sr-users April 2012

sr-users@lists.kamailio.org

75 participants
133 discussions

Stateless Proxy
by Akan 20 Apr '12

20 Apr '12

Hello, I am trying to setup a stateless proxy where certain requests are forwarded to another sip server that will handle those requests specifically. I have the request message forwarding to the other sip server but the reply is being sent back to the original sip server then to the client. When I look at the SIP request and reply, I see a new "Via" header line with the IP address of the original sip proxy server. What I am trying to setup is a stateless proxy which just forwards the original request to another sip server and then this second sip server will handle everything from there. Basically offloading the traffic to other servers. Currently i am rewriting the ruri host and port as done in the sample config and issuing the forward() function. I have tried t_relay() as also in the sample config but I get the same results. Is there something else that I need to do before forwarding to the other sip server? How can I get rid of the new "Via" line? I believe this is from record_route() but I am not issue this before I perform the relay. I have Kamailio 3.2.3 loaded and used the sample config as my base config. Thanks Nathaniel

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How does RLS work with kamailio.
by Gnaneshwar Gatla 19 Apr '12

19 Apr '12

Hello, I'm trying to implement RLS with integrated XCAP server. I have successfully run the Kamailio instance with the following snippet. Since I had no RLS clients available, I have tried to test this with SIPp. The attached pcap trace is the result. Test: 1. Send an XCAP signal over http to Kamailio. 2. Register and subscribe Kamailio. The test does store the xml received over http in the xcap table in the database. But rls module does not store the data neither in rls_presentity or rls_watcher. When I sent the Subscribe with sipp, the server throws error that subscription was not found in the database. I would like to know if I'm doing something wrong to implement this service. Is there any brief tutorial on how to implement this? Kamailio.cfg snippet: ##Presence modparam("presence", "server_address", PRESENCE_IP) modparam("presence", "clean_period", 65) modparam("presence", "db_update_period", 60) modparam("presence", "max_expires", 75) modparam("presence", "fallback2db", 1) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) ##rls modparams modparam("rls", "db_url", DBURL) modparam("rls", "waitn_time", 10) modparam("rls", "integrated_xcap_server", 1) modparam("rls", "max_notify_body_length", 32000) modparam("rls", "to_presence_code", 10) modparam("rls", "server_address", RLS_URI) modparam("rls", "outbound_proxy", PUA_OUTBOUND_PROXY) modparam("rls", "xcap_root", XCAP_ROOT) ##XCAP modparams modparam("xcap_server", "db_url", DBURL) modparam("xcap_client", "db_url", DBURL) modparam("xcap_client", "query_period", 50) 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")) { $var(ret_code)= rls_handle_subscribe(); xlog("Subscribe: $var(ret_code)"); if($var(ret_code)== 10) 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; } In the [xhttp:request]: switch($rm) { case "PUT": xcaps_put("$var(uri)", "$hu", "$rb"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } else if ($xcapuri(u=>auid)=~"resource-lists" || $xcapuri(u=>auid)=~"rls-services") { xlog("==== xhttp put: refreshing rls watchers $var(uri)\n"); rls_update_subs("$var(uri)", "presence"); } exit; break; Gnaneshwar Gatla | InTouch Health | Software Developer 6330 Hollister Ave. Goleta CA, 93117 | P: 805.562.8686 ext: 199

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Kamailio v3.2.3 Released
by Daniel-Constantin Mierla 19 Apr '12

19 Apr '12

Hello, Kamailio SIP Server v3.2.3 stable release is out. This is a maintenance release of the latest stable branch, 3.2, that includes fixes since release of v3.2.2. There is no change to database schema or configuration language structure that you have to do on installations of v3.2.0, v3.2.1 or v3.2.2. Deployments running previous v3.2.x versions are strongly recommended to be upgraded to v3.2.3. For more details about version 3.2.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/04/kamailio-v3-2-3-released/ RPMs and Debian/Ubuntu packages will be available soon. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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Distributed subscribers Kamailio
by Juanjo 19 Apr '12

19 Apr '12

Hello everybody I have a question about clustering kamailio, i want to register subscribers to two diferent kamailios, and the subscribers from one kamailio should be able to communicate with the subscribers registered in the other kamailio How can i do it? Is there something like DUNDI in the Asterisk world? Thank you in advance.

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uac module incoming INVITE become wrong dst_ip address
by Karsten Horsmann 19 Apr '12

19 Apr '12

Hello List, i am nearly finishing my multihomed config (public ip/internal ip) with kamailio. For incoming calls i want to use the uac module too. Then i want to locate an permanet entry and dispatch it to ivr system behind kamailio. i tried something like this: if (lookup("location") && (dst_ip==222.222.222.222)) { if(ds_select_domain("1", "4")) { Okay for registerd UAC this works perfect - but if made an call to the uac registerd number i got as $Ri the 172.20.100.74 - and thats is the private ip address. Why that??? Any hints to check "if its internal or external" call??? -- Mit freundlichen Grüßen *Karsten Horsmann*

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sqlite -> usrloc tables deleted after 1 minute
by Pedro Antonio Vico Solano 19 Apr '12

19 Apr '12

2 6

Releasing v3.2.3
by Daniel-Constantin Mierla 19 Apr '12

19 Apr '12

Hello, last patch release (v3.2.2) was done quite some time ago, therefore I am thinking of releasing v3.2.3 sometime next week, maybe Wednesday or Thursday. If there are patches in master branch to be backported to 3.2, please start doing them. Also, if anyone is aware of issues in 3.2 branch not reported to the tracker, announce them asap. Thanks, Daniel -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

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Stress test
by Lucas Alvarez 18 Apr '12

18 Apr '12

Can someone recommend a good stress tool for testing kamailio? Or can point through the right path? Thanks in advance. Lucas

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implementing Kamailio with FreeSwitch.
by Mehdi Badiezadegan 18 Apr '12

18 Apr '12

Hello and thanks for your helping,excuse me I'm newbie in mailing list. I have a question about implementing Kamailio with FreeSwitch. Our plan is a configuration like this at the follow: 1.Kamailio as registering and LoadBalancer ,response to chat and call(voice/video) in p2p 2.we have 2 FreeSwitch media engine just for Conference case(Voice/Video) . How we can implementing above kamailio as loadBalancer for 2 FreeSwitchs in conference case , And at the same time,It answer as voip/chat server for p2p case ? we have success by implementing Kamailio with one FreeSwitch, (FreeSwitch as media Engine just for conference and Kamailio for p2p call) But if we have 2 FreeSwitch with same activity,How we can implementing... And So I had Efforts with dispatcher module, but it was good only for route/Loadbalancer not for answering p2p call. Thanks a lot Mehdi Badiezadegan

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loose_route() for ACK message
by Laurent Schweizer 18 Apr '12

18 Apr '12

Hello all, I have a script working with openser1.2.1 and I try to move them to kamailio 3.2.1 I have actually an issue with the loose_route() function at least for ACK message. When I get the ACK the loose_route function is changing the URI with the value of the route header of the ACK message . The problem is that this value is the local proxy himself so the t_relay() function send the ACK on the loopback interface. Any Idea ? Regards Laurent Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: START REQ:[ACK] Tran:[7070] URI:[sip:peoplefone@95.128.80.93:7070] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: SANITY OK, REQ:[ACK] Tran:[7070] URI:[sip:peoplefone@95.128.80.93:7070] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: DIALOG1 REQ:[ACK] Tran:[7070] URI:[sip:peoplefone@95.128.80.93:7070] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] If (loose_route()) is called Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: DIALOG1ACK REQ:[ACK] Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: NATMANAGE REQ:[ACK] Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: DIAL_RELAY1 REQ:[ACK] Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] Apr 16 18:48:08 proxy kamailio[28033]: INFO: <script>: T_RELAY REQ:[ACK] Tran:[5060] URI:[sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2] from:[sip:41763770377@sip.call.lt;user=phone] to:[sip:37052077800@81.16.238.195;user=phone] CallID:[25694-UI-05aa645a-7000c6ef6@sip.call.lt] U 81.16.238.195:5060 -> 87.110.233.1:5060 SIP/2.0 200 OK. To: <sip:37052077800@81.16.238.195;user=phone>;tag=e868748c. Via: SIP/2.0/UDP 87.110.233.1:5060;rport=5060;branch=z9hG4bK-26E6-2AB3E9F. Record-Route: <sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2>. CSeq: 79950814 INVITE. Call-ID: 25694-UI-05aa645a-7000c6ef6(a)sip.call.lt. From: <sip:41763770377@sip.call.lt;user=phone>;tag=25694-BR-05aa645b-0f448aeb2. Content-Type: application/sdp. Contact: <sip:peoplefone@95.128.80.93:7070>. Content-Length: 205. . v=0. o=90706832381 5012 11 IN IP4 81.16.238.195. s=Mapping. c=IN IP4 81.16.238.195. t=0 0. m=audio 8170 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=sendrecv. U 87.110.233.1:5060 -> 81.16.238.195:5060 ACK sip:peoplefone@95.128.80.93:7070 SIP/2.0. Call-ID: 25694-UI-05aa645a-7000c6ef6(a)sip.call.lt. Contact: <sip:87.110.233.1:5060>. CSeq: 79950814 ACK. From: <sip:41763770377@sip.call.lt;user=phone>;tag=25694-BR-05aa645b-0f448aeb2. Max-Forwards: 30. Route: <sip:81.16.238.195;lr;ftag=25694-BR-05aa645b-0f448aeb2>. To: <sip:37052077800@81.16.238.195;user=phone>;tag=e868748c. User-Agent: Cirpack/v4.42o (gw_sip). Via: SIP/2.0/UDP 87.110.233.1:5060;branch=z9hG4bK-388F-2AB3F03. Content-Length: 0. .

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sr-users April 2012