Hi all,
I'm trying to resolve a weird issue with SIP routing.
I have a number of SIP trunks, from a selection of providers, all of
which are registered in sip.conf:
[general]
context=default
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
Hi List,
Until recently, I've been running an Asterisk server behind an MS ISA 2004
firewall. In general, this has worked fine - I've been able to connect to my
SIP provider to make/receive calls (sipgate.co.uk in the UK and
callcentric.com in the US), and DHADI runs the one traditional analogue
...@lists.digium.com] On Behalf Of
Ade Vickers
Sent: Friday, June 01, 2012 10:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Half-height PCIe analog FXO card
Hi,
Does anyone do a low profile PCIe FXO card? I just picked up
an HP ProLiant
Hi,
Does anyone do a low profile PCIe FXO card? I just picked up an HP ProLiant
microserver for $nuppence, which I'd hoped to migrate my Asterisk setup
onto. I currently use an A400P analog card, but the ProLiant only has PCIe
slots, and they're short ones too, so I can't use an A400E card. Even
Roger Burton West wrote:
I want to hook one of them to the PSTN. Given that I am in
the UK, what is a reasonably easily-available device to
provide an FXO interface from a Linux box, with a minimum of
faffing around with drivers? Just one line is needed, though
in theory two might
Brian wrote:
At the risk of being flamed
Has anyone had any success get the 'El cheapo' Wildcard W100P
clone's (£20 flavour) to work with UK Caller ID?
I'm not sure what the status of Asterisk 1.6 is with respect
to UK caller ID, being that we have an odd method of sending
the
: 08339 DCall: 16175 [**.**.***.***:4673]
USERNAME: 5111
DATE TIME : 2009-06-30 15:27:40
REFRESH : 60
APPARENT ADDRES : IPV4 **.**.***.***:4673
CALLING NUMBER : 5111
CALLING NAME: Ade Vickers (home)
Note in particular:
Tx-Frame Retry[ No] -- OSeqno
Gordon Henderson wrote:
Other than the price (nearly £150 difference), is there any
particular
reason not to pick an OpenVox A400-based solution for my UK
Asterisk needs?
None whatsoever.
I think the new digium cards are better at interrupt sharing,
but if that's not an issue
BJ Weschke wrote:
Ade Vickers wrote:
-Original Message-
Hi Folks,
I'm using the rawplayer program to provide my
music-on-hold, and it
works very well, with one small
drawback: every time I reset Asterisk, for any reason, the
MoH resets
to the beginning of the list
-Original Message-
Hi Folks,
I'm using the rawplayer program to provide my
music-on-hold, and it works very well, with one small
drawback: every time I reset Asterisk, for any reason, the
MoH resets to the beginning of the list. Since MoH isn't used
that often, it basically
Hi Folks,
I'm using the rawplayer program to provide my music-on-hold, and it works
very well, with one small drawback: every time I reset Asterisk, for any
reason, the MoH resets to the beginning of the list. Since MoH isn't used
that often, it basically means the same track is played over
Guillermo Salas M. wrote:
[ 830.118287] zaptel: Unknown symbol oslec_echo_can_identify
Make sure you get the latest version of OSLEC from SVN - the downloadable
tarball has a bug in it which prevents it from compiling properly (although
it acts like it worked just fine); which then prevents
Hi Hans,
Can't you leave the picking up of the cli to the isdn line?
Even if it is an ISDN1 (just a B-channel and a D-channel),
the chances of tranferring channel info, like CLI, is better.
If a call comes in over the POTS line, then I still need to get CLI over it.
I'm not sure if the ISDN
bilal ghayyad wrote:
I would like just to know one thing:
Where did u find a good IAX IP Phone?
I am looking in the market since long time to buy such device
and did not find a reliable one till now.
Any advise?
I haven't tried any yet; but http://x100p.eu have a few for sale; plus
Hi,
I've been asked to spec up a small Asterisk system, which needs to:
- Connect to ISDN2e (I'm thinking of using a B100P card here)
- Connect to the POTS (A400P with 1 FXO)
- Allow remote phones (thinking of an ETC 6050 utilising IAX2)
It is a requirement that the POTS analogue card picks
Paul Goodyear wrote:
I have had a BT phone plugged into these lines for about 3 week
prior to testing on asterisk, and all the lines are fine. Even
the first line, it rings and answers ok.
Apologies if this seems dumb, but have you done the swap the cables around
test? i.e. swap the cables
James Finstrom wrote:
Anyone have the telemarketer torture prompts? I would seriously like to
revive this.
I created a simple hold forever loop, thus:
[tele_loop]
exten = s,1,Answer()
exten = s,2,Set(DEVSTATE(Custom:telepark)=INUSE)
exten = s,3,WaitMusicOnHold(15)
exten = s,4,Wait(1)
Hi folks,
If you are running a call centre (large or small) using Asterisk, I'd be
interested to know how you log your agents in out:
E.g.
- Do you use AgentLogin (to force calls onto the agents, perhaps)?
- Do you still use AgentCallbackLogin?
- If you use AddQueueMember, are you
-
I have 2 asterisk servers - serverA and serverC - connected via IAX2.
On serverA, I have a telemarketer hold extension which, if I transfer a
caller into it, loops around playing music please wait messages, until
they give up hang up the phone.
Also on serverA, I have a custom devstate, which
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: 13 December 2007 14:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How do I do this?
- Original Message -
From: Ade
Steve Totaro wrote:
snippage
I suppose you could create a new context on server C, include
it in your internal context, and create an h exten on that
box to handle it locally. I am unsure why what you have does
not work but I assume the unable to transfer is a hint.
Except that, once
Philipp Kempgen wrote:
http://www.asterisk.org/node/48325
http://www.asterisk.org/node/48360
Brilliant, that works a treat, thanks! :)
Now, for my next question
I have 2 remote sites; 1 @ home, and 1 which I will shortly be transporting
to Spain. I've already set up my dialplan so
I wrote:
So, is there any way of monitoring the status of a device on
a remote
server, perhaps utilising an IAX2 channel?
To answer my own question, I did it :)
In case anyone's remotely interested in a similar setup/idea, here's the
relevant bits of my dialplan.
Assume that a call,
I recently implemented a simple spam trap extension for telemarketers -
once identified as a telemarketer (usually they ask to speak to the person
in charge of recruiting/website/purchasing/etc.), I simply offer to put them
through to the person in question, dump them on a special extension which
Compaq P3 1GHz server (about 6 or 7 yrs old) running 2gb RAM, 40(?)G HDD,
single AX100P.
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.488 / Virus Database: 269.14.3/1054 - Release Date: 06/10/2007
19:12
___
Hi,
Many thanks all for the useful tips - I've gone with a (simple!) mySQL table
with a flag in it, indicating the day/night mode, adding the following into
the dialplan:
[external]
; other stuff in here, excluded for clarity
; Include the SJS phone line controls
include = sjs_ctrl
[sjs_ctrl]
Hi folks,
I've been playing around with an Asterisk server in my office for a few
weeks now, and I've got it pretty much nailed down the way I want it, which
is nice.
One of the features I'm using is the ability to switch different contexts in
out of the dialplan on a schedule. So, for example,
Does such a beastie exist?
I've tried a couple of UT Starcom WiFi SIP phones (the F1000g and F3000
respectively), and found them both to be seriously lacking - regular crashes
(especially the F3000), poor battery life, and poor reception in particular.
However, whilst SIP phones are great, I'd
Brandon Kruse wrote:
/me goes to work.
There are none that I know of. There are only a couple of
IAX(2) hard phones, and none of them, that I know of, are
manufactured in the US anyways, and have problems.
(Of course, what is manufactured in the US these days)
That would be a
Hi...
I have what is, I am sure, a relatively common straightforward problem
(no, NOT that kind of problem!)... I'm trying to hook two asterisk servers
together so I can make a distributed PBX.
Here's the scenario:
[MASTER] is in the office. It has unrestricted access to the internet, and a
to the LAN address
instead of the WAN address).
Freaky? You betcha...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ade Vickers
Sent: 18 August 2007 23:47
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] 2 asterisk servers, how
Stephen Bosch wrote:
Ade Vickers wrote:
Hi Richard,
Thanks for those replies - I'll give them a shot shortly.
That's not really what I meant by configuration -- you can
choose the MOH source for Asterisk. It's only the native
player that restarts the music file every time someone
Stephen Bosch wrote:
Russell Bryant wrote:
Lacy Moore - Aspendora wrote:
On 6/29/07, Ade Vickers wrote:
What I'd like to do is have the music streaming
constantly, so the on hold
caller always gets music at the current position; even if
that's in
the middle or near the end
: [asterisk-users] Music on hold - 1.4.5
Richard Lyman wrote:
Ade Vickers wrote:
*snipped
Hi all, thanks for the responses so far.
I too understood it to be a configuration thing, with the
addition of
a streaming music server (which, obviously, provides
Hi,
Please bear with me if I'm asking stupid questions... I'm new to Asterisk,
newish to Linux, etc...
I've got MoH working nicely with my new Asterisk setup using the files
option; except that it always plays from the start of a (random) music file
when you first put someone on hold. Take them
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