Hi all,
I'm planning on picking up a Cisco IP phone or 2 and have a question
about the multiple lines feature of them, and Asterisk channels in
general. Lets say I have 2 Cisco IP phones and a call comes in, each
one rings line 1, and I pick up. Is there any way to have
notification on the
Forwarding isn't on. If I call from my cell: fax machine. Call from
a land line: call gets through. Call from Skype: fax machine again!
If only phone numbers were trace routable like IP addressed, to see
where the heck my calls are going.
On Mon, Mar 24, 2008 at 12:49 AM, John Faubion [EMAIL
I did do an upgrade to 1.4 (from 1.2), which is what lead me to try
phoning my BV number from my cell. It's long distance for me so I
don't call it ever. I wasn't able to get calls so I was placing a few
test calls to see if I could get it to ring. I was getting fast busy
signals at first but
Hi all,
I'm not sure if this is the correct mailing list for this (I was going
to send to Asterisk-Biz, but seems more for this one).
Anyway, I'm having more problems with Broadvoice. I still can't get
calls unless I comment out the secret= line in sip.conf, but now I
can't even place test
16, 2008 at 6:27 PM, Jon Miron [EMAIL PROTECTED] wrote:
Hi Raj,
Thanks for your response.
I'm a little confused though. Does this look as if it's a problem
with Broadvoice itself, and not my configuration? Any time I've
called them with problems where it's clearly not my
Hi all,
I just upgraded to Asterisk 1.4.18 a few days ago and I don't use
Broadvoice TOO often, however I have a Vermont number with them and so
my mother in law calls it to talk to my wife once in a while, so
that's why it took me so long to notice it wasn't working. Anyway,
when she calls she
problem.
--
Raj Jain
mailto:rj2807 at gmail dot com
sip:rjain at iptel dot org
On Sun, Mar 16, 2008 at 4:10 PM, Jon Miron [EMAIL PROTECTED] wrote:
Hi all,
I just upgraded to Asterisk 1.4.18 a few days ago and I don't use
Broadvoice TOO often, however I have a Vermont
Hey everyone.
I'm wondering if anyone has any ideas on a way to limit the number of
outbound calls at a time, and if the limit is reached a message is
played when someone tries to place the next call. I've searched the
wiki but have yet to come up with anything. Any help would be
appreciated.
Hey,
I'm trying to patch the latest CVS snapshot with the BroadVoice patch
but I get this when I try:
[EMAIL PROTECTED]:/usr/src/asterisk/channels# patch chan_sip.c sip_patch.diff
patching file chan_sip.c
Hunk #1 FAILED at 213.
Hunk #2 succeeded at 315 (offset 9 lines).
Hunk #3 FAILED at 485.
Hey all,
Wondering if this is possible.. Incoming call is
answered through X100P, then an extension is dialed
using the same X100P card. Basically I want to dial
in, enter 9 + phone# and have it do a flash then
have it dial *08 the same phone number + # on the
same PSTN line to have it transfer
--- Shaun Ewing [EMAIL PROTECTED] wrote:
On Wed, 22 Sep 2004 10:20:58 -0400 (EDT), Jon Miron
[EMAIL PROTECTED] wrote:
Hey all,
Wondering if this is possible.. Incoming call is
answered through X100P, then an extension is
dialed
using the same X100P card. Basically I want to
dial
Hey all,
I'm trying to get my Asterisk server up and running on
fwd.pulver.com just to get the hang of it until I get
my FXO card in a couple of days. It seems to connect
but that's about it. If I try to dial into it from
another fwd # it says user is not online.
In sip.conf I have the
--- Greg Hill [EMAIL PROTECTED] wrote:
This is relatively straightforward to implement in a
dialplan
(extensions.conf) either by implementing extensions
direction or by using
the DISA application. Keep in mind that a system
which allows an incoming
call to make an outgoing call has some
Hey All,
I have a question that I'm curious about. I want to
set up a 4 phone system in my home with 2 actual lines
coming into the house. Both or just regular lines
(not sure of this matters?), one being VoIP and the
other just a regular analog line. For now though I
just want the VoIP line
Hey,
I've checked all over and can't find what I need to
know, so I'm posting here. I want to use Asterisk
with my Primus VoIP service but it seems I need a
username and password to authenticate with at Primus.
Has anyone had any experience with this? How did you
get it? Is it stored
Jon,
Does Primus actually use MGCP though? I've heard mix
results (though keep in mind I only became interested
in all of this earlier today, so I know very little).
I checked the specs on my dlink and it says it's SIPs
with no mention of MGCP. However everywhere else says
Primus is not SIPs.
Jon,
Hmm I didn't know about the versions thing. I'll have
to get the exact model number off the device when i
get home.
I never set up any forwarding at all for it though. I
simply plugged it into my switch and everything was up
and running within a few seconds. Not sure if that's
a good
Geoff,
How frequent are your dropped calls? For a while all
my calls would go silent but I realized it was after
exactly 60 minutes. It's since been increased to 180.
Not sure if this is what you were experiencing.
Are there any providers in Canada that offer a similar
service to Primus that
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