Questions tagged [FreeSWITCH] (62)

I must say that ip telephony is not strong, but the problem we have in the office integration is configured softfone and freeswitch the problem is that either the outbound or inbound to freeswitch shows error ORIGINATOR_CANCEL though in fact the call was

Use mod_xml_curl module for the formation of the gateways and directory server N1 Apache+PHP+Mysql the data is transmitted to the server Freeswitch N2. After the user on the server N1 is entered data for SIP on the server N2, they only apply after you restart fs_cli -x "reload mod_sofia" how to do it automatically? After th...

FreeSwitch has 2, one on white address, the other for NAT your organization (and even without the white addresses, then even NAT provider), which simply made use the Internet. On an external server was created extension for authorization. From inside the network was created a trunk to an external server under this user. Ca...

Or only for INVITE'and it is necessary to send auth-ceiling?

Hello. Have rented a sip phone, the task is to choose what to use for calls via the browser. Think about webrtc, asterisk, freeswitch. on the Internet a lot of articles, unfortunately they are very old, and I think this year, webrtc is very important. What bundle to choose to make calls to/from the browser? Asterisk, frees...

The client calls the Manager desktop phone, if the Manager does not respond within 15sec. there is a redirection to the mobile number of the Manager.Relied on this information<include> <user id="100"> the <params> <param name="password" value="$${default_password}"/> <param name="vm-password" v...

Please tell me how to interrupt the execution of the extension on the condition. The call goes to the Manager, if he took up that as usual, if within 20 seconds not answered, the call goes to another number.<extension name="group_dial_sales5"> <condition field="destination_number" expression="^(437)$"> <act...

Hello. Interested in the question of the organization of the conference the local sip clients, for example, a mobile phone or landline phones and other organisations. Local all clear, the group is created in /etc/freeswitch/directory/default.xml <group name="support"> <users> <user id="007" type="pointer"/&g...

Attempting to register a user in Freeswitch I get an error IN_PROGRESS "The client is in a transaction and cannot initiate a new one." When using Zoiper everything works, but in my app Registration failed. What could it be? Here classes sip and IncomingCallReceiver, rewritten documentation Android: public class SIPObject {...

How to implement a peer-to-peer connection? For voice transmission. I tried freeswith, but there need a sip provider for it to work. I need a connection in a non-local network. And without intermediaries type of some providers, it's possible at all? The problem seems to be that peers can find each other even if there is a C...