Hello asterisk users,
Compiled asterisk-11.13.0 on openSUSE 13.1, however Channel driver chan_sip
is XXX in menuselect --- it depends on: chan_local(M), res_crypto(M),
res_http_websocket(M)
chan_local is [*] chan_local in menuselect,
res_crypto is in Resource Modules, Depends on:
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No chan_sip in compiled asterisk-11.13.0
Anthony Azzopardi wrote:
So this means that openssl(E) is holding everything?
Can someone give me some help on this?
On my Debian install, openssl shows as:
openssl
solved, permissions problem. Asterisks run with user asterisk at default, I
changed to asteriskpbx as the book says ;)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony
Azzopardi
Sent: 03 September 2014 20:57
Hello asterisk-users,
Just compiled and installed 11.12.0 however when I try to connect with
rasterisk I get:
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)
It seems that asterisk.ctl is not created.
--
Hello asterisk-users,
I need to compile dahdi and then export it to another system. I managed to
do this with DESTDIR=/root/destDir, then make a tar file and extract in / of
the other system. However the module is not loading and /dev/dahdi is not
created.
Anyone done this?
Thank you,
How do I monitor the whole conversation on a Zap channel without
answering it - the channel is hanging up, I think it's because it's not
answered.
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I want to monitor the zap channel while the phone is answered by a human
not by asterisk.
Tzafrir Cohen wrote:
On Wed, May 10, 2006 at 08:53:36AM +0200, Anthony Azzopardi wrote:
How do I monitor the whole conversation on a Zap channel without
answering it - the channel is hanging up, I
How do I monitor a Zap channel as soon as the telephone is off the hook,
till it is on the hook again?
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. And if by monitor you
mean record there's Monitor!
Anthony Azzopardi wrote:
How do I monitor a Zap channel as soon as the telephone is off the
hook, till it is on the hook again?
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Hello ppl,
I have my asterisk machine behind a Linux Nat router which is connected
to the internet. Please tell me the iptables rules and other
configurations that I need so that a sip phones on the internet can
access asterisk.
Best regards,
Anthony.
Where can I get the tar.gz sources of libnewt?
Reg,
Anthony Azzopardi.
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Can I configure the console to ring on one sound card and the headset on
another sound card?
Best regards,
Anthony.
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How can I send DTMF from the console?
Anthony.
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How come I don't have the MeetMe application registered?
Regards,
Anthony.
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How can I configure to call from the console by means of a sip phone,
any docs on this.
Regards,
Anthony.
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I can call from the console by means of the 'dial' command, now I need
to know how to call the console itself.
Anthony.
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