?
by Grant Bagdasarian 22 Nov '12

22 Nov '12

Hello, I've been searching the internet to find an explanation on how SIP transfer works using Re-INVITE and/or UPDATE, but I can't seem to find a good source. >From what I understand(and this is the way we do it), the following happens: Bob=Caller Alice=Called John=Transfer party 1) Bob calls Alice. The usual INVITE,Trying,200 OK, ACK. 2) Alice transfers the call to John using Re-INVITE. a. Alice calls John. The usual INVITE,Trying,200 OK, ACK. b. Alice Re-INVITEs Bob using INVITE with adjusted SDP. 3) Bob is connected to John through Alice in some magical way. I'm guessing because the SDP has been changed and for some reason the RTP stream flows between Bob and John through Alice? Is this correct? If not, perhaps someone could explain it to me from scratch. Maybe useful to know that we are using Cisco equipment for call handling (VXML and TCL scripts). Thanks, Grant

3 2

Forward Register Message
by kellon＠sis-grenada.com 21 Nov '12

21 Nov '12

Hi All Can anyone point to documentation tutorial on howto forward statefully a REGISTER message to another SIP Server. I have looked at http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, However I'm getting errors with the uac_req commands even though I have loaded the uac.so module. .

2 1

Shared Call Appearances module
by Andrew Mortensen 21 Nov '12

21 Nov '12

I've been working on a Shared Call Appearances module for the past several months. It implements the Broadsoft SCA feature as laid out in Broadworks SIP Access Side Extensions Interface Specifications document. (Another implementation of the same feature was added to Freeswitch a few years ago, but we didn't want to use Freeswitch.) We've been testing and improving the module over the last month, and I think it's ready to share. The module is available at GitHub here: <https://github.com/fitterhappier/sca> To date, we've only tested the module with Polycom handsets. I'm happy to answer questions. I hope this will prove useful for others. Best, andrew

6 18

cost script
by Ruben Mejia 21 Nov '12

21 Nov '12

IPTEL We needhelp or cost for the development a script for kamailio which can do the following: Register kamailio to another gateway using UAC modules. route all traffic coming towards a private ip 10.x.x.x We will use a postgres database We have installed kamailio 3.1 with postgres support and UAC modules Rubén Mejía Telefonia Cabletamps

1 0

I can not make call from kamailio to asterisk by 3.3
by James zhu 21 Nov '12

21 Nov '12

hello, guys:i follow the linkhttp://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-a… can registered users, kamctl online shows those status. but i can not make calls. this is debug message when i make a call from 102 to SIP 103--------------------------------------------------------------- 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=486 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=3 n=return 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=489 a=6 n=route 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=636 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=601 a=25 n=has_totag 8(1286) DEBUG: siputils [checks.c:103]: no totag 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=501 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=26 n=is_method 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=17 n=if 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=496 a=25 n=t_check_trans 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [tm.c:1015]: lookup_original: t_lookupOriginalT returned: 0xb4d36320 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=497 a=25 n=t_relay 8(1286) DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=29 , global msg id=28 , T on entrance=(nil) 8(1286) DEBUG: tm [t_lookup.c:527]: t_lookup_request: start searching: hash=60763, isACK=0 8(1286) DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed 8(1286) DEBUG: tm [t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found 8(1286) DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb4d3a698, callback type 1, id 0 entered 8(1286) DEBUG: tm [t_lookup.c:762]: DEBUG: t_lookupOriginalT: searching on hash entry 60763 8(1286) DEBUG: tm [t_lookup.c:470]: DEBUG: RFC3261 transaction matched, tid=-d87543-e16321701f723866-1--d87543- 8(1286) DEBUG: tm [t_lookup.c:859]: DEBUG: t_lookupOriginalT: canceled transaction found (0xb4d36320)! 8(1286) DEBUG: tm [t_lookup.c:862]: DEBUG: t_lookupOriginalT completed 8(1286) DEBUG: tm [t_fwd.c:1243]: DEBUG: e2e_cancel: e2e cancel proceeding 8(1286) DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.1.103, 192.168.1.103, 0) 8(1286) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 8(1286) DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 8(1286) DEBUG: tm [t_reply.c:703]: DEBUG: reply sent out. buf=0xb721ee38: SIP/2.0 200 cancelin..., shmem=0xb4d38c08: SIP/2.0 200 cancelin 8(1286) DEBUG: tm [t_reply.c:713]: DEBUG: _reply_light: finished 8(1286) DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed 8(1286) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=3 n=exit 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 8(1286) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list=================================anyone can give a hint for that? thanks! Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com > From: sr-users-request(a)lists.sip-router.org > Subject: sr-users Digest, Vol 90, Issue 72 > To: sr-users(a)lists.sip-router.org > Date: Wed, 21 Nov 2012 08:58:40 +0100 > > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Shared Call Appearances module (Daniel-Constantin Mierla) > 2. Re: Shared Call Appearances module (Andrew Mortensen) > 3. Re: LCR weight (Juha Heinanen) > 4. problem with $time Pseudo-Variables on 3.3.2 (Uri Shacked) > 5. Re: Dialog module not decrementing profile size > (Grant Bagdasarian) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 20 Nov 2012 22:56:10 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Shared Call Appearances module > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <50ABFC7A.2070602(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > On 11/20/12 10:41 PM, Ovidiu Sas wrote: > > On Tue, Nov 20, 2012 at 4:22 PM, Andrew Mortensen > > <admorten(a)isc.upenn.edu> wrote: > >> On Nov 20, 2012, at 3:16 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > >> > >>> On 11/20/12 8:43 PM, Andrew Mortensen wrote: > >>>> On Nov 20, 2012, at 4:08 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > >>>> > >>>>> Hello, > >>>>> > >>>>> thank for contribution! > >>>>> > >>>>> On 11/20/12 12:16 AM, Andrew Mortensen wrote: > >>>>>> On Nov 19, 2012, at 5:53 PM, David | StyleFlare <david(a)styleflare.com> wrote: > >>>>>> > >>>>>>> Also quick read at the readme, it looks like it does not support multi-domain setups? > >>>>>> Not yet, no. I don't think it would take much work, though. > >>>>> I see it binds to usrloc, what information needs from there? We have two such modules right now, it would be good to make it work with both, kamailio's one has more features in regard to GRUU and partial outbound extensions. > >>>> It's registering for contact expiration and deletion events. The callback terminates subscriptions for the expired/deleted contact. This should probably be configurable. > >>> Ok, so it is about when a contact record in removed from location. > >>> > >>> Just to avoid misunderstandings, do you consider to make configurable this termination of subscriptions based on location? > >> Yes. I can imagine a setup where the location data wouldn't be saved on the host handling SCA. > >> > >>> Also, you need only the AoR and contact address in a callback for location record removal event (un-registration or registration expiration), right? > >> Correct, although Ovidiu makes a good point about (well-behaved) clients unsubscribing at the time of unregistration. In registering for usrloc callbacks, I'm trying to handle a couple problems. First, I'd like to avoid unnecessarily sending and retransmitting NOTIFYs to offline clients. More importantly, I want to be able to hear about expired registrations so I can clear appearance state on the rest of the subscribers. > > For clearing appearance state, you should rely on the dialog module > > and register a callback on the dialog module. When a dialog expires, > > you can clear up the stale appearances. > > Also, by enabling the sst module, you can have a tight control over > > the duration of the dialog. > > This will make the implementation cleaner and more robust. > > > I will not jump to such conclusions. > > While dialog module is designed to track active calls, does not mean > everything has to be on top of it. Just for example, dispatcher module > has an embedded lightweight call tracking system. > > No idea here what the broadsoft extensions require, afaik, there were > some custom specs. It is no problem to have a dedicated dialog tracking > system that is better designed for a particular functionality. We do > have other duplicated systems, for example least cost routing, nat > traversal, a.s.o. > > Also, optimizations and reuse of other parts can be done later if found > to be better ways. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > > > > ------------------------------ > > Message: 2 > Date: Tue, 20 Nov 2012 17:17:38 -0500 > From: Andrew Mortensen <admorten(a)isc.upenn.edu> > Subject: Re: [SR-Users] Shared Call Appearances module > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <65EAE745-8966-4848-870E-6A50246E0AB7(a)isc.upenn.edu> > Content-Type: text/plain; charset=iso-8859-1 > > > On Nov 20, 2012, at 3:43 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: > > > Hello Andrew, > > > > First of all, thank you for sharing your work. > > I was following this thread and I have a couple of questions. Why do > > you need to bind to the usrloc module? The subscription itself should > > be sufficient because if a phone will unregister, it will also > > unsubscribe, > > Yes, that's good point. However, if the phone goes off-line instead of unregistering, the contact's registration will expire, and no unsubscribe will take place. As I mentioned in my reply to Daniel, I'm most concerned about catching this so I can release any appearances seized by the expired/deleted contact, and notify other members of the group. I'm certainly open to alternatives. > > > which leads me to the following question: why not add a > > new dedicated module for call-info presence and reuse the existing > > infrastructure from kamailio for handling subscriptions/presence. In > > this case, your module should just push PUBLISH events to the presence > > server and the server will automatically send out notifications for > > subscribers that subscribed to call-info events. > > We've worked extensively with the existing presence & pua modules, albeit primarily dialog;sla using OpenSIPS behind the proxy. (Thanks, Anca!) > > SCA's unusual entanglement with call processing made me hesitate about building on top of the existing presence module. For a first pass, I also felt working with an entirely distinct module gave me more control and transparency during development of a very loosely documented event package, especially as I became more familiar with the available API. > > I don't have any objection to revisiting design decisions, of course. I'm sure the module will continue to evolve, and it would be nice to eliminate redundancy if possible. > > > Based on your README file, you inspect SIP requests/replies and based > > on the presence of the call-info header, you create call-info events. > > This is great for sharing the appearances between phones, but how do > > you perform the retrieval of a call that was put on hold? Are you > > using a dedicated B2BUA behind kamailio? > > No B2BUAs are involved. > > The INVITE retrieving a call held by another member of an SCA group has a particular set of characteristics: RURI, To and From URIs are all the SCA AoR; new unconfirmed dialog (no to-tag); and a Call-Info header referring to the index of the held call. The module detects this type of INVITE, looks up the dialog associated with the information in the Call-Info header, and injects a Replaces header with the dialog of the held call before relaying it to the remote party. The remote party must support RFC3891. I've only worked with Polycoms and Cisco gateways to this point, and both do support that RFC. > > I'm very interested in hearing from users with Cisco (and Snom and Aastra?) SCA setups. I have no doubt they'll find bugs resulting from assumptions I've made in the code because I've only tested with Polycom, not least the dependency on RFC3891 support. > > Thanks for your feedback! > > Best, > andrew > > > > > On Tue, Nov 20, 2012 at 3:16 PM, Daniel-Constantin Mierla > > <miconda(a)gmail.com> wrote: > >> > >> On 11/20/12 8:43 PM, Andrew Mortensen wrote: > >>> > >>> On Nov 20, 2012, at 4:08 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> > >>> wrote: > >>> > >>>> Hello, > >>>> > >>>> thank for contribution! > >>>> > >>>> On 11/20/12 12:16 AM, Andrew Mortensen wrote: > >>>>> > >>>>> On Nov 19, 2012, at 5:53 PM, David | StyleFlare <david(a)styleflare.com> > >>>>> wrote: > >>>>> > >>>>>> Also quick read at the readme, it looks like it does not support > >>>>>> multi-domain setups? > >>>>> > >>>>> Not yet, no. I don't think it would take much work, though. > >>>> > >>>> I see it binds to usrloc, what information needs from there? We have two > >>>> such modules right now, it would be good to make it work with both, > >>>> kamailio's one has more features in regard to GRUU and partial outbound > >>>> extensions. > >>> > >>> It's registering for contact expiration and deletion events. The callback > >>> terminates subscriptions for the expired/deleted contact. This should > >>> probably be configurable. > >> > >> > >> Ok, so it is about when a contact record in removed from location. > >> > >> Just to avoid misunderstandings, do you consider to make configurable this > >> termination of subscriptions based on location? > >> > >> Also, you need only the AoR and contact address in a callback for location > >> record removal event (un-registration or registration expiration), right? > >> > >> > >>> > >>>> Would you consider adding it to main git repository and maintaining it > >>>> there? > >>> > >>> Yes, that would be great. > >> > >> > >> We will prepare write access for you. > >> > >> > >>> Would you prefer to have it in modules? I started work on it in > >>> modules_s, but there's no particular reason it has to stay there. > >> > >> To make it work with both variants of usrloc modules, I think it requires > >> some split of the code. IIRC, both modules still use same names for > >> structures, so including files from both of them will fail. > >> > >> One solutions I am thinking of is to add two small modules, one that binds > >> to usloc(k) and the other to usrloc(s). Each of these two modules will > >> export a function that allow registering callbacks for execution when a > >> location record expires, giving the aor, contact and event type. The > >> existing module will use the intermediate module instead of directly binding > >> to usrloc structures. > >> > >> Usrloc module from kamailio has more enhancements in regards to standard > >> extensions for location services, still it lacks the feature of being able > >> to store variables per contact record. It is the reason that kept back the > >> removal (upon merging) of usrloc module from ser. The solution with > >> intermediary modules should work until we have a merged usrloc module. > >> > >> What do you think about this proposal? > >> > >> Cheers, > >> Daniel > >> > >> > >> -- > >> Daniel-Constantin Mierla - http://www.asipto.com > >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > >> > > > > > ------------------------------ > > Message: 3 > Date: Wed, 21 Nov 2012 01:04:52 +0200 > From: Juha Heinanen <jh(a)tutpro.com> > Subject: Re: [SR-Users] LCR weight > To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <20652.3220.931000.738348(a)sip.test.fi> > Content-Type: text/plain; charset=us-ascii > > Camila Troncoso writes: > > > scripts are provided in lcr/utils directory that can be used to check > > the probabilities resulting from a given set of weight values. Same can > > be done with command 'kamctl eval_weights'. > > > > I really don't understand what is the probability finally assign to each > > gateway when they have the same priority. > > All I can see is that it doesn't correspond to percentage of capacity from > > the example above , but how can I determine what value gives the percent > > that I want? > > > > Please give an example like: > > > > Rule_id Gw_id Priority weight > > 36 68 1 60 > > 36 69 1 14 > > 36 70 1 13 > > 36 71 1 13 > > you have to figure that out yourself using the script referenced in > readme. > > -- juha > > > > ------------------------------ > > Message: 4 > Date: Wed, 21 Nov 2012 09:55:17 +0200 > From: Uri Shacked <ushacked(a)gmail.com> > Subject: [SR-Users] problem with $time Pseudo-Variables on 3.3.2 > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAMMbDhQn1r_ke72-u2is8MyRXUV6KkLnZqYwxL5eZG44yYG_2Q(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > I upgraded to 3.3.2 > > On 3.2 i used the variable $time() to get the current time for some process. > > Every 2 minutes a RTIMER process started, checked the $time(hour) and > $time(min) and on a certain time executed something. > > This process stopped working on 3.3.2 > > When i debug, i see that the $time() variable keeps on giving me the same > hour and minute that was given on the first run. > > Meaning the $time() variable does not give the current time. > > I notice also that the $Ts that on 3.2 gave a current time stopped doing it > and $TS does it now.... > > Any help here? > > Is it OK and i need to check time another way? Or something in the $time() > has changed...? > > Thanks, > > Uri >

1 0

dialog ka_timer and ka_interval problem
by Kristofer Signer 21 Nov '12

21 Nov '12

Hello all, first of all I would like to thank you all your help. I've tried to implement dialog keep-alive messages but cant make kamailio send more then just one SIP OPTION request to each end. The OPTION request is sent after ka_interval time but no more OPTION requests are sent during the call. How should this be implemented? and was is the difference and relation between ka_timer and ka_interval. tanks ./Kristofer

2 2

LCR weight
by Camila Troncoso 20 Nov '12

20 Nov '12

Hi, I have a question regarding LCR module and their weight variable. If I have 4 gateways for the same prefix, and with the same priority: Rule_id Gw_id Priority weight 36 68 1 60 36 69 1 14 36 70 1 13 36 71 1 13 The weight variable acts as a percentage of utility of that gateway? If for example I have 100 calls, the gateway with gw_id 68 will receive 60 calls , and the other 14, 13 and 13? Regards, Camila *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 3

Getting started with Kamailio
by Joel Smith | VOZELIA 20 Nov '12

20 Nov '12

Hi, First of all say that I'm new to Kamailio. I've been working with asterisk for a while now and it's time to get started with the sip router project. My idea for a start is to reduce the load of our asterisk servers and once we have good knowledge on how kamailio works, migrate some of our services. I've setup a server in our lab with kamailio 3.3.0 and siremis 3.3.0, created a user (subscriber) and registered with a softphone. So far OK. Now I'm trying to make a call and route it through one of our asterisk servers for testing. To do this I need LCR module (thanks Sébastien, I wasn't subscribed to this list yet) and configure it. After reading http://kamailio.org/docs/modules/stable/modules/lcr.html#id2553110 I've added in kamailio.cfg: #!define WITH_LCR #!ifdef WITH_LCR loadmodule "lcr.so" #!endif #!ifdef WITH_LCR # ----- lcr params ----- modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "tag_avp", "$avp(lcr_tag)") #!endif And now I see LCR module being loaded at startup. But, I don't understand what gw_uri_avp, ruri_user_avp or tag_avp mean. Where can I find some doc/manual to understand what AVP means and does? (For example the $avp(i:709), etc.) I've added a LCR GW, LCR TARGET and LCR RULE with the following data (I have no idea if it's completely wrong): mysql> select * from lcr_gw; +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | 1 | 0 | TEST | XX.XXX.XXX.X | test12345678.testing.com | 5060 | NULL | 1 | 1 | NULL | NULL | NULL | 0 | NULL | +----+--------+---------+--------------+--------------------------+------+--------+------------+-----------+-------+--------+------+-------+---------+ 1 row in set (0.00 sec) mysql> select * from lcr_rule; +----+--------+--------+----------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+--------+----------+-------------+---------+---------+ | 1 | 1 | .* | | | 0 | 1 | +----+--------+--------+----------+-------------+---------+---------+ 1 row in set (0.00 sec) mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 1 | 1 | 1 | 1 | 1 | +----+--------+---------+-------+----------+--------+ 1 row in set (0.00 sec) mysql> Am I still missing basic stuff to be able to place a call ? (To get started I'd like to route all calls to our Asterisk server) I'm glad to read documentation so If all of this is already somewhere please guide me cause I haven't found too much... Thanks in advanced. Regards, Joel.

4 4

t_branch_timeout()
by Alex Balashov 20 Nov '12

20 Nov '12

I am using the latest pull of branch 3.3, and t_branch_timeout() doesn't work if the previous branch resulted in a reply (i.e. a negative final reply). That is to say, t_branch_timeout() == FALSE on a genuine timeout, as long as the preceding branch provided a result (such as 404). t_branch_timeout() == TRUE if this is a timeout, on the first branch. In other words, it appears that if t_any_replied() == TRUE, then t_branch_timeout() == FALSE, even for this branch, and even if the branch really did time out (no response at all from destination). That's not how the documentation says it should behave. Am I doing something wrong? If so, how do I find out if _this particular_ branch timed out, from the failure_route[]? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 3

temporary SIP user
by Bolang 20 Nov '12

20 Nov '12

Hi all, is there any standard mechanism to create a temporary user? i know i can create it and then delete it after some amount of time. but, i'm looking for an established/standard way of doing this. Ideally, the user will be automatically deleted after unregistered. thanks

2 2

Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 20 Nov '12

20 Nov '12

I'd be curious to know if your performance might actually be higher, owing to improved CPUs since the study was done, hyperthreading, more frames per NIC interrupt, etc. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:Thank you, setting ulimits worked! And the performance is the same as stated in the document I mentioned! :) One more thing, I found these errors in syslog: Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:211]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5270]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location This is possibly something related to my sipp scenario, this is the INVITE sent from sipp <![CDATA[ INVITE sip:800@perftest.vm SIP/2.0 Via: SIP/2.0/[transport] 10.0.2.36:[local_port];branch=[branch] From: "700" <sip:500@10.0.2.36>;tag=[call_number] To: <sip:800@perftest.vm> Call-ID: [call_id] CSeq: 1 INVITE Contact: "700" <sip:700@10.0.2.36:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] 10.0.2.36 s=- c=IN IP[local_ip_type] 10.0.2.36 t=0 0 m=audio [auto_media_port] RTP/AVP 8 a=rtpmap:8 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 ]]> Do you see something that could cause this error ? Otherwise the call is initiated ok, but I really dont understand what is so strange to kamailio in this INVITE. On Thu, Sep 13, 2012 at 6:37 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >> You mean on the proxy side? I'm running rtpproxy as root, limits are >> still applied ? ulimit -s unlimited should do the trick ? > > Yes, they usually applied even for superuser, and yes - this should > help (if that's the issue). > > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

6 14

Clarification on is_known_dlg()
by David 20 Nov '12

20 Nov '12

Hello, I am running Kamailio 3.2.3. I am storing dialogs in the database table dialog. loadmodule "dialog.so" modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 1) later I have : if ( is_known_dlg() ) { } I would like a clarification on how the matching is done. Given that I have realtime table setup, does that mean that dialogs that are in the table will return true when I call is_known_dlg() ? Does that mean that if the dialog is NOT in the table, is_known_dlg will return false ? Thanks for the clarification ! David

2 1

UAC and realm
by Johan Wilfer 20 Nov '12

20 Nov '12

Hi, I've done some tests with the UAC module to authenticate to a remote proxy. I've based my config on this example: http://docs.huihoo.com/opensips/tutorials/uac/ar01s06.html (example 9) I have found that if I send a call from a asterisk via kamailio to my remote proxy the realm the remote proxy uses is the ip-address of the asterisk-server (aaa.bbb.ccc.ddd). Like this: WWW-Authenticate: Digest nonce="1353399363:8d2317487ee1521328d0e2237e444e2d",algorithm=MD5,realm="aaa.bbb.ccc.ddd",qop="auth",stale=false. If I have a param like this, the auth works: modparam("uac","credential","username:aaa.bbb.ccc.ddd:password") But as I would like to use more than one remote proxy (=more than one provider) per asterisk server this needs some more work. Is there a way to get UAC to ignore the realm from the remote proxy and send the auth anyway? Alternate paths I've examined: 1. Rewrite from-domain to get the provider to use another realm: modparam("uac","credential","username:example.com:password") uac_replace_from("sip:$fU@example.com"); Works, but I don't think this is a very good solution. 2. Use auth_username_avp / auth_password_avp / auth_realm_avp to feed the uac_auth()-function the right credentials. modparam("uac","auth_username_avp","$avp(s:uac_user)") modparam("uac","auth_password_avp","$avp(s:uac_pass)") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") in the failure route: $avp(s:uac_user) = "username"; $avp(s:uac_pass) = "password"; $avp(s:uac_realm) = "aaa.bbb.ccc.ddd"; This works as well, but I would like to either get the realm the remote proxy sends and use it or make uac ignore the realm. The realm is unimportant as we already know what proxy we are talking to. I've tried $hdr(WWW-Authenticate) but it evaluates to null. Any ideas how to do this? Thanks! -- Johan Wilfer

2 2

Re: [SR-Users] Returning problem
by Moacir Ferreira 19 Nov '12

19 Nov '12

I wish it were that simple... I have done some "reading" and found that Ubuntu has changed to upstart http://upstart.ubuntu.com/cookbook/. However, I could not find how I can start a non "upstart" style init script like Kamailio has before the upstart styled ones... I guess that I have to either to get the Kamailio community to create an upstart script or find some help from the Ubuntu community how to overcome the problem. Anyway, I think the Kamailio team should address it as it will become a problem for everyone using Kamailio on Ubuntu. Any help? Moacir From: 4lists(a)gmail.com To: sr-users(a)lists.sip-router.org Date: Mon, 12 Nov 2012 14:42:31 -0200 Subject: Re: [SR-Users] Returning problem This is more a system initialization problem, so, take a look in your /etc/rc<run-level>.d directory (or the correspondent one), specially in the S* files and find out the start sequence. There You'll find the answer. Edson. Em 09/11/2012 17:03, Moacir Ferreira escreveu: > My kamailio start script has the following BEGIN INIT INFO: > > # > ### BEGIN INIT INFO > # Provides: kamailio > # Required-Start: $syslog $network $local_fs $time mysql > # Required-Stop: $syslog $network $local_fs mysql > # Default-Start: 2 3 4 5 > # Default-Stop: 0 1 6 > # Should-Start: postgresql mysql radius > # Should-Stop: postgresql mysql radius > # Short-Description: Start the Kamailio SIP proxy server > # Description: Start the Kamailio SIP proxy server > ### END INIT INFO > > However, when I boot my server Kamailio will start before mysql. So, > after boot I can start Kamalio but at boot time it will fail. > > Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. > All the code has been compiled at the server. > > Cheers! > > Moacir > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

2 5

how to use uac_auth in module UAC
by Freddy C 19 Nov '12

19 Nov '12

Hello, I have a question about UAC and function module "uac_auth" for a REGISTER request. My idea is to use this when I get a 401 unauthorized response. This answer is captured from failure_route. Then generate the header Authorized and forward the request REGISTER to Asterisk. I could help with some examples?. Freddy Condori Ch Ingeniero Investigador Telefonía IP

1 0

Sync nonce between various servers
by Andreas Granig 19 Nov '12

19 Nov '12

Hi, There are lots of parameters controlling the creation of nonce values on a server, and I'm curious if there is a way to kind of "sync" them between servers. The use case would be to have a UA send for example its registration to Proxy1. Proxy1 would challenge it, UA will send the registration again, this time with credentials. Proxy1 would look up the user based on $au/$ar in the subscriber table, and if it's not found, will look up the responsible proxy from another table (with key being $au@$ar), forward it to Proxy2, which then would be able authenticate the user. The reason for this is that the auth credentials are unique across all servers and reliably identify a user, whereas for example From could be something else (e.g. in case of an IP-PBX sending a CLI in the From-userpart). Challenging the user on the second proxy again would theoretically be possible, but if the UA gets a 401 twice (once from Proxy1, once from Proxy2), it'll most likely pop up a password form for soft-clients, so I want to avoid that. Any ideas how to accomplish that? Andreas

4 9

Problem routing to voicemail
by Christophe ROY 19 Nov '12

19 Nov '12

Hi everyone I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT: - I had to adapt it because I use LDAP authentication with Kamailio - I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM - we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0" For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone(a)domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile: Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It seems to happen on the if (!t_relay()) line in ROUTE[RELAY] 192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk. What could be the problem? Thanks for your help Christophe Below is my kamailio.cfg: #!WITH_DEBUG #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # ###!define WITH_NAT # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ###!define WITH_ASTERISK ###!define WITH_VOICEMAIL #!define WITH_LDAP #!define WITH_AUTH #!define WITH_MYSQL ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openserrw:openserrw@192.168.14.28/openser" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://asterisk:asteriskpwd@192.168.14.28/asterisk" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL6 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="mydomain.corp" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=tcp:192.168.14.24:5060 #listen=tcp:192.168.14.25:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "192.168.14.32" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.14.32" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "192.168.14.24" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" # loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #chris loadmodule "ldap.so" modparam ("ldap", "config_file", "/etc/kamailio/ldap.cfg") #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_tmp") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # suppress the check for the CSEQ method # modparam("sanity", "default_checks", 967) # ----- 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) #edit asterisk #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 256) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif #chris commented out this part # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.14.25:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@teopad-toip.corp") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); xlog("L_INFO","apres REQINIT"); # NAT detection route(NATDETECT); xlog("L_INFO","apres NATDETECT"); # handle requests within SIP dialogs route(WITHINDLG); xlog("L_INFO","apres WITHINDLG"); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); xlog("L_INFO","apres t_check_trans"); # authentication route(AUTH); xlog("L_INFO","apres AUTH"); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); xlog("L_INFO","apres RECORD ROUTE"); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } xlog("L_INFO","apres INVITE"); # dispatch requests to foreign domains route(SIPOUT); xlog("L_INFO","apres SIPOUT"); ### requests for my local domains # handle presence related requests route(PRESENCE); xlog("L_INFO","apres PRESENCE"); # handle registrations route(REGISTRAR); xlog("L_INFO","apres REGISTRAR"); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); xlog("L_INFO","apres PSTN"); # user location service route(LOCATION); xlog("L_INFO","apres LOCATION"); route(RELAY); xlog("L_INFO","apres RELAY"); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. xlog("L_INFO","Dans route relay"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","avant manage branch"); t_on_branch("MANAGE_BRANCH"); xlog("L_INFO","avant manage reply"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { xlog("L_INFO","avant manage failure"); t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { xlog("L_INFO","reply error"); sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # if(!sanity_check("1511", "7")) # { # xlog("Malformed SIP message from $si:$sp\n"); # exit; # } } # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Dans WITHINDLG"); if (has_totag()) { xlog("L_INFO","dans has totag"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO","looseroute"); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_INFO","ack"); # ACK is forwarded statelessy route(NATMANAGE); } xlog("L_INFO","relay"); route(RELAY); } else { xlog("L_INFO","else"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests xlog("L_INFO","subscribe avant presence"); route(PRESENCE); xlog ("L_INFO","apres presence"); exit; } if ( is_method("ACK") ) { xlog("L_INFO","else ack"); if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server xlog("L_INFO","else ack avant relay"); t_relay(); xlog("L_INFO","else ack apres relay"); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO","else final"); exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #edit asterisk #!ifdef WITH_ASTERISK xlog ("L_INFO","avant regfwd dans registrar"); route(REGFWD); xlog ("L_INFO","apres regfwd dans registrar"); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #edit asterisk #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { #if new call from out there - send to Asterisk # - non-INVITE requests are routed directly by Kamailio # - traffic from Asterisk is router also directly by Kamailio xlog ("L_INFO", "avant toasterisk dans location"); route(TOASTERISK); xlog ("L_INFO", "apres toasterisk dans location"); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #edit asterisk #!ifdef WITH_ASTERISK #do not auth traffic from Asterisk: trusted! xlog ("L_INFO", "avant if route fromasterisk"); if(route(FROMASTERISK)) return; #!endif if (is_method("REGISTER")) # { # # authenticate the REGISTER requests (uncomment to enable auth) # if (!www_authorize("$td", "subscriber")) # { # www_challenge("$td", "0"); # exit; # } # # if ($au!=$tU) # { # sl_send_reply("403","Forbidden auth ID"); # exit; # } { #edit asterisk ##!ifdef WITH_ASTERISK # xlog ("L_INFO", "dans auth / authcheck sipusers"); # if (!auth_check("$fd","sipusers","1")) ##!else if(is_present_hf("Authorization")) ##!endif { # ldap search if (!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?teopad-Sip-Username,teopadSipPassword?one?(teopad-Sip-Username=$fU)")) # if (!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?sAMAccountName,?one?(sAMAccountName=$fU)")) { switch ($retcode) { case -1: # no LDAP entry found sl_send_reply("404", "User Not Found"); xlog("L_INFO", "ldap_search: NO found [$retcode] entries for (uid=$fU)"); exit; case -2: # internal error sl_send_reply("500", "Internal server error"); exit; default: exit; } } ldap_result("teopad-Sip-Username/$avp(s:username)"); ldap_result("teopadSipPassword/$avp(s:password)"); xlog("L_INFO", "ldap_search: found [$retcode] entries for (uid=$fU)"); if(!pv_www_authenticate("$td", "$avp(s:password)", "0")) { xlog ("L_INFO", "ldap pv_authenticate failed") ; www_challenge("$td","1"); exit; } save("location"); sl_send_reply("200", "ok"); xlog ("L_INFO", "ldap pv_authenticate ok") ; exit; } else { www_challenge("$td","1"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # # authenticate if from local subscriber if (from_uri==myself) { # if (!proxy_authorize("$fd", "subscriber")) { # proxy_challenge("$fd", "0"); # exit; # } if (is_method("PUBLISH")) { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; xlog ("L_INFO", "tU = $tU") ; xlog ("L_INFO", "fd = $fd") ; xlog ("L_INFO", "rd = $rd") ; if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID au!=fu ou au!=tu"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; # if ($au!=$fU) { # sl_send_reply("403","Forbidden auth ID au!=fu"); # exit; # } } #consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); #rtpproxy_manage("co","82.127.95.167"); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "tcp:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); xlog("L_INFO","tovoicemail ru: $ru"); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_INFO","dans managereply"); xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]"){ xlog("L_INFO","avant route natmanage"); route(NATMANAGE); } } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #edit asterisk #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { xlog ("L_INFO", "Dans FROMASTERISK? $si / $sp"); if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); xlog ("L_INFO", "Dans TOASTERISK $du") ; xlog ("L_INFO", "Juste avant route relay"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { xlog("L_INFO", "Dans REGFWD"); if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; xlog("L_INFO","avant if dans regfwd"); if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif

2 3

Problem with include_file quotes
by Pirjo Ahvenainen 19 Nov '12

19 Nov '12

Hi All, I've been away from Kamailio for a few months, but recently I started working with it again. I could use a hand in a small problem: I'm using include_file statement to give some structure to my config file, but when checking the config I keep getting error message: 0(4290) : <core> [cfg.lex:1554]: included file name must be between quotes: "defined_modules.cfg" I'm using Kamailio 3.3.2 (version: kamailio 3.3.2 (x86_64/linux) 78d4b1) on Ubuntu. And here's the beginnign of the config file: #!KAMAILIO # include_file "defined_modules.cfg" include_file "defined_values.cfg" include_file "global_parameters.cfg" Those look like quotes to me, but could the problem be something like encoding of the config file itself, something in the included files, or is there something else I can check? I don't have extra whitespaces etc, perhaps a typo I just don't see? Cheers, Pirjo

2 3

How to get dialog variable externally?
by Mino Haluz 19 Nov '12

19 Nov '12

Hi, I'm using dialog module and I set some dlg_var for every dialog. Is it possible to get this variable value with MI command somehow ? I know to get list of dialogs using dlg_list only. Thanks! Mino

2 1

db_sqlite bugs
by Pedro Antonio Vico Solano 19 Nov '12

19 Nov '12

5 11

Planning for memory
by Moacir Ferreira 19 Nov '12

19 Nov '12

Hi, I am trying to figure out how much memory a Kamailio server should have to support an X number of users, considering that I am running Kamailio, MySQL and RTPproxy on the same machine. I am planning to use Kamailio at medium/large enterprises, not at service providers but I got the following questions: - Will Kamailio dynamically allocate the system memory it needs or I have to tune up the memory it takes? I.e.: If I use 8G memory, will it take what it needs or I need to adjust it somewhere? - How much memory is reasonable for 10K, 50K and 100K users in such "enterprise" scenarios? - Can you reference any documentation from where I can get this information? Thanks, Moacir

2 1

Kamailio cfg file doubt
by Kamal Palei 17 Nov '12

17 Nov '12

Dear All When I get INVITE (received in UDP), I send it to another proxy in TLS. For this purpose I have modified kamailio.cfg route section , please refer the lines in red. Just before sending in TLS, I modify the IP address in SDP, so that media packets will pass through RTP proxy running When INVITE is reached the next proxy, I see the IP address is changed . So far so good. Next I am trying to modify the IP address in 200 OK to INVITE. However it is not taking effect. I hope we need to modify in onreply_route function. Something is wrong here. I am receiving 200 OK for INVITE in TLS. Next I am sending 200 OK for INVITE to endpoints using UDP. Not sure if onreply_route is being executed or not. Can somebody give me pointer whats wrong here. *# Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy("r"); } if (isbflagset(FLB_NATB)) { fix_nated_contact(); } #!endif } * * # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); setflag(FLT_NATS); # NAT detection route(NAT); if(is_method("PUBLISH")) { loose_route(); t_relay(); exit; } if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:443"); exit(); } if(is_method("INVITE|BYE|CANCEL|SUBSCRIBE|REFER|NOTIFY")) { xdbg("incoming request\n"); route(RTPPROXY); t_relay_to("tls:115.114.48.75:443"); exit(); } *Thannks Kamal

2 2

segfault when phone registers on TLS
by Denis 15 Nov '12

15 Nov '12

Hi, I am using kamailio-3.3.2 on Ubuntu 11.04 x64 and when trying to connect via TLS from my Yealink phone kamailio crashes with the following backtrace: Loaded symbols for /lib/x86_64-linux-gnu/libnss_files.so.2 Core was generated by `/opt/kamailio/sbin/kamailio'. Program terminated with signal 11, Segmentation fault. #0 handle_ser_child (p=0x7fcbaea2eba0, fd_i=-1) at tcp_main.c:3575 3575 if (unlikely(p->unix_sock<=0)){ (gdb) bt #0 handle_ser_child (p=0x7fcbaea2eba0, fd_i=-1) at tcp_main.c:3575 #1 0x000000000051e5bc in send2child (tcpconn=0x7f8c001597d8, ev=<value optimized out>, fd_i=-1) at tcp_main.c:3975 #2 handle_tcpconn_ev (tcpconn=0x7f8c001597d8, ev=<value optimized out>, fd_i=-1) at tcp_main.c:4310 #3 0x0000000000527dbc in io_wait_loop_epoll () at io_wait.h:1092 #4 tcp_main_loop () at tcp_main.c:4656 #5 0x00000000004726f5 in main_loop () at main.c:1727 #6 0x000000000047402e in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2546 (gdb) The manual I am following is http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates Thanks, Den

3 10

Kamailio as SMTP server
by Krishna Kurapati 15 Nov '12

15 Nov '12

Hi, Can Kamailio act as SMTP server to receive emails and forward those to SIP users as MESSAGEs? It would be nice if postfix or some other rich SMTP server could function as module of kamailio using all the services provided by kamailio scripting. Krish Kura

2 1

Problem routing to voicemail
by Christophe ROY 15 Nov '12

15 Nov '12

Hi everyone I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT: - I had to adapt it because I use LDAP authentication with Kamailio - I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM - we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0" For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone(a)domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile: Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It seems to happen on the if (!t_relay()) line in ROUTE[RELAY] 192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk. My kamailio.cfg is available here as it's too big for the list: http://pastebin.com/zhncJ6j1 What could be the problem? Thank you very much for your help Christophe

1 0

Replicate dialog state
by Kristofer Signer 15 Nov '12

15 Nov '12

In my current setup I've register messages replicated to a second server. Is it possible to also replicate other messages so I can hold dialog state on both servers? I've tried to read the documentation and also google but can't find any examples. Thanks ./Kristofer

2 1

RPC, XMLRPC or other
by Kristofer Signer 14 Nov '12

14 Nov '12

Hello, I'm trying to dig in to the kamailio XMLRPC interfaces and the limitations I read in http://www.kamailio.org/docs/modules/stable/modules/xmlrpc.html#xmlrpc.impl… is really a no go for us. For example, when to retrieve user locations and list of dialogs we get a xml document which is not so well formatted and we can not successfully parse it in an xmlrpc-parser. Is there other options for xmlrpc module? thanks ./Kristofer

4 11

check if the sent reply was local
by Uri Shacked 14 Nov '12

14 Nov '12

Hi, what is the command to check if the reply was recieved or localy generated? Uri

3 3

Re: [SR-Users] sr-users Digest, Vol 90, Issue 39
by Yufei Tao 14 Nov '12

14 Nov '12

What I did to overcome this problem was writing a bash script, that checks if mysql is running, if so, start Kamailio, if not, start mysql first, then start Kamailio, and retries if things fail. And in /etc/rc.local, call this script. Seems to work fine on Ubuntu 10.04. Not tried 12.04 yet. Yufei > > > ------------------------------ > > Message: 2 > Date: Tue, 13 Nov 2012 18:15:05 -0500 > From: Moacir Ferreira <moacirferreira(a)hotmail.com> > Subject: Re: [SR-Users] Returning problem > To: Kamailio Mail List <sr-users(a)lists.sip-router.org> > Message-ID: <COL125-W3986EA29065F0966C8AE6C86C0(a)phx.gbl> > Content-Type: text/plain; charset="iso-8859-1" > > > I wish it were that simple... I have done some "reading" and found that Ubuntu has changed to upstart http://upstart.ubuntu.com/cookbook/. However, I could not find how I can start a non "upstart" style init script like Kamailio has before the upstart styled ones... I guess that I have to either to get the Kamailio community to create an upstart script or find some help from the Ubuntu community how to overcome the problem. Anyway, I think the Kamailio team should address it as it will become a problem for everyone using Kamailio on Ubuntu. Any help? Moacir From: 4lists(a)gmail.com > To: sr-users(a)lists.sip-router.org > Date: Mon, 12 Nov 2012 14:42:31 -0200 > Subject: Re: [SR-Users] Returning problem > > This is more a system initialization problem, so, take a look in your > /etc/rc<run-level>.d directory (or the correspondent one), specially in > the S* files and find out the start sequence. There You'll find the answer. > > Edson. > > Em 09/11/2012 17:03, Moacir Ferreira escreveu: >> My kamailio start script has the following BEGIN INIT INFO: >> >> # >> ### BEGIN INIT INFO >> # Provides: kamailio >> # Required-Start: $syslog $network $local_fs $time mysql >> # Required-Stop: $syslog $network $local_fs mysql >> # Default-Start: 2 3 4 5 >> # Default-Stop: 0 1 6 >> # Should-Start: postgresql mysql radius >> # Should-Stop: postgresql mysql radius >> # Short-Description: Start the Kamailio SIP proxy server >> # Description: Start the Kamailio SIP proxy server >> ### END INIT INFO >> >> However, when I boot my server Kamailio will start before mysql. So, >> after boot I can start Kamalio but at boot time it will fail. >> >> Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. >> All the code has been compiled at the server. >> >> Cheers! >> >> Moacir >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > > > >

1 0

update the to_tag on a local generated reply
by Uri Shacked 14 Nov '12

14 Nov '12

Hi, When i set a timer to "request time out" kamilio generates a 408 with now to_tag. I saved the to_tag from the 183 and try to update the acc table with the to_tag (i need that for my CDR to work) My question is (understanding there is no option to set the to_tag on the 408) is: How to i get the right timing to update using sqlops? i must update after the acc inserted the row to the acc table.... Any ideas? Thanks, Uri

2 3

Presence Dialog Info synchronisation issue
by David 12 Nov '12

12 Nov '12

Hello, I have two Kamailio boxes running. First one routes calls between my phones and asterisk. Here are the extra lines compared to the default config that came with my installation on the first server. Let's call this server kamailio : loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_dialoginfo.so" modparam("pua_dialoginfo", "include_localremote", 0) modparam("pua_dialoginfo", "include_tags", 0) modparam("pua_dialoginfo", "include_callid", 0) modparam("pua_dialoginfo", "caller_confirmed", 1) modparam("pua", "outbound_proxy", "sip:kamailio_presence_ip:5060") if ( is_method("SUBSCRIBE") ) { t_relay("kamailio-presence-ip", "5060"); } The second one handles presence stuff ( SUBSCRIBE and PUBLISH ), let's call it presence : After the standard sanity checks : if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; exit; The problem is that I have two phones connected to the same username on the server. Here is the sequence of events of a correctly functionning call : 1. Kamailio receives an INVITE 2. It sends the INVITE to each of the two phones 3. The first phone replies 180 Ringing 4. Kamailio sends a PUBLISH to presence 5. Presence replies with an OK, which includes a SIP-ETag header 6. Kamailio recieves the OK 7. Kamailio receives the 180 Ringing from the second phone 8. Kamailio sends a new PUBLISH with the SIP-If-Match using the value from the SIP-ETag header 9. Presence receives it, updates the presence table and sends an OK. This is ok, the trouble is when the second 180 Ringing reaches Kamailio before the OK to the publish with the SIP-ETag. 1. Kamailio receives an INVITE 2. It sends the INVITE to each of the two phones 3. The first phone replies 180 Ringing 4. Kamailio sends a PUBLISH to presence 5. Kamailio receives the 180 Ringing from the second phone 6. Kamailio sends a PUBLISH to presence 7. Presence replies with an OK, which includes a SIP-ETag header 8. Presence replies with an OK, which includes a second SIP-ETag header 9. Kamailio recieves the OK 10. Kamailio recieves the OK The presence table now has two seperate dialogs for the call because Kamailio didn't send the SIP-If-Match header as it did not yet receive the reply from presence. When the two phones start ringing at the same time, the time between the two packets arriving at the server is shorter than a round trip to the presence server. In the second scenario, Kamailio never updates the presence information using the first SIP-ETag, so the light on my phone stays stuck at "ringing". Where do I go from here ? Thanks, David

1 0

Returning problem
by Moacir Ferreira 12 Nov '12

12 Nov '12

My kamailio start script has the following BEGIN INIT INFO: # ### BEGIN INIT INFO # Provides: kamailio # Required-Start: $syslog $network $local_fs $time mysql # Required-Stop: $syslog $network $local_fs mysql # Default-Start: 2 3 4 5 # Default-Stop: 0 1 6 # Should-Start: postgresql mysql radius # Should-Stop: postgresql mysql radius # Short-Description: Start the Kamailio SIP proxy server # Description: Start the Kamailio SIP proxy server ### END INIT INFO However, when I boot my server Kamailio will start before mysql. So, after boot I can start Kamalio but at boot time it will fail. Any hints how to fix it? I am running Ubuntu server 12.04 LTS - 64 bits. All the code has been compiled at the server. Cheers! Moacir

2 1

LUA 5.2 - Support
by David | StyleFlare 12 Nov '12

12 Nov '12

Curious to know if there is any plans to support LUA 5.2. Seems like this is a newer release, I tried to build kamailio lua modules using 5.2 but it was not supported. Seems like some minor changes in the way LUA handles some things. Any ideas? Thanks.

2 1

xcap: no more space for spec value
by Juha Heinanen 12 Nov '12

12 Nov '12

i got the these error messages when my xcap server tried to serve 'GET xcap-caps' request: Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: INFO: XCAP service GET <xcap-caps> Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [pvapi.c:1305]: no more space for spec value Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [pvapi.c:1314]: buffer overflow -- increase the buffer size... Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: <core> [sr_module.c:1665]: Could not convert the PV-formated string to str Nov 12 02:01:31 sip /usr/sbin/pres-serv[30613]: ERROR: xhttp [xhttp_mod.c:461]: unable to get body config code where this comes is this: xlog("L_INFO", "XCAP service $rm <$xcapuri(u=>auid)>\n"); $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.pres-content</auid> <auid>org.openmobilealliance.search</auid> <auid>org.openmobilealliance.xcap-directory</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:prs:pres-content</namespace> <namespace>urn:oma:xml:xdm:search</namespace> <namespace>urn:oma:xml:xdm:xcap-directory</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "OK", "application/xcap-caps+xml", "$var(xbody)"); i have not checked yet, which buffer should be increased, but would it be a good idea to make the default bigger? -- juha

2 2

Use different flags in rtpproxy_manage() based on from which route its called
by Aft nix 12 Nov '12

12 Nov '12

Hi, rtpproxy_manage() combines both rtpproxy_offer() and rtpproxy_answer(). Now if i want different flags set for rtpproxy_offer() and rtpproxy_answer() then how should i use rtpproxy_manage()? To specific, what i want : route[REQUEST]{ ..... rtpproxy_offer("z20"); ... } onreply_route[]{ .... rtpproxy_answer("z120"); .... } Now in kamailio.cfg there is one call rtpproxy_manage() in route[NATMANAGE]. route[NATMANAGE] is called from both route[REQUEST] and onreply_route[]. So i can't emulate the functionality described above. My question is using only rtpproxy_manage() is it possible to emulate the above functionality? Thanks in advance. -- -aft

3 3

Homer multiple domains
by Reda Aouad 11 Nov '12

11 Nov '12

Hi I am using homer and managing multiple SIP domains. My problem is that in the sip_capture table, the FROM and TO domains are not logged in the FROM and TO fields. The MSG field contains the domains, but the queries involving domains become tricky and heavy to execute. Is there a way I can have the domains in their respective FROM/TO columns? Thanks Reda

1 0

how to use "t_drop_replies" but keep the last reply for acc
by Uri Shacked 11 Nov '12

11 Nov '12

Hi, I am trying to make an option of "route when no answer" or " route when busy". What I am doing is checking the reply and if "busy", for example, I use "t_drop_replies". Then, I set the new number and route[relay] again. On the accdb table, I get the first invite with 183 and after that the second invite with 183 and with 200. I would like to do exactly what i do, but would like to see on the accdb the 486 reply from the first invite. how do i do it? BR, Uri

4 11

Kamailio language definitions- dictionary
by kellon＠sis-grenada.com 11 Nov '12

11 Nov '12

Hi All Can anyone direct me to some go literature/documentation on the language/functions used in the kamailio config file. E.g cfg_get(), strempty(), proxy_authorize(), sl_send_reply() etc.... Thanks Kellon

2 1

No Debug Log message
by kellon＠sis-grenada.com 10 Nov '12

10 Nov '12

Hi all I am currently having to modify the kamailio config file without the benefit of log messages in /var/log/message. I am using the default configuration parameter settings of: #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 Has anyone experience this issue before or have any suggestions on howto fix. Making changes to the config file line by line and restarting the daemon is killing me!!!

2 1

Kamailio CDR logging to RADIUS
by Owen Lynch 10 Nov '12

10 Nov '12

Hi, I have transaction accounting to RADIUS setup and get the start and stop events logged to file when a call is made. I want to extend this to also get CDR logs - call duration etc. To do this I understand I need to configure the acc module and load and configure the dialog module: #!define FLD_DLG 4 modparam("acc", "cdr_enable", 1) modparam("dialog", "dlg_flag", FLD_DLG) I was expecting extra entries in the RADIUS file but they don't appear. I think this is because I need to set and clear the dialog flag at appropriate places in the route script (e.g. setflag(FLD_DLG)) but I'm not sure where. Should this be done in the INVITE and BYE method handling as is currently done for the transaction accounting, e.g.: # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } Thanks, Owen Lynch

3 4

"branch" tag in the "Via" header of the ACK message for the re-INVITE
by Alex Solt 10 Nov '12

10 Nov '12

Hi, We have a problem with the "branch" tag in the "Via" header of the ACK message for the re-INVITE. The "branch" tag value is not a unique number so the end point thinks the ACK is belong to the previous call. This causes the active calls fail after 15 min (right after the ACK message for re-INVITE passed by kamailio proxy). Is there a way to set the "branch" tag in the "Via" header of the ACK message for the re-INVITE to be a unique number? PS1: We are running kamailio v3.3.PS2: we did not have this issue with the old version of openser. Thanks,AS

2 3

Dispatcher question
by Mino Haluz 10 Nov '12

10 Nov '12

Hi, I have simple dispatcher that is dispatching initial invites to the core router. The question is, how can I tell kamailio to not add the Via header in the INVITE. I mean, I want that this dispatcher will be completely transparent to the core router, so any responses will go directly to the original sender, not to the dispatcher. This is my request route: request_route { if (is_method("INVITE")) { if (!ds_select_domain(1, 0)) { xlog("L_INFO", "[MAIN] ERROR: Proxy1 failed"); if (!ds_select_domain(2, 0)) { xlog("L_INFO", "[MAIN] FATAL: Proxy2 failed"); exit; } } xlog("L_INFO","[MAIN] Dispatching INVITE to $rd"); forward(); } } 1 and 2 are defined in dispatcher.list. If it can be done in kamailio, please how. Thank you! Mino

3 5

presence module makes excessive cleanups when subs_db_mode=3
by Andrew Pogrebennyk 09 Nov '12

09 Nov '12

Hi, when presence module is running with subs_db_mode=3 it makes an excessive number of SQL select queries, litelly dozens per second: > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=64 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=116 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=116 AND event='presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=64 AND event='presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=14 AND event<>'presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=14 AND event='presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=118 AND event<>'presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=66 AND event<>'presence.winfo' > 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=118 AND event='presence.winfo' > 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=66 AND event='presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=16 AND event<>'presence.winfo' > 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=16 AND event='presence.winfo' Steps to reproduce: registers two subscribers in jitsi and add them to contact lists of each other. The kamailio version is 3.3.2. I'm not doing anything special configuration-wise: loadmodule "presence.so" modparam("presence", "db_url", "mysql://kamailio:snbF93@localhost/kamailio") # in 3.3 the fallback2db change to subs_db_mode modparam("presence", "subs_db_mode", 3) modparam("presence", "notifier_processes", 3) loadmodule "presence_xml.so" modparam("presence_xml", "db_url", "mysql://kamailio:snbF93@localhost/kamailio") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) # retry-after 5 minutes modparam("presence_xml", "xcapauth_userdel_reason", "probation;retry-after=300") Q: it is a bug of a feature? :) Thanks. Andrew

2 2

xmlrpc dlg.end_dlg problem
by Kristofer Signer 09 Nov '12

09 Nov '12

Hi all, I have an kamailio 3.3.2 server deployed with dialog awareness. When call dlg.end_dlg over xmlrpc I get an 200 OK response but the dialog is not ended. What I can do; call dlg.list (xmlrpc) and get the active dialog list, xmlrpc interface working. call dlg.dlg_list (xmlrpc) and get description of one dialog, xmlrpx method call with parameters working. call dlg_end_dlg (kamctl) and successfully end the dialog. so my questions are... Is the xmlrpc method dlg.end_dlg known to work? and if, what have I missed? thanks ./Kristofer Signer $ /usr/local/kamailio-3.3/sbin/kamailio -V version: kamailio 3.3.2 (x86_64/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, select, kqueue. id: unknown compiled on 09:53:26 Nov 2 2012 with gcc 4.2.1

2 3

dispatcher: What's "first destination" in a database
by Olle E. Johansson 08 Nov '12

08 Nov '12

Friends, Dispatcher has a mode 8 - "use first destination" which makes sense for a textfile driven dispatcher configuration. Now if I use a database - will dispatcher first sort on priority, then take the first? It seems so from the documentation of the text file: priority: sets the priority in destination list (based on it is done the initial ordering inside the set) Which one is top priority - lowest or highest value? I can update the docs to clarify if someone helps me with the answer :-) /O

2 1

Another NAT question
by Moacir Ferreira 08 Nov '12

08 Nov '12

Hi, I have a scenario where I got a firewall with 3 interfaces: internal, DMZ and external. All the traffic from internal going to external is NATed. However, the traffic between internal and DMZ is NOT NATed. The external and DMZ are using public IP addresses. On the DMZ I have a Kamailio server running with RTPProxy + NAT Helper. Everything works fine when public (from the internet) users (UAs) are behind NAT as Kamailio will force the RTP to go via RTPProxy. However, when the public user has a public IP, it will fail to establish the RTP to a user who registered on Kamailio coming from the internal firewall interface. The reason is that, as I don't do NAT between internal and DMZ firewall interfaces, Kamailio will not RTPRroxy in between the UAs because, from the way Kamailio detects NAT, they are not behind NAT. So the public user UA tries to reach a private IP address. If the "inside" user tries to call a public user (a user on the Internet with a public IP), it works. Yes, I could do NAT in between the internal and DMZ firewall interfaces. However, this would force all RTP traffic of my UAs at the LAN go to Kamailio RTPProxy, an bad effect that I would like to avoid. So my question is: what would be the best approach to solve this issue using Kamailio and RTPProxy in such scenario? Cheers! Moacir

2 4

siptrace buffer
by Martin Formanko 08 Nov '12

08 Nov '12

Hi, do you know what happens if the database for siptrace is not reachable ? I assume siptrace is non-blocking in new versions of kamailio, but is there any buffer? Or insert queries are discarded when timed out ? Mino

2 1

Gateway failover on slow call setup
by Spencer Thomason 08 Nov '12

08 Nov '12

Hello, I'm trying to address a problem where sometimes our upstream providers have slow call setup times. We use drouting with multiple providers for each route to handle failover in the event of an outage but in this case, we get a 100 Trying but sometimes a 18[03] response takes 4-5 seconds or we never get it because the caller cancels the request. What would be a "safe" amount of time to allow a PSTN gateway to reply with a 180/183? What I'm trying to avoid is a scenario where the PSTN legitimately takes a long time to setup (i.e. an international call to some cell phone providers) and we needlessly failover which will actually increase the post dial delay for the user. My thought is this for example: route[PSTN] { t_set_fr(1000); # set fr timer to 1 second so we failover quickly if no provisional response t_on_reply("PSTN_REPLY"); # Routing logic } onreply_route[PSTN_REPLY] { if (status=~"1[0-9][0-9]") { t_set_fr(4000); # set fr timer to 4 seconds } if (status=~"18[03]") { t_reset_fr(); # reset timer if we get a 180 or 183 } } Also, if we failover because of slow call setup for a particular route, I'd like to temporarily avoid that gateway for that route (but only that route). I'm curious if anyone has any pointers or how they solved this. Thanks, Spencer

2 1

Adding Users not excepting email option
by kellon＠sis-grenada.com 08 Nov '12

08 Nov '12

I have attempted to add users as per documentation. kamctl add <username> <password> <email> However, Kamailio responds with the following: -- command 'add|passwd|rm' - manage subscribers add <username> <password> .......... add a new subscriber (*) passwd <username> <passwd> ......... change user's password (*) rm <username> ...................... delete a user (*) Is the option to add user with email address etc still available in the current release of 3.3 Thanks Kellon

2 1

Kamailio not responding
by kellon＠sis-grenada.com 07 Nov '12

07 Nov '12

Thanks Anton, Fred, Jeremy thanks for the suggestions. The problem was the firewall. I don't know why I didn't think of it. I learnt one other thing also the use of kamctl monitor. Just two other problems at the moment. I'll start a new thread for them

1 0

TLS Certificate Verification Issue
by Kamal Palei 07 Nov '12

07 Nov '12

Dear All I have modified kamailio,cfg and compiled all the modules with TLS enabled, and able to bring up the kamailio proxy properly. Kamailio proxy will receive the REGISTER message from endpoints in UDP , and want to send this REGISTER message to another intermediate proxy in TLS. For this purpose, I have added few lines in kamailio.cfg file as below. I have created the certificates, private keys as explained by README file in kamailio-3.1.5/modules/tls/ path. if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:443"); exit(); } Looks like this is taking effect. When Kamailio receives REGISTER message it is trying to do handshake with intermediate proxy. I used wireshark to see the handshake messages. 1. From Kamailio proxy, a TCP SYNC message is going to intermediate proxy. 2. intermediate proxy sends SYNC + ACK 3. Kamailio sends CLIENT HELLO 4. intermediate proxy sends SERVER HELLO, CERTIFICATE and SERVER HELLO DONE 5. The Kamailio sends ALERT (Level: Fatal, Description: Unknown CA) ---> IS something going wrong here.............. 6. Then Kamailio sends FIN + ACK Can somebody please let me know why the certificate verification fails (I get this log in console). How can I put a work around to avoid certification verification failure. Best Regards kamal

2 10

Kamailio not responding
by kellon＠sis-grenada.com 07 Nov '12

07 Nov '12

I have recently installed kamailio on CentOS 6.3 and configured a couple of SIP phone Linksys and Grandstream to test inercommunication. The SIP-Server/ Kamailio server ha started and is listen on port 5060. The UA are sending registered message to the SIP Server but there is no response from the SIP server. I left the syslog at default but not seeing any messages on the /var/log/messages. Can someone tell me if I have missed some configuration information. Note: I registered two users 101 and 202 using the command kamctl add 101 101. Is it possible to check whether the Users added have been added to the system??. and does anyone have any idea how to to troubleshoot a response from the Server. I am using Wireshark to monitor communication between between UA and SIP Server

4 3

Problem with my kamailio installation
by Ramazan Yilmaz 06 Nov '12

06 Nov '12

Hi, I installed kamailio 3.2.x a few times before and they were really fine. I decided to install a fresh one and followed the instructions on http://www.kamailio.org/wiki/install/3.3.x/git. However there is a problem: clients cannot register to kamailio. Each registration attempt produces the following error in the syslog: Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [tcp_main.c:3578]: BUG: handle_ser_child: fd -1 for 0 (pid 32414) Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [pass_fd.c:103]: ERROR: recv_all: 1st recv on 56 failed: Bad file descriptor Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [tcp_main.c:3363]: ERROR: handle_tcp_child: read from tcp child 0 (pid 0, no 0) Bad file descriptor [9] Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 17 on 56: Bad file descriptor (9) Oct 29 20:07:38 ubuntu /usr/local/kamailio-3.3/sbin/kamailio[32421]: ERROR: <core> [tcp_main.c:4006]: ERROR: send2child: send_fd failed for 0x7f510bc41588 (flags 0x4018), fd 17 What could be the problem? Any suggestion? In case you need, the server is ubuntu server 11.10, and I installed including mysql and tls modules.

2 15

ERROR: presence not working correctly with Kamailio 3.3.2-1.1 and Postgresql 9.1
by Robert Verspuy 05 Nov '12

05 Nov '12

Hi All, I've got a problem with Kamailio 3.3.2-1.1 on CentOS 6.3 with locally postgresql 8.4.13-1.el6_3 libraries, connected to a remote postgresql91-9.1.6-1PGDP.rhel6 server The presence module tries to insert a record in the presentity table, with some xml in the body column. But the body value in the postgresql table is not represented as a string '<?xml ...etc' But encodes as '\x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462...etc' This gives problems when the presense module is trying to send out a notify based on the value in the database. With debugging, this gives the error: Entity: line 1: parser error : Start tag expected, '<' not found x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462d38273f3e3c The \x in front of the hex encodes string is not properly interpreted by postgresql or kamailio. The body column of the table presentity is a bytea column, and I believe postgresql 9 outputs these bytea columns a bit differently then postgresql 8. When I try to run the sql statement: insert into presentity (domain,username,event,etag,sender,body,received_time,expires ) values ('newsip.lifexs.nl','00086','presence','a.1352107949.18632.27.0','','<?xml version=''1.0'' encoding=''UTF-',1343534532,1345213723) On the postgresql 9.1 server locally, I still get the '\x3c3f786d6c... etc' value in the database. The bytea_output setting in postgresql 9 is now standard set to 'hex' in stead of escape. I've changed the bytea_output setting in postgresql.conf to 'escape', and then the presence is working without any issues. So maybe the database module of postgresql has to be changed to pick up these bytea encodings properly? Because i think kamailio should work correctly with the default postgresql settings. With kind regards, Robert Verspuy -- *Exa-Omicron* Eenspan 8-K 3897 AL Zeewolde http://exa.nl

2 1

Kamailio sending re invite to the next proxy
by Maurice Vistan 05 Nov '12

05 Nov '12

Hi All, I am using kamailio as dispatcher and set up kamailio just to send 200 okay back for the method register, I would like check if its normal for the kamailio send another invite on the next proxy listed on my dispatcher.lst after the first proxy process the invite and auth. Below is how the call is being process. user kamailio 1proxy 2proxy user2 invite |----------> | 100 | <---------- | invite | | ---------> | 100 |<--------- | auth |<-----------| auth | <------------------------| ack |------------->| ---------->| Invite |-------------->| --------------------------> | then process Is there any I can limit this? or it is realy the behavior? Thanks a lot! Maurice

2 2

ERROR: presence not working correctly with Kamailio 3.3.2-1.1 and Postgresql 9.1
by Robert Verspuy 05 Nov '12

05 Nov '12

Hi All, I've got a problem with Kamailio 3.3.2-1.1 on CentOS 6.3 with locally postgresql 8.4.13-1.el6_3 libraries, connected to a remote postgresql91-9.1.6-1PGDP.rhel6 server The presence module tries to insert a record in the presentity table, with some xml in the body column. But the body value in the postgresql table is not represented as a string '<?xml ...etc' But encodes as '\x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462...etc' This gives problems when the presense module is trying to send out a notify based on the value in the database. With debugging, this gives the error: Entity: line 1: parser error : Start tag expected, '<' not found x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462d38273f3e3c The \x in front of the hex encodes string is not properly interpreted by postgresql or kamailio. The body column of the table presentity is a bytea column, and I believe postgresql 9 outputs these bytea columns a bit differently then postgresql 8. When I try to run the sql statement: insert into presentity (domain,username,event,etag,sender,body,received_time,expires ) values ('newsip.lifexs.nl','00086','presence','a.1352107949.18632.27.0','','<?xml version=''1.0'' encoding=''UTF-',1343534532,1345213723) On the postgresql 9.1 server locally, I still get the '\x3c3f786d6c... etc' value in the database. The bytea_output setting in postgresql 9 is now standard set to 'hex' in stead of escape. I've changed the bytea_output setting in postgresql.conf to 'escape', and then the presence is working without any issues. So maybe the database module of postgresql has to be changed to pick up these bytea encodings properly? Because i think kamailio should work correctly with the default postgresql settings. With kind regards, Robert Verspuy -- *Exa-Omicron* Eenspan 8-K 3897 AL Zeewolde http://exa.nl

1 0

Entries in PUA List and RLS Watchers List
by Sangeeta Shah 03 Nov '12

03 Nov '12

Hello All, I am hoping the authors of the RLS/PUA modules can answer this question since I am not sure what's going on. I have the rls module enabled, with the following config spec: # ------rls module params ------ modparam("rls", "db_url", DBURL) modparam("rls", "db_mode", 2) modparam("rls", "integrated_xcap_server", 1) modparam("rls", "to_presence_code" ,10) modparam("rls", "server_address", "sip:rls@ip:5060") #!endif and modparam("pua_reginfo", "default_domain", "ip") modparam("pua_reginfo", "server_address", "sip:reginfo@ip") modparam("pua", "db_mode", 2) So for both I have DB_MODE set to DB ONLY. For whatever reason, even though all I am doing is bringing up my client I see over 250+ entries in the rls_watchers table for the same watcher and over 100 entries in the pua list table for the same presentity? Anyone have any idea what would be triggering this. There is definitely not anymore more than the normal messaging going on between the client and the server. Only error I see in the syslog: Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: db_mysql [km_dbase.c:122]: driver error on query: Duplicate entry '1350937406' for key 'expires_idx' Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11744]: DEBUG: presence [subscribe.c:1216]: subs->contact= sip:rls@ip:5060 - len = 25 Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11761]: DEBUG: <core> [db_val.c:117]: converting STRING [8475551001] Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11746]: DEBUG: <core> [parser/msg_parser.c:626]: method: <SUBSCRIBE> Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: <core> [db_query.c:210]: error while submitting query Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11744]: DEBUG: rls [rls.h:241]: did= 41826031568472f9-11752@127.0.0.1;533cb9e91f4b999cf76861cbb9ed54ed-93a3;a6a1c5f60faecf035a1ae5b6e96e979a-464d Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11761]: DEBUG: <core> [db_val.c:117]: converting STRING [10.50.251.12] Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11746]: DEBUG: <core> [parser/msg_parser.c:628]: uri: <sip:8475551001@ip> Oct 22 15:22:22 RCS-Presence /usr/local/sbin/kamailio[11745]: ERROR: pua [pua_db.c:1149]: DB insert failed -- Sangeeta Shah

5 18

Error while "Make All"
by Techie Sup 02 Nov '12

02 Nov '12

Hello, I followed the instruction mentioned in Kamailio website. When I do a make all, I get the error. Please let me know how to fix this. I am new to Kamailio. ------------------- Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found Package mono-2 was not found in the pkg-config search path. Perhaps you should add the directory containing `mono-2.pc' to the PKG_CONFIG_PATH environment variable No package 'mono-2' found CC (gcc) [M app_mono.so] app_mono_api.o In file included from app_mono_api.c:37: app_mono_api.h:27:26: warning: mono/jit/jit.h: No such file or directory app_mono_api.h:28:39: warning: mono/metadata/environment.h: No such file or directory In file included from app_mono_api.c:37: app_mono_api.h:35: error: expected specifier-qualifier-list before ‘MonoDomain’ app_mono_api.c:39:39: warning: mono/metadata/mono-config.h: No such file or directory app_mono_api.c:49: error: expected specifier-qualifier-list before ‘MonoDomain’ app_mono_api.c: In function ‘sr_mono_load_script’: app_mono_api.c:93: error: ‘sr_mono_load_t’ has no member named ‘next’ app_mono_api.c: In function ‘mono_sr_init_load’: app_mono_api.c:139: warning: implicit declaration of function ‘mono_config_parse’ app_mono_api.c:141: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:148: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:148: warning: implicit declaration of function ‘mono_jit_init’ app_mono_api.c:149: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:157: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:157: warning: implicit declaration of function ‘mono_domain_assembly_open’ app_mono_api.c:157: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:158: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:162: error: ‘sr_mono_load_t’ has no member named ‘next’ app_mono_api.c: In function ‘mono_sr_initialized’: app_mono_api.c:208: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c: In function ‘app_mono_exec’: app_mono_api.c:230: error: ‘sr_mono_env_t’ has no member named ‘msg’ app_mono_api.c:237: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:246: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:246: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:247: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:256: warning: implicit declaration of function ‘mono_jit_exec’ app_mono_api.c:256: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:256: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:257: warning: implicit declaration of function ‘mono_environment_exitcode_get’ app_mono_api.c:261: warning: implicit declaration of function ‘mono_jit_cleanup’ app_mono_api.c:261: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c: In function ‘app_mono_run’: app_mono_api.c:282: error: ‘sr_mono_env_t’ has no member named ‘msg’ app_mono_api.c:284: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:284: error: ‘sr_mono_load_t’ has no member named ‘domain’ app_mono_api.c:285: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:285: error: ‘sr_mono_load_t’ has no member named ‘assembly’ app_mono_api.c:286: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c:291: warning: implicit declaration of function ‘mono_domain_set’ app_mono_api.c:291: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:298: error: ‘sr_mono_env_t’ has no member named ‘domain’ app_mono_api.c:298: error: ‘sr_mono_env_t’ has no member named ‘assembly’ app_mono_api.c: At top level: app_mono_api.c:310: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ token app_mono_api.c:314: error: expected declaration specifiers or ‘...’ before ‘MonoString’ app_mono_api.c: In function ‘sr_mono_log’: app_mono_api.c:316: warning: implicit declaration of function ‘mono_string_to_utf8’ app_mono_api.c:316: error: ‘text’ undeclared (first use in this function) app_mono_api.c:316: error: (Each undeclared identifier is reported only once app_mono_api.c:316: error: for each function it appears in.) app_mono_api.c:316: warning: assignment makes pointer from integer without a cast app_mono_api.c:318: warning: implicit declaration of function ‘mono_free’ app_mono_api.c: At top level: app_mono_api.c:321: error: expected ‘)’ before ‘*’ token app_mono_api.c:328: error: expected ‘)’ before ‘*’ token app_mono_api.c:338: error: expected ‘)’ before ‘*’ token app_mono_api.c:408: error: ‘sr_mono_api_version’ undeclared here (not in a function) app_mono_api.c:410: error: ‘sr_mono_err’ undeclared here (not in a function) app_mono_api.c:411: error: ‘sr_mono_dbg’ undeclared here (not in a function) app_mono_api.c:412: error: ‘sr_mono_modf’ undeclared here (not in a function) app_mono_api.c: In function ‘sr_mono_load_class_core’: app_mono_api.c:420: warning: implicit declaration of function ‘mono_add_internal_call’ app_mono_api.c: At top level: app_mono_api.c:427: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘*’ token app_mono_api.c:478: error: expected ‘)’ before ‘*’ token app_mono_api.c:530: error: expected ‘)’ before ‘*’ token app_mono_api.c:579: error: expected ‘)’ before ‘*’ token app_mono_api.c:632: error: expected ‘)’ before ‘*’ token app_mono_api.c:679: error: expected ‘)’ before ‘*’ token app_mono_api.c:729: error: ‘sr_mono_pv_gets’ undeclared here (not in a function) app_mono_api.c:729: error: initializer element is not constant app_mono_api.c:729: error: (near initialization for ‘_sr_M_export_pv[0].method’) app_mono_api.c:730: error: ‘sr_mono_pv_geti’ undeclared here (not in a function) app_mono_api.c:730: error: initializer element is not constant app_mono_api.c:730: error: (near initialization for ‘_sr_M_export_pv[1].method’) app_mono_api.c:731: error: ‘sr_mono_pv_seti’ undeclared here (not in a function) app_mono_api.c:731: error: initializer element is not constant app_mono_api.c:731: error: (near initialization for ‘_sr_M_export_pv[2].method’) app_mono_api.c:732: error: ‘sr_mono_pv_sets’ undeclared here (not in a function) app_mono_api.c:732: error: initializer element is not constant app_mono_api.c:732: error: (near initialization for ‘_sr_M_export_pv[3].method’) app_mono_api.c:733: error: ‘sr_mono_pv_unset’ undeclared here (not in a function) app_mono_api.c:733: error: initializer element is not constant app_mono_api.c:733: error: (near initialization for ‘_sr_M_export_pv[4].method’) app_mono_api.c:734: error: ‘sr_mono_pv_is_null’ undeclared here (not in a function) app_mono_api.c:734: error: initializer element is not constant app_mono_api.c:734: error: (near initialization for ‘_sr_M_export_pv[5].method’) app_mono_api.c:750: error: expected ‘)’ before ‘*’ token app_mono_api.c:801: error: expected ‘)’ before ‘*’ token app_mono_api.c:850: error: expected ‘)’ before ‘*’ token app_mono_api.c:895: error: expected ‘)’ before ‘*’ token app_mono_api.c:930: error: ‘sr_mono_hdr_append’ undeclared here (not in a function) app_mono_api.c:930: error: initializer element is not constant app_mono_api.c:930: error: (near initialization for ‘_sr_M_export_hdr[0].method’) app_mono_api.c:931: error: ‘sr_mono_hdr_remove’ undeclared here (not in a function) app_mono_api.c:931: error: initializer element is not constant app_mono_api.c:931: error: (near initialization for ‘_sr_M_export_hdr[1].method’) app_mono_api.c:932: error: ‘sr_mono_hdr_insert’ undeclared here (not in a function) app_mono_api.c:932: error: initializer element is not constant app_mono_api.c:932: error: (near initialization for ‘_sr_M_export_hdr[2].method’) app_mono_api.c:933: error: ‘sr_mono_hdr_append_to_reply’ undeclared here (not in a function) app_mono_api.c:933: error: initializer element is not constant app_mono_api.c:933: error: (near initialization for ‘_sr_M_export_hdr[3].method’) make[1]: *** [app_mono_api.o] Error 1 make: *** [modules] Error 1 Regards Selvam

3 4

Kamailio WebSocket support
by Owen Lynch 02 Nov '12

02 Nov '12

Hi, I see Peter Dunkley's websocket work is not in the latest 3.3 release (3.3.2). What is the road map for this? Thanks, Owen Lynch

2 1

Re: [SR-Users] Kamailio and sylkserver integration
by Vijay Thakur 01 Nov '12

01 Nov '12

Thanks for your reply. Will you please explain that how can i forward calls from kamailio to sylkserver. Regards, VJ++ Message: 7 Date: Wed, 31 Oct 2012 08:26:50 +0100 From: "Olle E. Johansson"<oej(a)edvina.net> Subject: Re: [SR-Users] Kamailio and sylkserver integration To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List"<sr-users(a)lists.sip-router.org> Message-ID:<D7F883EA-4231-4D21-A79F-BB3449F972B1(a)edvina.net> Content-Type: text/plain; charset=us-ascii 31 okt 2012 kl. 06:49 skrev Vijay Thakur<vijay.thakur(a)loopmethods.com>: > Dear Sir, > > Is there any one in the list who have set up (integrated) Kamailio with sylkserver for conference. There is very less available information on the internet on this topic. All the required changes are made in sylkserver config files. > Kindly advise for required changes in Kamailio. As far as I understand, Sylkserver is just a SIP server. Forward calls from kamailio to sylkserver using any of the forwarding commands. /O -- * Olle E. Johansson -oej(a)edvina.net * Kamailio & SIP Masterclass Miami FL December 2012 *http://edvina.net/training/

2 1

Two kamailio servers using same DB but different version table
by Uri Shacked 31 Oct '12

31 Oct '12

Hi, can two kamailio servers, one running 3.2 and the other 3.3 run using the same DB? one will turn to version table X and the other to version table Y (i am not using the modules that the DB table changed... BR, Uri

2 1

statefull proxy Q
by Alex Solt 31 Oct '12

31 Oct '12

Hi, Is there any function in kamailio that could differentiates between a new incoming INVITE message and a re-INVITE message for an existing active call? Thanks, AS

3 6

2600Hz selects Kamailio
by Olle E. Johansson 31 Oct '12

31 Oct '12

Friends, The 2600Hz project that builds an API-driven telephony service have selected to switch to Kamailio as a SIP proxy for their SBC. They blog about the switch, comparing Kamailio with their previous choice and coming to a conclusion: "To our minds, a more active development community means that other users are likely to help solve bottlenecks for us, so we spend less time worrying about our SBC, and more time writing fun new applications! Conclusion: We’re switching to Kamailio because: • Many of our Major Clients already use it • The Naming Conventions are more sensible • The Pace of Innovation on Kamailio is faster" Join me in welcoming 2600Hz to the Kamailio project! /O Links: * http://www.kamailio.org/w/2012/10/2600hz-selects-kamailio/ * http://blog.2600hz.com/post/34639934434/our-other-border-brother-kamailio

1 0

Kamailio and sylkserver integration
by Vijay Thakur 31 Oct '12

31 Oct '12

Dear Sir, Is there any one in the list who have set up (integrated) Kamailio with sylkserver for conference. There is very less available information on the internet on this topic. All the required changes are made in sylkserver config files. Kindly advise for required changes in Kamailio. With Regards, Vijay Thakur

2 1

additional user data for SIP gateway usage?
by Jeremy Ardley 31 Oct '12

31 Oct '12

Hi, I'm after advice. I'm developing a script to provide a degree of PBX functionality for a small to medium office using Kamailio & SEMS. This includes mapping internal extensions to external PSTN numbers via the SIP trunk provider. The incoming mapping is easy using the db_alias functionality. The outgoing mapping is more difficult as the SIP trunk provider requires both the Full National Number and an account number matching the FNN in the outgoing From header. I'd appreciate suggestions on the best way to store, retrieve, and manage the additional information required for outgoing PSTN calls. I think avpops will work but is there some other way to do this? Thanks,

2 1

SIP connection tracking
by Daniil Baturin 31 Oct '12

31 Oct '12

Hi all, We have two softswitch hosts that don't support TLS. We have a peer that requires TLS, they terminate all our connections on a single address. We want to use kamailio as a proxy for it. Could you point me to the documentation and/or working examples that describe how to implement connection tracking to ensure replies to connections originated from host A are relayed back to host A, and connections from host B get back to host B? Thanks in advance. -- #!/usr/bin/env perl @a=split(//, "daniil @ baturin . org" );# Daniil Baturin @b=split(//,q/Px%!+o0Q6lh*7dp$.@8#%|y{/);while($i<24){$_.= chr((ord(@b[$i])-ord(@a[$i])+62)%94+32);$i++};print"$_\n"#

2 1

Re: [SR-Users] Blacklist initialization
by Javi Gallart 30 Oct '12

30 Oct '12

Hi Olle What about doing the init stuff in event-route[htable:mod-init] ? Regards Javi ------Original Message------ From: Olle E. Johansson Sender: sr-users-bounces(a)lists.sip-router.org To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List ReplyTo: SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing List Subject: [SR-Users] Blacklist initialization Sent: Oct 30, 2012 21:58 Hi! I am looking for a way to add addresses to the IP blacklist in Kamailio at startup. Is that possible? I know I can do it with sercmd, but I fail to find a documented way to do it in the configuration... Must be missing something simple. Thank you for your help! /O _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Blacklist initialization
by Olle E. Johansson 30 Oct '12

30 Oct '12

Hi! I am looking for a way to add addresses to the IP blacklist in Kamailio at startup. Is that possible? I know I can do it with sercmd, but I fail to find a documented way to do it in the configuration... Must be missing something simple. Thank you for your help! /O

1 0

Default database named now kamailio
by Daniel-Constantin Mierla 30 Oct '12

30 Oct '12

Hello, just short note to let everyone know that the default database for Kamailio is now named 'kamailio' - the change happened over the weekend as part of migrating from last bits from openser to kamailio namespace. Also, the default usernames and passwords to access the database are now: - kamailio with password kamailiorw for read-write operations - kamailioro with password kamailioro for read-only operations If you set db_url parameters in the config file, there should be no impact on existing configurations. You also have to make sure you set the right value in kamctlrc file if you update kamctl. This change impacts development version (to be released as 3.4.0). Two other components are in the to-do list for this change: perl and snmpstats Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Question about acc module
by Thorsten Willand 30 Oct '12

30 Oct '12

3 2

Unknown status in presence_dialog
by David 29 Oct '12

29 Oct '12

Hello, I am using the presence_dialoginfo module on a stand alone presence server. I have this : if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; If the server does not have any status information, it will send out a blank notify with a 0 content length. Is there anyway to tell Kamailio to send an xml doc with status "terminated" if the server does not know the status that was requested by the SUBSCRIBE? Thanks, David Server : version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 14:19:23 Apr 19 2012 with gcc 4.4.3

1 0

siptrace vs. sipcapture
by Mino Haluz 29 Oct '12

29 Oct '12

Hi, we are thinking about switching from siptrace to sipcapture + HEP encapsulation. Do you know some reasons why we should switch to sipcapture or why we should not ? ;) We have multiple SIP proxies and Cisco gateways. In earlier versions of kamailio was siptrace blocking, that's why we started to looking for some other capturing mechanism. I think now siptrace is buffered, so we are hesitating .. Mino

2 1

rtpproxy ipv4/ipv6 bridging question
by Juha Heinanen 29 Oct '12

29 Oct '12

is it possible to make rtpproxy to include both ipv4 and ipv6 address in sdp? if not, i don't see how ipv4/ipv6 bridging can be made to work when $rd is domain name. -- juha

1 0

NEW: new module: xhttp_pi
by Ovidiu Sas 28 Oct '12

28 Oct '12

Hello all, A new module is available in trunk. It provides a web provisioning interface: http://kamailio.org/docs/modules/devel/modules/xhttp_pi.html A sample config example is provided in the source tree: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex… An additional xml file is required for web config framework (what to configure, where and how). A sample xml file is provided in the source tree for dispatcher and dialplan tables: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=ex… When kamailio is up and running, a web provisioning interface for managing db records will be available. No supplementary web server needs to be installed in order to manage the db records. Also, it provides a tight control over what can be modified and how. Testing and feedback is appreciated. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

2 1

web homer not displaying data
by Christopher Holland 26 Oct '12

26 Oct '12

I have sipcapture and siptrace setup on the same server using kamailio 3.4.0-dev4 and webhomer 3.3.0 setup to capture sip traffic from a mirriored port on the LAN. Interface ETH1 is to capture and ETH0 is for managment. The sipcapture database is getting data put into it. I can log into webhomer, but see no data in the graphes nor can I search. I have double checked my database permissions and they seem to be set up correctly. webserver is set up with: Apache/2.2.16 (Debian) PHP5 MYSQL Json phpmyadmin Here is a copy of my kamailio.cfg and configuration.php rev_dns=no auto_aliases=no debug=1 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=5 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* IP and port for HEP capturing) */ listen=udp:127.0.0.1:9060 mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" loadmodule "pv.so" loadmodule "db_mysql.so" loadmodule "sipcapture.so" loadmodule "textops.so" loadmodule "siptrace.so" # ----- mi_fifo params ----- ####### Routing Logic ######## modparam("sipcapture", "db_url", "mysql://openser:******@localhost/homerdb") modparam("sipcapture", "capture_on", 1) /* activate HEP capturing */ modparam("sipcapture", "hep_capture_on", 1) /* IP to listen. Port/Portrange apply only on mirroring port capturing */ modparam("sipcapture", "raw_socket_listen", "10.0.0.0:5060-5080") /* Name of interface to bind on raw socket */ modparam("sipcapture", "raw_interface", "eth1") /* activate IPIP capturing */ #modparam("sipcapture", "raw_ipip_capture_on", 1) /* My table name*/ modparam("sipcapture", "table_name", "sip_capture") /* children for raw socket */ modparam("sipcapture", "raw_sock_children", 4) /* insert delayed */ #modparam("sipcapture", "db_insert_mode", 1) /* activate monitoring/mirroring port capturing. Linux only */ modparam("sipcapture", "raw_moni_capture_on", 1) /* Promiscious mode RAW socket. Mirroring port. Linux only */ modparam("sipcapture", "promiscious_on", 1) /* activate Linux Socket Filter (LSF/BPF) on mirroring interface. Linux only */ #modparam("sipcapture", "raw_moni_bpf_on", 1) /* sipcapture node */ modparam("sipcapture", "capture_node", "homer01") # check IP and port of your capture node modparam("siptrace", "duplicate_uri","sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag",22) modparam("siptrace", "trace_on", 1) # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { sip_capture(); drop; } onreply_route { sip_capture(); drop; } ################################## configuration.php <?php /*********************************************************************************/ /* Access db of homer */ define('HOST', "localhost"); define('PORT', 3306); define('USER', "webhomer"); define('PW', "******"); define('DB', "web_homer"); /* Homer connection * this user must have the same password for all Homer nodes * please define all your nodes in homer_nodes table */ define('HOMER_HOST', "localhost"); /* DEFAULT. Don't forget insert this host to your DB nodes table */ define('HOMER_PORT', 3306); define('HOMER_USER', "openser"); define('HOMER_PW', "******"); define('HOMER_DB', "homerdb"); define('HOMER_TABLE', "sip_capture"); /*********************************************************************************/ /* webHomer Settings * Adjust to reflect your system preferences */ define('PCAPDIR',"/var/www/webhomer/tmp/"); define('WEBPCAPLOC',"/webhomer/tmp/"); define('APIURL',"http://localhost"); define('APILOC',"/webhomer/api/"); /* INCLUDE preferences */ include("preferences.php"); ?>

2 2

Message Routing To/From Gateway
by Ed Brady 26 Oct '12

26 Oct '12

Hi, I am trying to route SIP messages to a gateway. I am using SIP MESSAGE over UDP as well as the MSILO module for storing messages for users that aren't currently logged in. I have read through the Dynamic Routing module info but am not able to determine the best way to accomplish the routing I require. Our scenario is outlined below. User A sends a message addressed to "sip:12345678900@mydomain.com" Kamailio at mydomain.com checks if 12345678900 is a registered user in the system. If not, the message header is translated to "sip:12345678900@gatewaydomain..com" and forwarded. Incoming messages need to be accepted from their IP as well. It would also be useful if the address came as "sip:2345678900@mydomain.com" could be translated to "sip:*1*2345678900@gatewaydomain.com". Is there any built in functionality I can configure for this, or will this need to be scripted? Thanks, Ed

2 9

[OT] the role of SBCs
by Daniel-Constantin Mierla 26 Oct '12

26 Oct '12

Hello, based on the outcome of the discussion carried in the thread: http://lists.sip-router.org/pipermail/sr-users/2012-August/074480.html I am looking again the clarify some aspects out there in the VoIP world. So, as mainly dealing with proxy/sip signaling deployments, it's very often to be the first one hit by support issues claiming things don't work. Then you investigate and end up in conclusions like in the above thread: "The problem was at the SBC, __where I did not expect it__." The underlined part heats me up a bit, because I never understood from where it comes this perception that SBC is a MUST-TO-HAVE and the PERFECT (never mistaken or breaking things) node in a VoIP networks. To some extent, the SBC is just a very costly SIP ALG, and a SIP ALG is there to break the things. I don't want to start like a flame war, but is it something that I am obviously missing in regards to what benefits a SBC can bring? I see only inconveniences: - another point of failure - it is a b2bua, therefore very unlikely to offer the same performances of a proxy - if transcoding is needed, a media server can be used behind the proxy, properly protected of attacks by the proxy and eventually deployed as a farm load balanced by the proxy - if topology hinding is wanted in the b2bua fashion (not the proxy fashion with encoding headers), then the b2bua can be behind the proxy, properly protected of attacks by the proxy and eventually deployed as a farm load balanced by the proxy - nat traversal was solved long time ago in proxy environment, being scalable by deploying a farm of rtp proxy I don't want to go to other features, including the transport layer, it's a clear win of the proxy in my experience (ok, being deep involved in this project). Then, what makes the SBC so desirable in many companies/voip deployments? If any SBC user here that can share, what was the reason to buy such a device? Any conceptual functionality that cannot be achieved with the proxy as the first hop in front of the (wild) clients? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

10 14

MSRP: Generation of REPORTs on the relay
by Peter Dunkley 26 Oct '12

26 Oct '12

Hello, One of my colleagues at Crocodile has been doing some testing with the msrp module. With the auth change that was made yesterday, and the updated configuration I put in the README example it seems to mostly work. However, there is one set of scenarios relating to failure REPORTs that don't seem to be possible with the current module. RFC 4976 section 6.4.1 paragraph 3 states: If the Failure-Report header is "yes" or "partial", and if there is a problem processing the SEND request or if an error response is received for that SEND request, then the relay MUST respond with an appropriate error response in a REPORT back to the original source of the message. This means that when a SEND request that contains a Failure-Report: header with value of "yes" or "partial" is relayed by Kamailio we need to maintain some state so that when the TCP send fails, or we receive a failure response from the next hop, Kamailio can generate and send a REPORT. At the moment I don't think the msrp module has these capabilities. If the module is getting some re-work before the next release it would be good to get this in there. I am quite happy to help out in any way I can, but I am not sure of the best way to proceed with this. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Kamailio on Twitter: @kamailioproject
by Olle E. Johansson 26 Oct '12

26 Oct '12

The marketing arm of the Kamailio project has without further discussion opened up a Twitter account for Kamailio, to support our web site news, G+ and Facebook pages. Feel free to contribute with case stories and news we can publish on the web site and follow up in the social media flow. The twitter account is @kamailioproject - please follow it. And yes, Kamailio is THE best thing since the invention of sliced bread! This week's EXTRAORDINARY offer: Download TWO licenses of Kamailio for the same licensing cost as we've been offering for ONE license! And get an unlimited number of user licenses for your Kamailio SIP accounts! Kamailio greetings! Have a great SIP weekend! /Marketing

1 0

MSRP relay authentication
by Peter Dunkley 25 Oct '12

25 Oct '12

Hi, MSRP AUTH requests must be authenticated using HTTP Digest authentication. Part of the concatenated data when doing this authentication is the method name. Because of the way MSRP requests are formatted, the part of the request-line that would contain the method name in HTTP or SIP requests is always MSRP. The MSRP method name (AUTH in this case) is at the end of the request-line. When Kamailio converts an MSRP request to SIP it has a method name of MSRP. This means that when Kamailio authenticates an MSRP request it uses "MSRP" as the method name, not the actual MSRP method name (AUTH). Is this correct? Should MSRP requests be authenticated with a method name of "MSRP" (which kind-of makes sense as this is the string at the start of the MSRP request line - which is where the method names are in HTTP and SIP) or should the MSRP method name of "AUTH" be used? If it is the later, does anyone know of an easy way to add this to Kamailio? Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 4

dns queries on ipv6 addresses
by Juha Heinanen 25 Oct '12

25 Oct '12

from the mailing list archives i have learned that dns_try_ipv6=yes in config file is supposed to prevent dns query on ipv6 address. if i have it "no", i get to syslog: Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: INFO: Request URI is <sip:+35832345679@[2002:c062:670a::10]:4050;transport=udp> Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: ERROR: tm [ut.h:314]: ERROR: uri2dst: failed to resolve "[2002:c062:670a::10]" :unresolvable A or AAAA request (-7) Oct 24 14:07:35 siika /usr/sbin/sip-proxy[17958]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches and sip proxy naturally does not send anything out. if i have it "yes", request is sent out properly, but still sip proxy makes dns queries on the ipv6 address: Oct 24 14:08:35 siika /usr/sbin/sip-proxy[18196]: INFO: Request URI is <sip:+35832345679@[2002:c062:670a::10]:4050;transport=udp> Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail Oct 24 14:08:35 siika pdns[4269]: Received a malformed qdomain from 127.0.0.1, '[2002:c062:670a::10]': sending servfail why is that? how to completely prevent dns queries on ipv6 addresses? -- juha

6 16

rtpproxy delete fails
by Juha Heinanen 25 Oct '12

25 Oct '12

i made rtpproxy test in setup where two sip phones have registered the same AoR test(a)test.fi. one is behind nat and the other is not. when i call this AoR, my sip proxy executes rtpproxy_manage("FROW3"); in branch route of the branch that is behind nat. syslog shows: Oct 19 15:49:59 siika /usr/sbin/sip-proxy[18596]: INFO: Calling rtpproxy_manage on INVITE <sip:test@192.168.0.31:5074;transport=tcp> Oct 19 15:49:59 siika mediaproxy-ng[12832]: Got valid command from udp:127.0.0.1:48714: 18596_16 US ptonjivixvatrhz(a)siika.tutpro.com;z9hG4bKqxklorkm 192.98.103.10 8000 wnzdf;1 Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com] Creating new call Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Opened ports 50104/50105 for RTP Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Opened ports 50106/50107 for RTP Oct 19 15:49:59 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com - z9hG4bKqxklorkm] Returning to SIP proxy: 18596_16 50104 192.98.103.10 4 then the uas hehind nat replies with 480 and sip proxy executes in onreply route rtpproxy_manage("FROW3"); that fails and i get to syslog: Oct 19 15:50:15 siika mediaproxy-ng[12832]: Got valid command from udp:127.0.0.1:56183: 18594_12 D ptonjivixvatrhz(a)siika.tutpro.com;z9hG4bKqxklorkm wnzdf shsqn Oct 19 15:50:15 siika mediaproxy-ng[12832]: [ptonjivixvatrhz(a)siika.tutpro.com] Tags didn't match for delete message, ignoring looks like this mismatch is due to the missing ";1" in from tag param that was there when rtpproxy_manage was executed on invite. if my theory is correct, why is ";1" missing? if not correct, what did i do wrong? -- juha

4 17

udp transport in rtpproxy sockets
by Juha Heinanen 25 Oct '12

25 Oct '12

rtpproxy module readme gives impression that udp transport is the only possibility: 4.1. rtpproxy_sock (string) Definition of socket(s) used to connect to (a set) RTPProxy. It may specify a UNIX socket or an IPv4/IPv6 UDP socket. if that is the case, what is the justification for it, i.e, why is tcp (and better yet tls) transport not supported? -- juha

3 3

rtpproxy module flags 1 and 2 question
by Juha Heinanen 25 Oct '12

25 Oct '12

i'm new to rtpproxy (have been using mediaproxy for years) and now try to understand if it would make sense to switch over. rtpproxy readme has this on flags value 1: + 1 - append first Via branch to Call-ID when sending command to rtpproxy. ... This is especially useful if you have serially forked call scenarios where rtpproxy gets an "update" command for a new branch, and then a "delete" command for the previous branch, which would otherwise delete the full call, breaking the subsequent "lookup" for the new branch. how it would be possible in SERIAL forked scenario to get anything for a new branch before previous branch has first failed and gone? -- juha

4 7

Manually add Via Header
by Vassilis Radis 25 Oct '12

25 Oct '12

I have a situation where a far end SIP provider doesn't behave properly when sending 487 replies. The scenario is this: I have a registered user calling into my kamailio which ,using lcr module, routes the call to a SIP provider. When the caller Cancels the call, my kamailio forwards the cancel message to the provider (along with a 200 Cancelling message to the caller). The problem is with the 487 sent by the provider to my kamailio: It does not contain all the two Via headers. It only contains the via header that names my kamailio. So when kamailio gets the 487, it removes its Via header and sends the 487 back to the caller, which now doesnt find any via header. Note that the far end provider, correctly sends all the Via Headers in other replies (183 etc). So now I am trying to "intercept" the 487 from this specific provider in the onreply route, and patch it with the header that the provider should have included. How can I do it? The problem is twofold: 1.I need to detect which header I should manually add (maybe store the Via Header set from the initial invites or the 183 replies, and get it from there? .I dont know how to do that) 2.I need to add it. (How can I manually add a Via header? .This seems easier Can anyone help me, maybe with another solution?

4 5

ip address fields
by Juha Heinanen 25 Oct '12

25 Oct '12

lcr_gw table has: ip_addr varchar(47) trusted table has: src_ip varchar(50) address table has: ip_addr varchar(48) why is it that they are not all the same? also, why the common length is not 8 * 4 + 7 = 39 (http://en.wikipedia.org/wiki/IPv6#Address_format)? -- juha

4 19

Kamailio & OMA Presence Compliance
by Thomas Hoffman 24 Oct '12

24 Oct '12

My colleague and I have been trying to test Kamailio configured as a Presence Server using OMA standards/messages. It appears that Kamailio does not support the OMA standards as it pertains to Presence. This is primarily in the area of the XCAP messages. Can someone please confirm if this is the case? If this is the case, it is on the roadmap to support it? If not, we'd like to know that too and we'll continue trying to get it to work. Thank you, Tom ______________________________________________________________________________________________________________________________ This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify so to the sender by e-mail and delete the original message. In such cases, please notify us immediately at sanchar-sadhan(a)infinite.com . Further, you are not to copy, disclose, or distribute this e-mail or its contents to any unauthorized person(s) .Any such actions are considered unlawful. This e-mail may contain viruses. Infinite has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachments. Infinite reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infinite e-mail system. ***INFINITE******** End of Disclaimer********INFINITE********

1 0

RFC5626 SIP Outbound Support of Flow-Timer Header
by Jonathan Hunter 24 Oct '12

24 Oct '12

Hi, Testing with RFC5626, can you confirm that the Flow-Timer header field is supported and configurable with Kamailio 3.3 ? Thanks Jon

2 3

sercmd compiles and installs as kamcmd for kamailio
by Daniel-Constantin Mierla 24 Oct '12

24 Oct '12

Hello, short note to let everyone know that sercmd tool compiles and installs as kamcmd for flavour kamailio as of today in the master git branch. There was a conflict when trying to install both flavours from packages, because of overlapping sercmd file. I added a basic manpage for the tool, a recommendation for each binary app in many linux distros. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Registering Phones
by Re Alvarez 24 Oct '12

24 Oct '12

Hi I am setting Kamailio as SBC in my organization using Freeswitch as PBX. The integration between the two systems is working perfectly and now I am trying to add some CISCO 7970 phones to the mix. Here's a summary of the setup 1. Kamailio has 2 NICs (1 public, 1 private), RTPproxy is running 2. Freeswitch is on private network. 3. All the CISCO phones are also on the private network. We will never have a SIP phone outside our network, My question, in this case. Should I register my phones to Kamailio or Freeswitch. I am new to Kamailio and still getting down with the SIP concepts. I would prefer to register the phones with Freeswitch and only want Kamailio to get involved when calls are leaving/entering the network. If two users, registered to the same freeswitch server call each other, Kamailio should not have to do anything (ideally, not sure if its possible with SBC setup). Thanks Re

1 0

getting siptrace error on start up
by Christopher Holland 23 Oct '12

23 Oct '12

getting this error: kamailio: ERROR: <core> [modparam.c:151]: set_mod_param_regex: parameter <trace_to_database> not found in module <siptrace> parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 63, column 46: Can't set module parameter I have set this to modparam("siptrace", "trace_to_database", "0") in my .cfg /I didn't compile siptrace when I compiled kamailio. I don't know how to add a module after compiling and installing kamailio. I haven't been able to find any documentaion on how to add a module after installing only compiling before installing. Thanks all for your help Christopher /

3 3

Kamailio Hosted Solutions
by Satyajeet Thokchom 23 Oct '12

23 Oct '12

I am looking for hosted Kamailio Solutions for my VOIP calling app in android and iphone. I have the app ready and have even setup kamailio server in our rackspace cloud but there is many problems like one way calling, noisy calls, no call connection etc. (application VOIP library , we are using IMSdroid and Idoubs , based on doubango SIP) I am looking for a hosted solutions affordable for me - with a support for 300 concurrent calls at a time. Best Regards, Satyajeet Thokchom Business Manager, NGBM Solutions Ltd Phone: India:+91-11-25541320 Mobile: 91 7428565264 Website: www.ngbm.net Web Support: www.ngbmsolutions.com Note: "This e-mail is confidential and may also be privileged. If you are not the intended recipient, please notify us immediately; you should not copy or use it for any purpose, nor disclose its contents to any other person"

1 0

turning off rev. dns
by Christopher Holland 22 Oct '12

22 Oct '12

I'm setting up Homer with kamailio for sip capturing. I've got it all set up and am getting this error. I have one interface for packet capture on a mirroired port and one interface for managment. I'm thinking that I have to only have kamailio listen on the packet capture interface and not the managment interface. Would that resolve this issue? loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(1832) WARNING: <core> [socket_info.c:1406]: WARNING: fix_hostname: could not rev. resolve 192.168.111.88 Listening on udp: 127.0.0.1:5060 udp: 192.168.111.88:5060 Aliases: udp: localhost:5060 config file ok, exiting... sense it's a local address it's not possible to do a rev. resolve of that address. Is there a way to turn off the rev. resolve. I've looked at this in the trouble shooting faq. How do I avoid Reverse-DNS-Lookups? A: Simply not use src_ip = "11.22.33.44" and use instead src_ip = 11.22.33.44! We had in our installation some serious trouble with requests coming from systems, where the hostname could not be reverse resolved (the DNS-Server responsible for this IP-Range was poorly configured and would not answer at all).

3 3

Remote Commands
by Ed Brady 22 Oct '12

22 Oct '12

Hi, I am trying to get a company who handles our billing and account provisioning connected to our Kamailio instance. I understand that XMLRPC is likely the best method for doing this, but I am unable to find direct commands equivalent to "kamctl add" or "kamctl rm". Is there a resource that lists the MI or XMLRPC commands, or do I need to create even simple commands? Thanks, Ed

2 4

Hangup call on leg B side
by pars3c 22 Oct '12

22 Oct '12

Hi , i don't know how to realize this: A --- INVITE---->proxy ----INVITE----> B <--- 200 OK ---BYE----> ---INVITE ----->C <--- 200 OK proxy <--- 200 OK -------ACK ---> -------ACK ---> When B answer the call, the proxy should immediately hang up the call and make a new invite. For user A is always the same call. I tried this : onreply_route { ################################ connect if (is_method("INVITE") && status == "200") { dlg_bye("callee"); drop(); } } But it doesn't work. When arrive 200 , the proxy drop the message to the user a , wait for a while, and then send a cancel to b (it should send a BYE, instead send a CANCEL, probably because it not have a ACK). How can I try this? Thanks

2 1

Kamailio 3.1.5 crashing during TLS connection setup
by Kamal Palei 22 Oct '12

22 Oct '12

Dear All I have setup Kamailio 3.1.5 with TLS enabled. Whenever I receive the REGISTER request from endpoint, trying to forward to next proxy. For that I have added below code in kamailio.cfg * if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:5061"); exit(); } *I see once REGISTER received at Kamailio proxy, it is sending TCP synch packet to next proxy, next proxy sends back a synch+reset packet. Then kamailio proxy is crashing with log as below. *root@B2BUA:/usr/local/src/scripts# 9(2347) ERROR: <core> [tcp_main.c:4139]: connect 115.114.48.75:5061 failed 9(2347) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting * The complete log is given as below. Please let me know how to get out of this issue. I used -m 64 option while starting proxy, but same result. Thanks, Kamal, NECS, Bangalore * 0(2336) INFO: <core> [tcp_main.c:4730]: init_tcp: using epoll_et io watch method (config) 0(2338) INFO: usrloc [hslot.c:53]: locks array size 512 0(2338) INFO: tls [tls_init.c:511]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1c 10 May 2012" (0x1000103f), kerberos support: off, compression: on 0(2338) INFO: tls [tls_init.c:519]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1c 10 May 2012" (0x1000103f), kerberos support: off, zlib compression: off compiler: gcc -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -DL_ENDIAN -DTERMIO -O3 -fomit-frame-pointer -Wall -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM 0(2338) INFO: tls [tls_init.c:373]: tls: init_tls: disabling compression... 0(2338) WARNING: tls [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 5242880 and 2621440 bytes 0(2338) INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 5242880 0(2338) INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2621440 0(2338) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2338) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 0(2338) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2338) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 0(2338) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 0(2338) INFO: tls [tls_domain.c:176]: TLSs<default>: tls_method=9 0(2338) INFO: tls [tls_domain.c:188]: TLSs<default>: certificate='/etc/certs/pocserver.com/cert.pem' 0(2338) INFO: tls [tls_domain.c:195]: TLSs<default>: ca_list='(null)' 0(2338) INFO: tls [tls_domain.c:202]: TLSs<default>: crl='(null)' 0(2338) INFO: tls [tls_domain.c:206]: TLSs<default>: require_certificate=0 0(2338) INFO: tls [tls_domain.c:213]: TLSs<default>: cipher_list='(null)' 0(2338) INFO: tls [tls_domain.c:220]: TLSs<default>: private_key='/etc/certs/pocserver.com/key.pem' 0(2338) INFO: tls [tls_domain.c:224]: TLSs<default>: verify_certificate=0 0(2338) INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 0(2338) INFO: tls [tls_domain.c:544]: TLSs<default>: No client certificate required and no checks performed 0(2338) INFO: tls [tls_domain.c:176]: TLSc<default>: tls_method=9 0(2338) INFO: tls [tls_domain.c:188]: TLSc<default>: certificate='(null)' 0(2338) INFO: tls [tls_domain.c:195]: TLSc<default>: ca_list='(null)' 0(2338) INFO: tls [tls_domain.c:202]: TLSc<default>: crl='(null)' 0(2338) INFO: tls [tls_domain.c:206]: TLSc<default>: require_certificate=1 0(2338) INFO: tls [tls_domain.c:213]: TLSc<default>: cipher_list='(null)' 0(2338) INFO: tls [tls_domain.c:220]: TLSc<default>: private_key='(null)' 0(2338) INFO: tls [tls_domain.c:224]: TLSc<default>: verify_certificate=1 0(2338) INFO: tls [tls_domain.c:227]: TLSc<default>: verify_depth=9 0(2338) INFO: tls [tls_domain.c:529]: TLSc<default>: Server MUST present valid certificate 2(2340) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 1(2339) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 4(2342) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 0(2338) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 6(2344) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 3(2341) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 7(2345) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 6(2344) INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_et io watch method (config) root@B2BUA:/usr/local/src/scripts# 9(2347) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 8(2346) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled root@B2BUA:/usr/local/src/scripts# 9(2347) ERROR: <core> [tcp_main.c:4139]: connect 115.114.48.75:5061 failed 9(2347) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting 0(2338) ALERT: <core> [main.c:742]: child process 2347 exited by a signal 6 *

2 2

Listen on all IPv6 addresses
by Anton Kvashenkin 22 Oct '12

22 Oct '12

Hello guys. In the Kamailio v.3.3.0 release notes http://www.kamailio.org/w/kamailio-v3-3-0-release-notes/ it says: all IPv6 network interfaces are auto-detected and Kamailio start listening on them (if no strict listening rules are set) But it doesn't happen. version: kamailio 3.3.1 (x86_64/linux) d75cb5 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d75cb5 listen is commented out @ cfg

2 1

Re: [SR-Users] rtpproxy delete fails
by Juha Heinanen 22 Oct '12

22 Oct '12

the patch below adds 't' flag to rtpproxy_destroy flags. if 't' flag is present when rtpproxy_destroy is called, to tag is not included in D command sent to rtpproxy thus causing destroy of the full call. 't' flag can be used to avoid piling up of unused rtpproxy calls by making it possible to call rtpproxy_destroy when 200 ok is received from a branch where rtpproxy is not used. is it ok for me to commit this patch or do long term rtpproxy users have a better idea? it would still make sense to include a timer in rtpproxy that cleans up un-used calls at an interval that corresponds to sip proxy's invite timeout. -- juha *** /usr/src/orig/sip-router/modules/rtpproxy/rtpproxy.c 2012-10-19 14:31:27.658485534 +0300 --- modules/rtpproxy/rtpproxy.c 2012-10-20 09:52:10.308389743 +0300 *************** *** 1678,1683 **** --- 1678,1684 ---- str callid, from_tag, to_tag, viabranch; char *cp; int via = 0; + int to = 1; int ret; struct rtpp_node *node; struct iovec v[1 + 4 + 3 + 2] = {{NULL, 0}, {"D", 1}, {" ", 1}, {NULL, 0}, {NULL, 0}, {NULL, 0}, {" ", 1}, {NULL, 0}, {" ", 1}, {NULL, 0}}; *************** *** 1700,1705 **** --- 1701,1710 ---- via = 1; break; + case 't': + case 'T': + to = 0; + break; case 'a': case 'A': case 'i': *************** *** 1736,1742 **** return -1; } to_tag.s = 0; ! if (get_to_tag(msg, &to_tag) == -1) { LM_ERR("can't get To tag\n"); return -1; } --- 1741,1748 ---- return -1; } to_tag.s = 0; ! to_tag.len = 0; ! if ((to == 1) && get_to_tag(msg, &to_tag) == -1) { LM_ERR("can't get To tag\n"); return -1; }

2 1

mediaproxy-ng fails to parse start_recording() command
by Juha Heinanen 22 Oct '12

22 Oct '12

i tried what happens when i call start_recording() after calling rtpproxy_manage() and got this kind of error to syslog: Oct 20 19:57:43 siika mediaproxy-ng[8423]: Failed to properly parse UDP command line '8554_13 R ngsgznnwgzpuovt(a)siika.tutpro.com mgpwe' from 127.0.0.1:51497, using fallback RE why the error? it is so that mediaproxy-ng does not support recording? -- juha

4 3

Re: [SR-Users] dangers of excluding 100rel?
by Alex Balashov 19 Oct '12

19 Oct '12

You can strip 100rel as long as it's Supported. However, not so good things happen if you strip it from Required. :-) -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Andrew Mortensen <admorten(a)isc.upenn.edu> wrote:Can I safely strip 100rel support from INVITEs sent by callers registered to my proxy? We're using proxy-generated 302 redirects for call forwarding in our deployment. In general this works better for us than appending branches, as we rapidly ran into branch limits when dealing with large multiple appearance lines. However, we've run into a case where the caller, a Polycom 550, seems to get confused when a parallel forked call is redirected, 100rel is in use and the redirection target is also a multiple appearance line. The caller is correctly responding with PRACKs to all 180s, before and after the redirection. However, hanging up the call on the caller after the redirection, but before any remote party has answered, does *not* cause the caller to send a CANCEL to tear down the call. In fact, the caller doesn't send anything at all. The remote parties continue to ring until the ring timer expires, and the proxy sends CANCELs to all callee endpoints. That's clearly a Polycom bug, and we'll need to work with them to get it resolved. In the meantime, I'm looking for workarounds. One quick, crude option is to strip 100rel from the Supported header in the caller's INVITE. I've tested this, and it does solve the problem, but I'm concerned about side effects. I'm aware of the value 100rel has for calls to the PSTN, but in this case the calls never leave the local network. Are there situations where 100rel is critical for a parallel forked call? andrew _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

dangers of excluding 100rel?
by Andrew Mortensen 19 Oct '12

19 Oct '12

Can I safely strip 100rel support from INVITEs sent by callers registered to my proxy? We're using proxy-generated 302 redirects for call forwarding in our deployment. In general this works better for us than appending branches, as we rapidly ran into branch limits when dealing with large multiple appearance lines. However, we've run into a case where the caller, a Polycom 550, seems to get confused when a parallel forked call is redirected, 100rel is in use and the redirection target is also a multiple appearance line. The caller is correctly responding with PRACKs to all 180s, before and after the redirection. However, hanging up the call on the caller after the redirection, but before any remote party has answered, does *not* cause the caller to send a CANCEL to tear down the call. In fact, the caller doesn't send anything at all. The remote parties continue to ring until the ring timer expires, and the proxy sends CANCELs to all callee endpoints. That's clearly a Polycom bug, and we'll need to work with them to get it resolved. In the meantime, I'm looking for workarounds. One quick, crude option is to strip 100rel from the Supported header in the caller's INVITE. I've tested this, and it does solve the problem, but I'm concerned about side effects. I'm aware of the value 100rel has for calls to the PSTN, but in this case the calls never leave the local network. Are there situations where 100rel is critical for a parallel forked call? andrew

1 0

rtpproxy_destroy([flags]) does not accept flags
by Juha Heinanen 19 Oct '12

19 Oct '12

according to rtpproxy module readme, rtpproxy_destroy function takes optional flags param: 5.4. rtpproxy_destroy([flags]) however, the code does not support it: {"unforce_rtp_proxy", (cmd_function)unforce_rtp_proxy_f, 0, 0, 0, ANY_ROUTE}, {"rtpproxy_destroy", (cmd_function)unforce_rtp_proxy_f, 0, 0, 0, ANY_ROUTE}, what has happened? -- juha

2 3

MSILO Configuration
by Ed Brady 19 Oct '12

19 Oct '12

Hi, I am having trouble configuring MSILO. I copied the example config into my kamailio.cfg file and edited the areas that apply to my domain, but when restarting Kamailio I am faced with an error. Below is the error followed by my complete config file. Please advise what I am doing wrong. Is there some sort of resource for HowTos on Kamailio? The module documentation seems very thorough but the examples at the end seem to only be useful if that module is the only one you need to configure. I am trying to do a fairly straight forward configuration I think (SIP messaging with offline message storing and a single incoming/outgoing gateway that is used for all accounts that do not locally exist), but I am unable to find straight forward documentation on how to configure something like this. Please let me know what I am missing. Thanks, Ed Not starting Kamailio: invalid configuration file! > -e > 0(30863) INFO: tls [tls_init.c:377]: tls: init_tls: disabling > compression... > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: syntax error > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: bad command > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: bad command > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 878, column 12: syntax error > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 878, column 12: > ERROR: bad config file (5 errors) > *Config* #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_TLS #!define WITH_NAT #!define WITH_ACCDB # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:R4U8oO8Er872sNHr@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "cfgutils.so" loadmodule "msrp.so" loadmodule "htable.so" loadmodule "msilo.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # -- msilo params -- modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@MYDOMAIN.com") modparam("msilo","contact_hdr","Contact: registrar(a)MYDOMAIN.com :5060;msilo=yes\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","*** User $rU is offline!") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=1800;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # Route to MSILO 10-18-12 route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay(); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } event_route[msrp:frame-in] { xdbg("============#[[$msrp(method)]]===========\n"); xdbg("============*[[$si:$sp]]\n"); xdbg("============ crthop: [$msrp(crthop)]\n"); xdbg("============ prevhop: [$msrp(prevhop)]\n"); xdbg("============ nexthop: [$msrp(nexthop)]\n"); xdbg("============ firsthop: [$msrp(firsthop)]\n"); xdbg("============ lasthop: [$msrp(lasthop)]\n"); xdbg("============ prevhops: [$msrp(prevhops)]\n"); xdbg("============ nexthops: [$msrp(nexthops)]\n"); xdbg("============ srcaddr: [$msrp(srcaddr)]\n"); xdbg("============ srcsock: [$msrp(srcsock)]\n"); xdbg("============ sessid: [$msrp(sessid)]\n"); msrp_reply_flags("1"); if(msrp_is_reply()) { msrp_relay(); exit; } # handle AUTH MSRP requests if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } # frame for local server - send Use-Path # -- passwd can be loaded from DB based on $au $var(passwd) = "xyz123"; if(!pv_www_authenticate("myrealm", "$var(passwd)", "0")) { if(auth_get_www_authenticate("myrealm", "0", "$var(wauth)")) { msrp_reply("401", "Authorization Required", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } $var(cnt) = $var(cnt) + 1; pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)"); $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr); $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock); # - Use-Path: the MSRP address for server + session id $var(UsePath) = "Use-Path: msrp://127.0.0.1:5060/" + $var(sessid) + ";tcp\r\n"; msrp_reply("200", "OK", "$var(UsePath)"); exit; } if($msrp(method)=="SEND") { if($msrp(nexthops)>1) { msrp_reply("200", "Received"); msrp_relay(); exit; } $var(sessid) = $msrp(sessid); if($sht(msrp=>$var(sessid)::srcaddr) == $null) { # one more hop, but we don't have address in htable msrp_reply("481", "No Such Session"); exit; } msrp_relay_flags("1"); msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)", "$sht(msrp=>$var(sessid)::srcsock)"); msrp_relay(); exit; } msrp_relay(); }

3 2

Re: [SR-Users] Kamailio v3.3.2 Released
by Paul Cupis 18 Oct '12

18 Oct '12

On 16/10/12 22:29, Jon Bonilla (Manwe) wrote: > El Tue, 16 Oct 2012 17:00:03 +0200 > Daniel-Constantin Mierla<miconda(a)gmail.com> escribió: >> >> Kamailio SIP Server v3.3.2 stable release is out. > > Debian and Ubuntu packages are ready. Please find the instructions > below: > > http://www.kamailio.org/dokuwiki/doku.php/packages:debs I'm seeing a problem with the 3.3.2 packages on Debian at the moment. There is a file conflict between the kamailio package and kamailio-mysql-modules preventing the latter from installing. $ dpkg-deb -c /var/cache/apt/archives/kamailio_3.3.2+squeeze1_amd64.deb | grep srdb -rw-r--r-- root/root 74296 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1.0 -rw-r--r-- root/root 28360 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1 -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1 -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so -> libsrdb2.so.1.0 $ dpkg-deb -c /var/cache/apt/archives/kamailio-mysql-modules_3.3.2+squeeze1_amd64.deb | grep srdb -rw-r--r-- root/root 28360 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1.0 -rw-r--r-- root/root 74296 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1 -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1 -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so -> libsrdb1.so.1.0 Regards,

2 3

Kamailio- Redirect message 300 with LCR reroute
by Fatima Chahrour~Vanrise Support 18 Oct '12

18 Oct '12

Dear List, I'd like to get help regarding my case. I have the following script where many thanks to Daniel has helped me in if (is_method("INVITE")) { if (!load_gws(1, $rU, $fu)) { sl_send_reply("502", "Unable To lOad GatEwAyS"); exit;} if(!next_gw()){ sl_send_reply("503", "Unable To fInD a gateWaY"); exit;} while(next_gw()){ km_append_branch(); } sl_send_reply("302","Moved Temporary"); exit; } The problem am facing is that the call is rerouting from the first gateway to the next gateway successfully when I have two gateways. BUT when I have three gateways, and the first two gateways are off, the call is not rerouted to the third gateway. It keeps hitting the seconds gateway and gives request time out at the end Why is that? Thanks in advance, F Chahrour

1 0

Kamailio v3.3.2 RPM Packages
by Daniel-Constantin Mierla 18 Oct '12

18 Oct '12

Hello, for the record, RPM packages of Kamailio v3.3.2 were built for Centos 5.5, Centos 6, RedHat 5, RedHat 6, Fedora 15, Fedora 16, Fedora 17, OpenSuse 11.4 and OpenSuse 12.1, on i586 and x86_64 architectures. They are available on opensuse build service repos, see the links at: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Question about compliance with RCS/OMA Simple Presence
by Sangeeta Shah 17 Oct '12

17 Oct '12

Hello All, I need some guidance on whether the Kamailio presence/presence_xml and integrated xcap module implementations are RCS/OMA Simple presence compliant. Based on a few things I've seen in the code I don't see any handling for the oma presence rule xml extensions. Is there any plan to be RCS/OMA compliant in the new future? I can get SIMPLE PRESENCE with XCAP to work but not with OMA presence rules and resource lists. Thanks, Sangeeta -- Sangeeta Shah

1 0

has_sdp()??
by Juha Heinanen 17 Oct '12

17 Oct '12

while reading rtpproxy module readme, i found this kind of example: if (is_method("INVITE")) { if (has_sdp()) { if (rtpproxy_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } if (is_method("ACK") && has_sdp()) rtpproxy_answer(); does has_sdp() function really exist? it is not listed on wiki page: http://www.kamailhttp://www.kamailio.org/wiki/alphaindexes/3.4.x/modfunctio… -- juha

4 13

make option to clean the installed binaries
by Ajay Sabat 17 Oct '12

17 Oct '12

Hi, I am a newbie here and recently started using kamailio/ser. I am wondering is there a makefile option to cleanup the installed binaries and library files. Thanks, Ajay

2 3

how to search previous messages in the forum
by Ajay Sabat 17 Oct '12

17 Oct '12

Hi, Many times I get doubts and questions about Kamailio that I want to search in the SR-Users forum before posting a new message, as it might have already been answered in the past. Is there a easier way to do that than looking up the subject lines in the monthly archive? Thanks, Ajay

2 1

Errors issued by Kamailio
by Sangeeta Shah 17 Oct '12

17 Oct '12

Hi, I am seeing the following errors in syslog: Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence_xml [notify_body.c:635]: while extracting tuple node Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence_xml [notify_body.c:75]: while constructing offline body Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence [presentity.c:1220]: preparing body Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence [publish.c:171]: Marking presentity Oct 8 16:21:55 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event message-summary Oct 8 16:21:55 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event presence.winfo Oct 8 16:21:57 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event presence I am using Kamailio 3.3 with Presence and RLS with integrated XCAP. The only significant change i made today is update the presence and RLS config to DB ONLY mode. Thanks, Sangeeta -- Sangeeta Shah

2 9

$fs syntax
by Juha Heinanen 17 Oct '12

17 Oct '12

wiki tells: $fs - reference to the forced socket for message sending (if any) in the form proto:ip:port should it be [proto:]ip:port instead? -- juha

3 7

Kamailio 3.2.4 presence
by Vitaliy Aleksandrov 17 Oct '12

17 Oct '12

Hi all, I'm facing problems with active subscription restoration after sipproxy restart. db_mode is 0 (DB_MEMORY_ONLY) and according to comments in the presence.h kamailio holds subscription in memory and periodically updates to db, but retrieves from db only at startup. And it's true, I see that kamailio tries to update the active_watchers table, but it never inserts new subscriptions there. There are no problems with access to the database since when db_mode is 2 (DB_ONLY) kamailio inserts new records into the active_watchers without any problems. Here are presence module settings: modparam("presence", "db_url", "sqlite:////var/db/presence.db") modparam("presence", "db_mode", 0) modparam("presence", "timeout_rm_subs", 1) modparam("presence", "pres_htable_size", 8) modparam("presence", "subs_htable_size", 8) modparam("presence", "max_expires", 3600) modparam("presence", "db_update_period", 60) modparam("presence", "clean_period", 60) Does anybody have success with db_mode = DB_MEMORY_ONLY and active_watchers db table ?

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