Of Robert Lister
Sent: martedì 12 gennaio 2010 18.51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about SIP registration
On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote:
Then I have configured an account as following:
[999]
type
Hi guys,
I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC,
with eth0 set to address 192.168.1.1 (NATted over public network, with address
89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option
bindaddr=0.0.0.0; IP address to
Hi,
I have occasionally experienced the same problem too, and I suspect it was
caused by some spikes in network traffic (e.g. for an intensive file transfer)
that delayed too much SIP OPTION response, so that Asterisk marked these
devices as UNREACHABLE; I was able to use the devices too: in
Hi,
maybe this link can be useful:
http://www.voip-info.org/wiki/view/IAX+encryption
In particular, in your configuration I can't see the authentication method,
which must be md5, and a username to authenticate with, in either server.
But have a further look at the article, maybe you'll be able
-Commercial Discussion'
Subject: Re: [asterisk-users] destroy zombie session
What does the zombie call look like in core show channels?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aggio Alberto
Sent
As far as I could try some solutions, the only one that works as you like
involved use of Transfer() application, defining as 'tecnology' something like
that:
SIP/exten@ip_address
Where ip_address is the address of the peer you want to transfer the call to.
By the way, I found a scenario where
Hi all,
Some time ago I posted an issue regarding the hangup of active calls from the
CLI and someone told me that soft hangup should work. Well, in fact it does
work, but only if the channel is known, i.e. it doesn't work for zombie
channels. For example, I have this scenario (CLI output of
Hi,
it's quite straightforward: you can do your dialplan like this (default is the
default context answered when inbound calls happen) - remember the underscores!
-
[default]
exten = _1703,1,Goto(place-IVR,s,1)
exten = _1567 ,1,Goto(place-other,s,1)
[place-IVR]
exten =
Hi all,
I have a question regarding pending (zombie) SIP sessions: on Asterisk CLI,
with command 'sip show channels' , I see two channels in use with callID and
other infos detailed; also 'sip show inuse' give me same result (in terms of
channels usage):
PeerUser/ANRCall ID
