Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFF..
, we are upgrading our Asterisk production server from 1.6.24 to 1.8.12 version and face the following problem: one of our peer (voicetrading.com) doesnt accept our calls anymore, we receive a timeout error Packet timed out after 32000ms with no respon..
Hi I have a small problems with incoming call. I have a peer actually configured for outcall : sip.conf: [Trunk-Telco] type=peer host=domaineofmysupplier.net outboundproxy=domaineofmysupplier.net session-timers=originate session-expires=7200 qualify=..
This is a really simple problem that I just cant get to work. There are two Asterisk servers with the following sip user and peer. When a call is attempted, Asterisk is not sending authentication details in response to the 401. Note, if the secret..
Your Server Voipon isnt responding- See if internet is working fine, or your Voipon sip trunk/peer is registered OK? On Fri, Feb 3, 2012 at 5:53 PM, virendra bhatiwrote: > Call is not routing from server to destination. > > > app8*CLI> console dial 00918885268..
On 01/28/2012 10:22 AM, Din Assegaf wrote: > All, > > Im trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, > > But when making A Call from SIP Client, I got cli Warning … and no > call has been made. > > My Sip Client is using lib j..