Archives : January-2014
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Im looking at setting type=peer vs type=user (in both IAX and SIP conf entries), and I found a comment attributed to digium (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that type=user is depricated and that we should only ..
Okay – maybe Im just suffering from a moment of horrible ADD – but, Im a little lost. I see that Asterisk 12 has a nice REST API – very nice – something I can use. However, and this is gonna sound dumb – but all the CLI commands are different now. W..
We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 resource unavailble. Most of the time I see this error on the fi..
is there a mailinglist where I can post questions regarding Digium IP-phones ?I have the following question :Im trying to provision a Digium D70 IP-phone from a https provisioning server.The Digium D70 contacts the provisioning server correctly but se..
I face a problem which look like the same as David with his thread Asterisk not receiving call from VPN address.I had an Asterisk (Elastix) working well in a VM (Debian Wheezy – KVM) having IP 192.168.111.14, my phone network is in the range 192.168.10..
All,Asterisk: 1.8.13.0Dahdi : 2.6.2Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686i686 i386 GNU/Linux OS : CentOS 6.4When I show meetme room details using meetme list command it shows Minus in activity column.Any Idea.Conf ..
Is there anyway to encrypt or scramble a bit the secret used to register with a provider? Im talking about theregister => fromuser@fromdomain:secret@hostdirective in sip.confThis clever dude modified the code back in 1.4:http://www.oneharding.com/voip/asterisk_md5_register.ht..
Im having some trouble turning with trunk monitoring while using an outbound proxy.While all other sip messaging (e.g. calls) respects the outboundproxy setting, Options packets from setting qualify=yes do not.Asterisk tried to send the Options mess..
When I issue a core show channels command I notice that long usernames (and channel number) are truncated.For example, if the username is FONEMITEL1234567890 for a trunk, then it will showSIPPrivilege: Command ChannelLocation State Application(Data)IAX2/FONEMITEL123..
Is there a mapping of AMI versions to Asterisk versions?eg:AMI 1.0 = Ast 1.4AMI 1.1 = Ast..