sr-users May 2019

sr-users@lists.kamailio.org

84 participants
96 discussions

Testing the KEMI interpreters for functions with 4 and 5 parameters

by Daniel-Constantin Mierla

Hello, because there were many variants not handled for KEMI functions with 4 and 5 parameters, I made a script to generate the code in order to be sure none is missing. The script was pushed in Kamailio repo at: * https://github.com/kamailio/kamailio/tree/master/misc/tools/kemi You don't have to use it, I listed it for reference, just in case the code needs adjustments in the future. If anyone is using a KEMI interpreter and has functions exported by Kamailio with 4 or 5 parameters, it would be appreciated if you can run some tests with master branch and report the results. All kemi interpreter modules were updated, so it doesn't matter which one you test. If the results are good, then the patches will be backported to branch 5.2, so there will be no case of unhanded functions with these numbers of parameters. The cases with up to 3 parameters were not affected. For 6 params, still only string values are handled. The other variants can be added, but I think they are not needed for the moment, so I tried to keep it small for the moment. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

4 years

Kamailio World thanks

by Daniel Pocock

Hi Daniel and Kamailio team, Well done on another successful Kamailio World in 2019. As usual, it was a great opportunity to meet people and learn new things. Regards, Daniel

4 years

siputils add_uri_param not available in KEMI python3

by Patrick Murphy

Hello, I am using the following version of Kamailio and using app_python3 with KEMI but I cannot find any exported function of add_uri_param from the siputils module. Any idea how I can perform: KSR.siputils.add_uri_param("name=value") ? version: kamailio 5.2.2 (x86_64/linux) 9260a2 Cheers, Pat.

4 years

Using a single virtual domain in a multi-domain setup

by Rhys Hanrahan

Hi All, I am trying to setup Kamailio to allow me to have multiple different domains (and IPs) treated as one. I am using the domain module for this. As an example I have: * A handset registered to Kamailio by IP address. * A SIP trunk between Asterisk and Kamilio using an FQDN that uses SRV records for redundancy on Asterisk's side. I have added all domains and IPs under the same "domain identifier". It seems the "old" domain module now uid_domain had this capability of a single virtual domain, but is this possible with the more current domain.so module? From what I've said the uid_* modules older modules from the SIP Router project, and should be replaced by more recent Kamailio modules? - https://www.kamailio.org/docs/modules/5.0.x/modules/uid_domain.html VS. - https://kamailio.org/docs/modules/5.1.x/modules/domain.html MariaDB [kamailio]> select * from domain; +----+------------------------------------+------------+---------------------+ | id | domain | did | last_modified | +----+------------------------------------+------------+---------------------+ | 1 | 12.34.56.53 | vocalpoint | 2000-01-01 00:00:01 | | 2 | 23.34.56.54 | vocalpoint | 2000-01-01 00:00:01 | | 3 | sbc.foo | vocalpoint | 2000-01-01 00:00:01 | | 4 | sbc.nsw.foo | vocalpoint | 2000-01-01 00:00:01 | | 5 | sbc.vic.foo | vocalpoint | 2000-01-01 00:00:01 | | 6 | sbc.foo2 | vocalpoint | 2000-01-01 00:00:01 | | 7 | sbc.nsw.foo2 | vocalpoint | 2000-01-01 00:00:01 | | 8 | sbc.vic.foo2 | vocalpoint | 2000-01-01 00:00:01 | +----+------------------------------------+------------+---------------------+ 8 rows in set (0.000 sec) I managed to get registrations working by hardcoding a static realm in the config, but when when usrloc does a lookup on a handset, Kamailio can't find user(a)sbc.nsw.foo and Kamailio returns 404. It looks like this is because in the locations table it is listed with a domain of 12.34.56.53 (because the handset is registered by IP). How should this work normally? Or is it even possible? Thanks, Rhys.

4 years

Sip proxy for Asterisk with RFC3966 and 100Rel

by paolo.visnoviz＠vipcomputers.it

Dear Sirs, a telco provider give us a trunk that require 100rel and TEL RFC 3966. Usually we use Asterisk. Now Asterisk 16 with PJSIP support 100Rel, but not RFC3966, and they don't actually works to implement it. Asterisk chan_sip, instead, support RFC3966, but not 100Rel and is not actively developed on by Digium or Sangoma. So I'm stuck. Now, do you think it was possible to use Kamailio to act like a proxy sip but implementing RFC3966 for incoming and outgoing calls and RFC3966 and 100Rel for outgoing calls? And it is possible to connect all to an Asterisk server? Thank you in advance Best regards. Paolo Visnoviz

4 years

Possible memory leak in siptrace module in kamailio 5.1/5.2

by davy van de moere

Dear, I'm using siptrace within a python handler in the KEMI framework. KSR.siptrace.sip_trace_dst("sip:<myhomerip>:9060") I notice an ever increasing usage of memory, until the system runs out of memory. (definition of a memory leak :) ). Looking at the memory information kamailio is giving me I notice this: May 14 05:49:48 server kamailio[36452]: NOTICE: PYT {INVITE} {1769595029-1513210001-1955500877} {1} : qm_sums: count= 79286 size= 5812048 bytes from siptrace: siptrace.c: ki_sip_trace_dst_cid(729) May 14 05:51:17 server kamailio[36452]: NOTICE: PYT {REGISTER} { 4_1419955802(a)192.168.0.153} {2582} : qm_sums: count= 79335 size= 5814840 bytes from siptrace: siptrace.c: ki_sip_trace_dst_cid(729) I notice this on kamailio 5.1.8 (from a debian repo), and on kamailio 5.2.2 (from source). I can reproduce the problem very easily, and I can apply patches in the source to try out things. Could I be wrongly using siptrace? Can I give you more debugging info? Kind regards, Davy

4 years

define a DBURL with SLQOPS fails

by David Villasmil

Hello guys, So, i'm defining: DBURL "mysql://user:password@dbhost/dbname" and later on... modparam("sqlops","sqlcon","cb=>"DBURL) and this fails miserably.. What do do on this case? Thanks all! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 years

Npdi question

by Duarte Rocha

Greetings, Regarding portability issues, is there a difference between "npdi" and "npdi=yes" ? Are there any more possible values besides "yes" and no value? Best Regards, Duarte Rocha

4 years

sdp_remove_codecs_by_id does not remove 101

by Juha Heinanen

I tried to use sdp_remove_codecs_by_id("101") to remove "codec" 101 (dtmf) from sdp, but nothing happened. It is still there: m=audio 31128 UDP/TLS/RTP/SAVPF 101 96. c=IN IP4 192.168.43.88 a=label:1. a=rtcp-rsize. a=ssrc:2880178656 cname:sip:test@test.tutpro.com. a=rtpmap:101 telephone-event/8000. a=rtpmap:96 opus/48000/2. a=fmtp:101 0-15. Is that by design or a bug? If not a bug, is there some other means to remove 101? -- Juha

4 years

sip trunk with kamailio

by arish haque

Hi, I am trying the SIP trunk from kamailio with following configurations - cfg_get.pstn.gw_ip="10.0.x.y route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(!($rU=~"^(\+|00)918884978235")) 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; } I am able to see kamailio sending INVITE to SIP Provider but no response back. Any other, configuration is required which I'm missing. I have attached the tcpdump snapshot of the above scenario. Thanks, Arish

4 years

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sr-users May 2019