Tag : sip clients
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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:[169]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=..