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..
Asterisk cannot act as a proxy, it is a B2BUA. If you want to make its behavior appear to be a proxy, there are a number of things you can do, but it will never just ‘pass along’ headers from an incoming INVITE to an outgoing INVITE.When Aster..
I have a Grandstream HT-502 device connected to my Asterisk PBX.It is configured not to place anonymous calls, and from my mostly layman reading of the invitation that the device sends, it should not be anonymous.However, the Asterisk PBX sends an anonym..
I am facing an issue while testing the video mail service of Asterisk. I have two different setup on one setup client being used is Mercuro while on the other client is Android based. On the Mercuro setup video mail is stored and retrieved properly wh..
Hi.I am using asterisk 1.8 and everything was working fine when I was at svn342661.I then upgraded to vrsion 349339 and discovered the following problem — one of the end points is a freeswitch box which offers a number of codecs, including PCMU.Howev..