sr-users May 2012

sr-users@lists.kamailio.org

84 participants
140 discussions

Routing based on gruu
by Krishna Kurapati 03 May '12

03 May '12

Hi, I am trying to use GRUU for routing MESSAGEs. For example if I want to route to specific instance of the registered user, what do I need to send in the MESSAGE? If I know pub-gruu of the instance, where do I insert it so that kamailio can correctly process it and send it to the right instance? Thanks Krish Kura

2 1

kamctl trouble
by Vineet Menon 03 May '12

03 May '12

Hi, I have a kamailio installation running, but in kamctl the moni command works, but stop, restart, start command doesn't work... why is it so?? Regards, Vineet Menon

3 3

AVP assignment question
by Klaus Darilion 03 May '12

03 May '12

Hi! For example: $avp(i:test) = 1; $avp(i:test) = 2; # now: test[0]=2, test[1]=1 I would think that assigning or deleting using an index would overwrite the respective element in the AVP list, but assigning $null always deletes [0] and assigning a value always add another AVP to the top, e.g: $(avp(i:test)[1]) = 3; # I would expect: test[0]=2, test[1]=3 # but it is: test[0]=3, test[1]=2, test[2]=1 $(avp(i:test)[2]) = $null; # I would expect: test[0]=3, test[1]=2 # but it is: test[0]=2, test[1]=1 So, is the current behavior really the correct behavior? It is not what I would expect when using indexes. If it is correct, I will improve the documentation accordingly. regards Klaus

2 2

dialog - keep alives with lower cseq
by Daniel-Constantin Mierla 03 May '12

03 May '12

Hello, several days ago, just before freezing the development for v3.3.0, I added to dialog module the option to send keep alives for ongoing calls in order to detect if caller/callee is gone. SIP specs require to increment the CSeq for requests within dialog, but because the keep alives are sent from the sip server, it would results de-synchronization of the CSeq values hold in phones themselves (e.g., a BYE created by caller/callee after a keep alive will be with lower cseq than the other side would expect and accept). One solution would be to update cseq when BYE passes the server to the right value. This implies more processing, as a call can have many re-INVITEs or other requests within dialog sent by caller/callee, including periodic updates to database to store cseq numbers. So I went for a different approach, like stated in subject -- let's see your opinion if you think is going to work. Practically the keep alives will be OPTIONS with CSeq equal or lower than the last valid value (e.g., the cseq of the INVITE creating the call). If the caller/callee is gone, that is simply, the OPTION will be timed out and dialog module will send BYE. If the caller/callee are reachable, but for some reason the call was destroyed (e.g., a crash and restart meanwhile), since there is a To-tag, the OPTIONS should get a 481 Call/Leg transaction does not exists. Again, a case when kamailio will end the dialog. The one to be discussed here would be caller/callee are still on the call. Based on RFC (and the feedback from people on another thread here), a requests coming within a dialog with lower CSEq should be replies with 500. CSeq numbers are still ok in both sides (note that requests with lower CSeq can happen in reality, like two fast re-INVITEs sent over UDP, the second arriving first due to network transmission). I tested with snom phones and jitsi so far, seems to be a working solution (well, jitsi was replying 500 after 20secs to keep alive OPTIONS request, not sure for what reason, just reported back to the project). Is anyone here seeing any possible issues with the approach? Call are ended by dialog, with proper CSeq in the BYE, after 10seconds from the moment 408 or 481 is received, no other replies are taken in consideration for ending the dialog from server side. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 5

Can't load perl module
by Ladislav Jurák 02 May '12

02 May '12

Hello, I have problem with loading perl module on Kamailio v3.2 installed from repository on Debian sqeeze amd64. Perl modules was also installed from repository package kamailio-perl-modules. I have also installed all perl dependences listed in perl module documentary. Into Kamailio config file I added: loadmodule "perl.so" and modparam("perl", "filename", "/etc/kamailio/skript.pl") Now when I'm trying to start Kamailio I'll get following: May 1 23:13:46 server kamailio: DEBUG: <core> [cfg.y:1712]: loading module perl.so May 1 23:13:46 server kamailio: DEBUG: <core> [sr_module.c:557]: load_module: trying to load </usr/lib64/kamailio/modules_k/perl.so> May 1 23:13:46 server kamailio: WARNING: <core> [sr_module.c:620]: /usr/lib64/kamailio/modules_k/perl.so: exports dlflags interface is deprecated and it will not be supported in newer versions; consider using mod_register() instead May 1 23:13:46 server kamailio: DEBUG: <core> [modparam.c:96]: set_mod_param_regex: 'perl' matches module 'perl' May 1 23:13:46 server kamailio: DEBUG: <core> [sr_module.c:763]: find_param_export: found <filename> in module perl [/usr/lib64/kamailio/modules_k/perl.so] May 1 23:13:46 server kamailio: DEBUG: <core> [modparam.c:113]: set_mod_param_regex: found <filename> in module perl [/usr/lib64/kamailio/modules_k/perl.so] May 1 23:13:46 server /usr/sbin/kamailio[31192]: DEBUG: <core> [sr_module.c:928]: DEBUG: init_mod: perl May 1 23:13:46 server /usr/sbin/kamailio[31192]: ERROR: perl [perl.c:238]: failed to load perl file "/etc/kamailio/skript.pl". May 1 23:13:46 server /usr/sbin/kamailio[31192]: ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module perl (/usr/lib64/kamailio/modules_k/perl.so) I am confused about the WARNIG too, can it have some influence to the errors? File skript.pl is in the specified directory. I also changed permissions to 777. It's content is from sample "branches.pl". I have similar problem with module app_python.so. Can you please give me some advice? It's important for me to get this module working. Thank you. -- Regards Ladislav Jurak

2 6

sdpops Siremis
by Grégoire Vandendeurpel 02 May '12

02 May '12

Hi, I'm using Siremis 3.2 ans Kamailio 3.2.3 for my final project of my bachloor degree I would like to know how Siremis and Kamailio work together? Like: If I decide to set a protocol in the acl trusted in Siremis like TLS, ok, it works well !! But I need to know the different links between siremis and kamailio to working well.. Why? Cause I need to implement in Siremis a choise form about the sdpops module, and it has to work well of course with the database of ser! So.. What are the different steps to do this? - Create new table sdpops in ser - create the form in siremis and view .. - what in kamailio.cfg?? I found this: http://kb.asipto.com/siremis:install32x:new-views OK, but in my case, I need to do that for SDPOPS, which is a module of kamailio, whithout any table and components about! I mean, there is no openser.sdpops in the database of kamailio, so create the table ok.. but what are the columns, datas, to set in ?? cause I tried php gen_meta.php Serdb sdpops ser.acl.sdpops of course it does not work --> no openser.sdpops ...... thx in advance for your help Best regards -- *Grégoire Vandendeurpel, * * * *IT Sector*

1 0

Problem with Route header
by Geoffrey Mina 02 May '12

02 May '12

Greetings, I am confused at some functionality I am seeing with Kamailio 1.5.4. I know this is an old version, but I don't have the time to go through a lengthy upgrade process right now. The issue I am seeing is that the server is inserting a Route header with it's own IP address for an unknown reason. Here is the initial invite (removed SDP for simplicity): INVITE sip:13@67.207.130.146:5060 SIP/2.0 Via: SIP/2.0/UDP 68.64.220.108:5060;branch=z9hG4bK78dd33c6;rport From: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;tag=as1cad6370 To: <sip:13@67.207.130.146:5060> Contact: <sip:9546496707@68.64.220.108> Call-ID: 43134ece101abfca6ecab20212295909(a)dev-asterisk.mydomain.com CSeq: 102 INVITE User-Agent: G-Tel v1.0 Max-Forwards: 70 Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;privacy=off;screen=no Date: Tue, 01 May 2012 18:17:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Route: <sip:13@boulder-voip.mydomain.com> P-Account-ID: 99990023 P-Proxy-Route: Yes Content-Type: application/sdp Content-Length: 240 The basics of what happen next are: t_check_trans(); record_route(); remove_hf("P-Proxy-Route"); if(loose_route()){ route(3); } route[3]{ t_on_reply("1"); if(!t_relay()){ sl_reply_error(); } } The INVITE that goes out has the funky Route: header with the Kamailio IP in there. This is causing problems for some of the upstream proxy servers (obviously). INVITE sip:13@boulder-voip.mydomain.com SIP/2.0 Record-Route: <sip:67.207.130.146;lr;ftag=as1cad6370> Via: SIP/2.0/UDP 67.207.130.146;branch=z9hG4bKf183.456d51e1.0 Via: SIP/2.0/UDP 68.64.220.108:5060 ;received=68.64.220.108;branch=z9hG4bK78dd33c6;rport=5060 From: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;tag=as1cad6370 To: <sip:13@67.207.130.146:5060> Contact: <sip:9546496707@68.64.220.108> Call-ID: 43134ece101abfca6ecab20212295909(a)dev-asterisk.mydomain.com CSeq: 102 INVITE User-Agent: G-Tel v1.0 Max-Forwards: 69 Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com >;privacy=off;screen=no Date: Tue, 01 May 2012 18:17:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces P-Account-ID: 99990023 Content-Type: application/sdp Content-Length: 240 Route: <sip:13@67.207.130.146:5060> Any idea what may be causing this to happen and how I could prevent it? I have tried removing the Route header using the remove_hf("Route") before doing the t_relay, but that doesn't seem to help. Thanks, Geoff

2 2

kamailio and openser
by Vineet Menon 01 May '12

01 May '12

Hi, I have a ubuntu 12.04 installation with me, and want to install kamailio in it... I have added repo from kamailio.ord site, but they are pretty old and require libmysqlclient16, which is no longer available. Ubunut 12.04 has a canonical repo for openser.... are they both interchangeble?? can i install openser and feel the same as kamailio?? Regards, Vineet Menon

3 3

LCR module from_uri module how to ?
by Sébastien Cramatte 01 May '12

01 May '12

Hello, Does anyone can give me a proper example how setup LCR to be able to use prefix and from_uri field. Does not a route should be preferred if prefix or from_uri is more accurate ? I mean that if you st from_uri value instead of default NULL route should be considered as better ? Thank you for you help

2 1

OpenXcap always returns full resource list
by Fabian Bernhard 01 May '12

01 May '12

Dear list, we are running Kamailio 3.2 together with the included OpenXcap server. I try to retrieve only part of a certain resource list using XPath, but the server always returns the full list, see example below. I was so far not able to retrieve only part of a resource list, independent of the XPath expression I'm using. Do I make something wrong or is this a bug? Thanks and regards, Fabian fubeh@orion:~$ curl --digest -u test4:XXXX http://192.168.1.134:5060/xcap-root/resource-lists/users/sip:test4@192.168.… <?xml version="1.0"?> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="Business"> <entry uri="sip:test1@192.168.1.134"> <display-name>Test 1</display-name> </entry> <entry uri="sip:test122@192.168.1.134"> <display-name>Test 1</display-name> </entry> </list> <list name="Family"> <entry uri="sip:test2@192.168.1.134"> <display-name>Test 2</display-name> </entry> <entry uri="sip:test222@192.168.1.134"> <display-name>Test 2</display-name> </entry> </list> <list name="Friends"> <entry uri="sip:test3@192.168.1.134"> <display-name>Test 3</display-name> </entry> <entry uri="sip:test322@192.168.1.134"> <display-name>Test 3</display-name> </entry> </list> <list name="Blocked" /> </resource-lists>

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sr-users May 2012