I have a problem where on an outgoing call a Grandstream phone (GXP2130)closes the incoming voice stream about 1 second into the call (the remote party hears the Grandstream, the Grandstream doesnt hear thr remote party). I have verified with logs ..
Author : Harel Cohen
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How can I subscribe over AMI and what response should I look for ifI need to make a script that will SUBSCRIBE to the status of certain SIP..
List, Im working on an autodialer project. At the moment I use the Originate application then I throw it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate fin..
Max,Thank you for your detailed reply.Indeed all my nat-related settings are configured properly and I make use of two NIC cards to access the various networks Im connected to. When you mentioned externip I assume you meant externaddr because there..
List, I need your help with information going out on my SDP. Is it possible to update the Media Address on a per-call basis or a per-channel basis?Reason:My Asterisk is in a private network and needs to connect to UA on its internal network and a..
I have a question regarding incoming fax to local file (on the Asterisk server). While the fax is received properly (I have the tiff file generated as expected) I get the warning FAX CNG detected but no fax extension on the consol.If the fax is recei..
list, Im trying to install gmime22 package which is one of the packages reported as required by ./contrib/scripts/install_prereq test. Whatever I do Im getting to a dead end. On the regular yum repositories that I use (CentOS, epel, rpmforge, asteri..
List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIPUA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to ..
List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIPUA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to ..