Good day.I have a problem when using android native sip client. When dialplan used Progress (sending 183 Session Progress) after some seconds android native sip client declines a call (the logs are at the end of). No ealry media be heard.In same c..
Im trying to configure my Asterisk machine to work with Vitelitys vMobile service. I can place calls to the vMobile device and it rings as expected. However, I have no audio in either direction. Theres no NAT involved though. My asterisk machine ..
I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All the SIP clients are on a LAN, so no NAT is involved. I have been experiencing an intermittent problem where a call will be successfully answered, but then dropped by Aster..
I am running Asterisk 11 on CentOS 6.4 with about 150 local SIP clients on a LAN. The SIP clients are a mixture of Yealink phones (e.g SIP-T32G, SIP-T42G, etc). I have configured the system as follows:sip.conf:secret1111dtmfmode=rfc2833directmedia..
I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the appropriate ports. The SIP clients can be anywhere on the Internet, including behind NATs.I am able..
I have an Asterisk box with a public IP address and two SIP clients behind the same NAT device(I also have SIP clients behind different NATs). I want to know is it possible for Asterisk to detect if both clients are behind the same NAT and use dir..
all, I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this [general]context=internal allowguest=no allowoverlap=no bindportP60bindaddr=0.0.0.0srvlookup=no disallow=all allow=ulaw alwaysauthreject=..
Im running Asterisk 11.3.0 on wheezy. Im trying to do TLS +SRTP with blink SIP clients as shown here https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial.TLS is fine and I can call between clients. SRTP is a different matter, my SIP clie..
My wife and I have a home telephone/answering machine.Weve decided that having voicemail stuck in the machine and waiting for us to return home is not working for us any longer.Wed like for calls coming into our home to be routed to our cell phones..
I am using iax2 trunks between asterisk servers and am having a callerid problem. We are using realtime sip clients distributed between multiple servers. Only in test now but have run into a calleeid problem – the name of the called party is not displa..