November 2016 - sr-users

Kamailio + IMS + sipml5 call to softphone
by Serhat Guler 01 Nov '16

01 Nov '16

Hi, I have a setup as follows: IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for webrtc calls. Calls(both audio and video) between to sipml5 clients using firefox web browser is possible. The session is setup for the calls from sipml5 to Mercuro, but then there isn't audio flow as the codecs are not compatible. Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and OPUS codecs as firefox but this time the session isn't being setup. Boghe replies with "Reason: SIP; cause=488; text="Bad content" " I have seen a similar issue has been mentioned here: https://github.com/c00lz3r0/boghe/issues/157 but the initial invite request from sipml5 does have the SDP with media attributes.

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ACC module and BYE attacks
by Gholamreza Sabery 01 Nov '16

01 Nov '16

If you enable acc module to log CDR data into a database a user can send an infinite number of BYE requests to the server and all of them will be logged into the database as "481 Call/transaction does not exist"! How can we prevent this?

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/usr/sbin/kamailio - unresolvable destination error
by Ginti Saurabh 01 Nov '16

01 Nov '16

Hi, Please help to sort the problem. Getting "Unresolvable destination" error in /var/log/messages.

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Terminating PCSCF unresolvable request URI (Kamailio IMS Websockets)
by Serhat Guler 01 Nov '16

01 Nov '16

Dear all, I am trying to make a call between 2 webrtc clients(sipml5) using my kamailio IMS setup. The invite request fails at the terminating PCSCF with error code "Status: 478 Unresolvable destination (478/TM)". I have traced the packages and saw that the package that arrives at the terminating PCSCF has an unresolvable request URI due to the websocket connection. The package being sent from terminating ICSCF to SCSCF has the "Request-URI: sip:bob@net1.test" The package from terminating SCSCF to terminating PCSCF has the "Request-URI: sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws" I guess the terminating SCSCF replaces the request URI with the one it stored during the registration process, but the terminating PCSCF cannot resolve the host name&port number; therefore, it reports back "Unresolvable destination". Is that really the case happening in here ? How can I get over this problem ? Any ideas are welcome. Cheers, Serhat

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Server error for SIP client
by Hermann Norpois 01 Nov '16

01 Nov '16

Hello, if my SIP client tries to connect ... a server error occured (see below). Could you please help me to figure out the reason? Kamailio is installed on an amazon instance with an elastic IP (52.208.249.214, hostname: synth.ddns.net) Thanks Hermann ubuntu@ip-172-31-32-10:~sudo /etc/init.d/kamailio restart * Stopping Kamailio SIP Server: kamailio [ OK ] * Starting Kamailio SIP Server: kamailio loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: 172.31.32.10:5060 tcp: 127.0.0.1:5060 tcp: 172.31.32.10:5060 Aliases: tcp: ip-172-31-32-10.eu-west-1.compute.internal:5060 tcp: localhost:5060 udp: ip-172-31-32-10.eu-west-1.compute.internal:5060 udp: localhost:5060 *: synth.ddns.net:* ubuntu@ip-172-31-32-10:~$ kamctl db show subscriber +----+----------+----------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+----------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | Johanna | synth.ddns.net | 12345 | | 4fa26cedd456d7105c01a48742e71e42 | 1e249ac341d95701f855fce18bec0fba | NULL | ubuntu@ip-172-31-32-10:~$ sudo ngrep -d any -qt -W byline . port 5060 Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.5ad567803b13610279818912da2fcd81.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6a2baefed6cf4f6b6d34c7563d829e44.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6d74cec9e5f5e46dd9cd67338143127f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.a3a37d6bd0cd5d68fc786568f587f06e.0. Via: SIP/2.0/UDP 188.174.185.176:5060 ;received=188.174.185.176;rport=5060;branch=z9hG4bKPjAgF18hfvv20KOQ9H8G-fqfIPMjjcu9wq. From: <sip:Johanna@synth.ddns.net>;tag=vzlAJNNNES67ir6nsLMjvyFh5D0Thy6s. To: <sip:Johanna@synth.ddns.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3300. Call-ID: 9Uj4TSeQAVwaMs3UHf8UKIhuoA4xBEZt. CSeq: 39586 REGISTER. Server: kamailio (4.3.6 (x86_64/linux)). Content-Length: 0. . U 2016/10/20 13:44:33.727161 52.208.249.214:5060 -> 172.31.32.10:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.2605d82fd11bd8f788cfad582884eb6d.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.e425ad98e0436abc4661d1452a49670d.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.2d047656fb9df0edfccd4a40d20237b7.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.f54ab9c35b22814b15acbe16bcae387f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.db5960988332c2f4899a5e1c2b57e3c6.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.62f1191bd5af8480698ce118984536c0.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.5ad567803b13610279818912da2fcd81.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6a2baefed6cf4f6b6d34c7563d829e44.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6d74cec9e5f5e46dd9cd67338143127f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.a3a37d6bd0cd5d68fc786568f587f06e.0. Via: SIP/2.0/UDP 188.174.185.176:5060 ;received=188.174.185.176;rport=5060;branch=z9hG4bKPjAgF18hfvv20KOQ9H8G-fqfIPMjjcu9wq. From: <sip:Johanna@synth.ddns.net>;tag=vzlAJNNNES67ir6nsLMjvyFh5D0Thy6s. To: <sip:Johanna@synth.ddns.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3300. Call-ID: 9Uj4TSeQAVwaMs3UHf8UKIhuoA4xBEZt. CSeq: 39586 REGISTER. Server: kamailio (4.3.6 (x86_64/linux)). Content-Length: 0. 2) My logfile /var/log/kamailio/kamailio.log is empty though /etc/rsyslog.d/kamailio.conf exists with the entry: local0.* -/var/log/kamailio/kamailio.log And in /etc/kamailio/kamailio.conf: log_facility=LOG_LOCAL0 Why is the logfile empty? Thanks Herman

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Store and add custom variables for sip registrations
by Markus Bönke 01 Nov '16

01 Nov '16

Hallo all, we want to use kamailio as registrar server in front of freeswitch. When a user registers we need to lookup and store some user variables. To every INVITE that is received from a registered user we want to add the variables as x sip headers before we relay them to freeswitch. The question is what is the best way to implement this? We tried to store the variables in a htable on register and then lookup and attach the variables on every INVITE from there. Is this the correct approach, one issue we see is that htable is not writing new inserts directly to the database. I’ve also seen that there is a table location_atts in the kamailio db which looks like that it is intended to store custom key value pairs for registrations, but the seems to be no module which writes to that table, or is there one? Thanks for any help or suggestions Markus

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