1. Brekeke Product Name and Version: 2.4.7.3/286.1

Hi,

We do have the BKK Sip Server advanced edition version: 2.4.7.3/286.1.

Is the Failover plugin available in this version?
Do we have to install this plugin or is it out of the box available?

Thx

Thanks for your anwser! If i use another softphone the registration is correct. The ONLY difference i see is in the VIA header. We are able to make it work on our softphone by rewriting the address in the VIA header. However my question still is: Does BKK use the VIA header for the contact URI at ...

Hi, The remote IP address of UA is correct. It is the 172.29 address, see trace below and this detail in particular: 172.29.0.6.6022 > 172.29.0.1.5070: UDP, length 443 So, i still assume that BKK uses the VIA header for the contact URI at registration. Is that assumption correct? Full server trace ...

1. Brekeke Product Name and version: BSS 2.4.7.3/286.1 2. Java version: 1.6.0_17 3. OS type and the version: 2.6.18-194.32.1.el5.centos.plus 4. UA (phone), gateway or other hardware/software involved: softphone 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network ...

Hello Tschaikowskinksi,

We do have the same situation. Were you able to solve it or does someone else knows how to do this?

Thanks

Hi Harold,

Thanks, this works exactly as i wanted it. Great!

Is there somewhere a list of all the dialplan rules? I could not find this one anywhere in the documentation.

1. Brekeke Product Name and version: Brekeke SIP Server advanced 2.4.7.3/286.1 2. Java version: 1.6.0_17 3. OS type and the version: Linux 2.6.18-194.32.1.el5.centos.plus 4. UA (phone), gateway or other hardware/software involved: several 5. Select your network pattern from http://www.brekeke-sip ...

> Do you want to receive keep-alive packets at the certain SIP UA?
> And do you not want to receive them at others?

Yes that is exactly what i want!

And, your suggested dialplan did the job! Thank you very much!

Hi, I did not get this to work. This is what i tried: Matching pattern: $request=^REGISTER To=<sip:dummy_1@ Deploy Pattern user-agent=pipo2 $nat=true $action=register $continue=true I use the "user-agent = pipo2" rule for a quick check in the registererd client tab to see if the dial plan is being ...

Thanks, going to try it

1. Brekeke Product Name and version: Brekeke SIP Server advanced 2.4.7.3/286.1 2. Java version: 1.6.0_17 3. OS type and the version: Linux 2.6.18-194.32.1.el5.centos.plus 4. UA (phone), gateway or other hardware/software involved: several 5. Select your network pattern from http://www.brekeke-sip ...

Ok, that explains it, thanks.

So i assume there is no problem if i delete the expired records directly from the database by using a schedule or something.

Do you know what the awake field does? There seems no relation between that field and the expiration of a registration?

Thanks.

1. Brekeke Product Name and version: Brekeke SIP Server advanced 2.4.7.3/286.1 2. Java version: 1.6.0_17 3. OS type and the version: Linux 2.6.18-194.32.1.el5.centos.plus 4. UA (phone), gateway or other hardware/software involved: several 5. Select your network pattern from http://www.brekeke-sip ...

That explains a lot

We just ordered the advanced license

Thanks all for the help.

> does "select aaa from bbb where 1=1;" return anything if run from database directly? Yes > change log level and check if the dial plan is used from sv.xxxx.log We have sip, registrar and tcp loglevel at 255 but not any message from this dialplan in the log. >Which edition of Brekeke SIP Server are ...

>> I moved the dialplan rule to be the first rule to be executed and now it works: >Do you mean the DialPlan rule solved your problem?? Yes, with this rule a P2P connection is established! Since we will using the Sip server in a closed user group (with full control over the UA's) we will send ...

I moved the dialplan rule to be the first rule to be executed and now it works:

mode:RTPrelay = auto
mode:Auth = auto
mode:NAT = auto

I am still wondering why BKK does not decide for himself that p2p is possible, however, i can work with this solution by writing my own dialplan

Thanks!

Hi, > put mysql-connector-java-5.1.14-bin.jar to tomcat\webapps\proxy\WEB-INF\lib\ > restart tomcat > if there is any other dial plan set at sip server, put the one with database access on the top done, but still same problem > check sv.xxx.x.log with the date you test call > at tomcat\webapps\proxy ...

Hi Hope, > does the query "select aaa from bbb where 1=1;" work if run it from mysql server db? Yes > Is "mydb" your database name? Yes >is there mysql db driver copied at sip server\webapps\proxy\WEB-INF\lib\? Yes: /usr/share/tomcat6/webapps/ROOT/WEB-INF/lib/mysql-connector-java-5.1.14-bin.jar This ...

1. Brekeke Product Name and version: Sip Server 2.4.7.3/286.1 2. Java version: 1.6.0_17 3. OS type and the version: 2.6.18-194.32.1.el5.centos.plus 4. UA (phone), gateway or other hardware/software involved: X-lite on windows7 5. Select your network pattern from http://www.brekeke-sip.com/bbs ...

1. Brekeke Product Name and version: Sip Server 2.4.7.3/286.1 2. Java version: 1.6.0_17 3. OS type and the version: 2.6.18-194.32.1.el5.centos.plus 4. UA (phone), gateway or other hardware/software involved: several tested 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network ...

Hi Haddas,

I tried but it did not work, still relaying

> &register.contact.nat = false
What does this rule do exactly?

any other suggestions that i could try?

>Have you enable callee's STUN?

Yes, i did and i am pretty sure that it works because p2p calls are working well if i force BKK to do not relay (using a dialplan rule)

Why does BKK not recognise callee is using stun?

Hi Haddas, >Is the callee registered in the SIP Server? >Is the callee behind NAT? callee and caller are both registered and both behind NAT and both using STUN >Set the following setting and check the [mode:NAT] and >[mode:RTPrelay] fields in the log. mode:RTPrelay = auto mode:NAT = on { Dst-Far ...

Hi Haddas, We used several different UA's with same result. Here is an example (amsip softphone). First part is the initial invite from caller, second part is the invite as it is received bij callee (With BKK rtp relay setting: auto) Initial invite from caller *************************** INVITE sip ...