i wanted to make dial plan in such a way that the any incoming call to the sip phone should auto answer.(auto pickup) . Help.regar..
We realize there is an issue with a ticket system subscribed to the list and responding directly to members posts. We are working on resolving the issue, please bear with ..
This can and will happen, if the multi-interface Asterisk server is also being used as the router between these networks.In this case, the packet sent by the client system to 192.168.1.1, is being sent by the client to the clients active (possibly default/only)rout..
Kevin P. Flemingwrites:Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to move phones between the networks without changing the SIP server address, so you ..
people,I want to connect Asterisk with OpenBTS and make a call with a mobile phone.I use:Ubuntu 11.10 + Kernel 3.0.22GnuRadio 3.3.0Asterisk 1.8.13OpenBTS 2.8Nokia Mobile PhoneOpenBTS works and I can send sms from the OpenBTS server to the mobile pho..
I have this very specific problem with two dect sets. Problem that Ihave is one-way audio, in this very rare situation.I am calling with a Siemens N510 with C610 handset to Panasonic KX-TGP500 with KX-TPA50 handset. This gives me problems when I am call..
everyone,I am trying to apply thispatch on chan_ss7-2.1.0 for RingBack tone but its not accepting and throwing errors:Hunk #1 FAILED at 704. Hunk #2 FAILED at 715.I have done the patch modifications manually in l4isup.cThere is just one question, ..