GitLab

No longer useful

Hangup if NO number have been dialed, except if call->CALL_STATE_TRANSFERT

You have to add your IAX2 account in /etc/asterisk/extensions.conf
No more segmentation fault when an incoming call is answered.

Replace iax_quelch by iax_quelch_moh ( with boolean = true )

Bug: With an IAX account, when you put someone on hold and then took him off hold, there was no outgoing audio from sflphone
Fix: We have to call audiolayer->startstream() to make the capture starts

Looking for translators ..

Almost no glitch on playback ( asterisk vocal messages )

Use of GTK_FRAME

The user can choose in the account window if he wants to enable STUN