Recent content by voip.batsman

Hello , is this on asterisk ?. Please mention issue clearly . It could be from your trunk provider end . You have 2 options 1. Ask your provider to turn off session timer request . 2. In sip.conf fiel of your asterisk try to changing session-timers configuration . Below is the extracted...

Asterisk CDR can be stored in mysql database using odbc conenctor. We can choose which cdr database module to use at the time of compilation and installation of asterisk using make menuselect . You may encounter module errors If you configure wrong odbc drivers . For example if you look at...

Fusion PBX or ASTPP cluster can distribute calls between nodes to achieve high availability and scalability . Practically, a correctly configured freeswitch can run thousands of concurrent calls on modern high performance server hardware. At times you may need to setup freeswitch cluster to...

Freeswitch is the best open source VoIP Platform for production use . Adding Kamailio or OpenSIPS nodes in front of freeswitch makes VoIP networks more scalable and secure . It's practically a necessity for VoIP admins or engineers to learn some basic freeswitch commands to manage and...

We can design sip networks with openSIPS or Kamailio as a sip load balancer in many ways based on requirement and availability of resources. Refer to the below illustration in which load balancing and HA are designed at openSIPS level . You can also implement HighAvailability at freeswitch...

Is your issue resolved ?. If so please post it in this thread to help other people

Is your issue resolved ?

Following is a guide on " How To install openSIPS 2.4 on Cent OS 8 " and "fixing opensipsdbctl CREATE USER failed for 'opensips'@'localhost' Error" . The error may read like "You have an error in your SQL syntax; check the manual that corresponds to your My SQL server version for the...

Why you are posting this in voipnews section ?. You can not read or you just copy paste ?. On this forum many times it has been told that you should post in related sections only . Even there is a banner on top ? .

if you can yes ! . I may try to give you some hint based on that . Distros like freepbx have their own dialplans . You need to look at the lines where queue processing and mixmonitor section .

@Pavel Let me know if you can help me with async configuration push notification for kamailio proxy as front end . Idea is to to put kamailio as front end proxy and all registrations will be forwarded to asterisk or freeswitch or any other . Kamailio can hold database of client push tokens and...

@Akrobah I do not know your Xivo PBX dialplan . You can set and access Queue variables in dial plan something like .. exten => n,Set(thisQueue=${GLOBAL(QUEUE_${EXTEN})}) Mixmonitor monitor application funcion is MixMonitor(filename.extension,[options,[command]]) . So access the...

Hello , what is the issue ? Do you want to rename files ?. Can you explain ?

Hi mr.pavel welcome to global voipforum . Do you have solution for push notification forward using kamailio for asterisk or freeswitch sip users with nonblocking asynchronous operation . Kamailio is excellent choice for call routing with its high performance structure . Pbx/softswitch...

Asterisk is most of the IPPBX administrators favorite IPPBX . Asterisk is being actively updated with new features and improvements . Installing 18.4 on Cent OS 8 is a straight forward job. Let us see how we can install asterisk 18 on Cent OS 8 . First let us make sure our Cent OS is...