Which version of spandsp are you using, current versions check on
ASTERISK_VERSION_NUM and handle the callerid accordingly.
Some time ago the makefile changed, but yours seems fine for that as you are
carrying CCFLAGS through, ensure that you're running a recent copy (and that
you've also copied
We've got the same issue, I'm
just starting to investigate it.
Have you resolved your issue now?
If not, I'll keep you updated on
what I find.
Steve
From:
Lars L. Christensen [mailto:[EMAIL PROTECTED]
Sent: 27 March 2005 16:14
To: 'Asterisk Users Mailing List -
also
noticed I do get a ring-tone when I call from line 1 to line 2 on the sipura,
so the problem must be the Asterisk. L
Looking forward to hear from you...
Cheers, Lars
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 5. april 2005 18:03
Have you tried the latest CVS, there was a bug relating to ALERTING which
was fixed yesterday...
-Original Message-
From: Ugur GUNCER [mailto:[EMAIL PROTECTED]
Sent: 08 April 2005 04:54
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Answering
I'm guessing he wants to do it the other way around, i.e. the external
calling party hears music, not the internal calling party making an external
call.
-Original Message-
From: Peter Svensson [mailto:[EMAIL PROTECTED]
Sent: 28 December 2004 21:53
To: Asterisk Users Mailing List -
:49
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Music instead of Tunes
On Wed, 29 Dec 2004, Steve Hanselman wrote:
On Tue, 28 Dec 2004, Marc Storck wrote:
more and more operators in Europe offer music instead of ring tunes.
E.g. instead
Discussion
Subject: RE: [Asterisk-Users] Music instead of Tunes
On Wed, 29 Dec 2004, Steve Hanselman wrote:
Are the two cases different in any way? The external call comes in, goes
to a context which eventually leads to a Dial(...) calling the internal
user. That Dial call provides music
Thinking about it we may well be able to do this in the UK as one of the
complaints I get about Asterisk is that our ring tone has changed to
external callers, (the zone is set correctly for zaptel, but it's different
from the normal ring tone), so the tones must be coming from the TE405, not
just
Accounts by itself would be useful.
-Original Message-
From: David Boyd [mailto:[EMAIL PROTECTED]
Sent: 30 December 2004 00:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Final call for departments
HOw about :
development
Dave
On Wed,
Some of the Cisco phones do (7940,7960 etc, also the 7910+SW but this is
skinny only).
Steve
-Original Message-
From: Erick Perez [mailto:[EMAIL PROTECTED]
Sent: 02 January 2005 21:35
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] phones with two ethernet ports
Hi
Has anybody looked into implementing a fax send interface for Asterisk using
the FSP code, that way it would plug straight into outlook and all the other
windows bits'n'pieces?
The information contained in this email is intended for the personal and
confidential use
of the addressee only. It
Has anyone also logged a support call with Digium, it has to be either the
card, Linux or the Zaptel drivers.
Steve
-Original Message-
From: Joshua McAdam [mailto:[EMAIL PROTECTED]
Sent: 14 January 2005 06:30
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
On Fri, 2005-01-14 at 09:23 +, Steve Hanselman wrote:
Has anyone also logged a support call with Digium, it has to be either the
card, Linux or the Zaptel drivers
DL380 G4 server
Steve Hanselman wrote:
I'm assuming that other non Digium cards work in it, but yes, you're
right.
Has anybody run any other PCI cards in those slots under Linux and seen
interrupts from those cards?
You'd be hard pressed to find a standard card requiring accurate
interrupts
Nothing to do with skinny, drop the new file(s) in your tftp directory and
edit the .xml file to specify the new version, the phone will upgrade itself
when it loads the config.
Steve
-Original Message-
From: Subhi S Hashwa [mailto:[EMAIL PROTECTED]
Sent: 23 January 2005 06:33
To:
]
Sent: 24 January 2005 14:46
To: Steve Hanselman
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [Asterisk-Users] chan_skinny and firmware upgrade
Monday, January 24, 2005, 9:23:50 AM, Steve Hanselman wrote:
Nothing to do with skinny, drop the new file(s) in your tftp
Stick that on the Wiki
-Original Message-
From: Tom Ivar Helbekkmo [mailto:[EMAIL PROTECTED]
Sent: 24 January 2005 16:31
To: Subhi S Hashwa
Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Steve Hanselman
Subject: Re: chan_skinny and firmware upgrade
Subhi S Hashwa
Get a converter to Q.931, we use one called
an IQ200 from (I vaguely recall) Teltrend, search the web, works fine, easy to
setup, we've used them at two customers now with no problems at all.
Steve
From:
Stephen Owen hosted [mailto:[EMAIL PROTECTED]
Sent: 08 February 2005
: Called
equipment is non-ISDN. (2) ]
-- Attempting native bridge of Zap/46-1 and Zap/32-1
Protocol Discriminator: Q.931 (8) len=5
Call Ref: len= 2 (reference 32527/0x7F0F) (Originator)
Message type: CONNECT ACKNOWLEDGE (15)
Steve Hanselman
Brendata (UK) Ltd
Tel: +44 (0)1268 466111
Fax: +44
problems (telewest - * - LG GDK 186)
Hi Steve,
please could you post your zapata.conf and zaptel.conf files?
Regards
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 15 June 2004 10:28
To: '[EMAIL
harm.
Is there a reason why you have only provisioned half your B channels on each
PRI span?; if that's all you need then that's fair enough.
HTH
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 15
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 15 June 2004 12:59
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] PRI problems (telewest - * - LG GDK 186)
Certainly, here they are (I've
Ok, changing the dial type makes no difference, but Telewest sends a
Sending Complete whereas Asterisk doesn't, this is almost certainly the
issue.
The channel is EuroISDN and overlap, so we should send it according to
q931.c?
-Original Message-
From: Steve Hanselman [mailto:[EMAIL
-thankyou) in new stack
skinny_answer(Skinny/[EMAIL PROTECTED]) on [EMAIL PROTECTED]
-- Playing 'queue-thankyou' (language
'en')
Steve Hanselman
Brendata (UK) Ltd
Tel: +44 (0)1268 466111
Fax: +44 (0)870 1387283
Mob: +44 (0)7973 750993
The information contained
I've got everything up and running, but I've hit an
issue and I'm not sure how to go about resolving it.
Here's our config:
LG GDK-186 PBX --PRI--- TE405P/Asterisk
---PRI--- Telewest (Telco provider)
I can make and receive calls from the GDK, and
make/receive calls from the VOIP
You just need to ensure that the first thing you say is This is an engineer
making a 999 test call and clear down as soon as they have confirmed.
-Original Message-
From: Wayne [mailto:[EMAIL PROTECTED]
Sent: 18 June 2004 23:07
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Testing UK
:27:52PM +0100, Steve Hanselman wrote:
LG GDK-186 PBX --PRI--- TE405P/Asterisk ---PRI--- Telewest (Telco
provider)
--snip---
Any ideas on where to start?
This is most likely to be a timing issue. You need to make sure your
asterisk is get timing from your telco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 20 June 2004 16:18
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk
I've changed the zaptel.conf to set both as internal, and it now seems to
work
* ?
HTH
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 20 June 2004 16:48
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk
They look odd to me for sure
Here's a diff that will patch it properly.
-Original Message-
From: Damian Minkov [mailto:[EMAIL PROTECTED]
Sent: 21 June 2004 10:14
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problem compiling fax applications
I'm tring to compile fax applications on Debian system.
the spandsp
That sound's like the right thing to do, you'd probably have contexts
related to what phones could do
unrestricted,localdialling,extensionsonly and then within those
contexts include the relevant contexts.
If you look at the sample configs you can see how this is done for the
international and
As I understand it, you'd enter the extension at which you wish to be called
back at, your 9665 has nothing to do with it.
Instead of dialling 28 you could dial 9665 and that would add that SIP phone
as an agent to the cytelcs queue.
Steve
-Original Message-
From: Harold Workman
PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
Sent: 20 June 2004 16:48
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk
They look odd to me for sure, I'm certain (99.9%) that Telewest would not
clock off of us, but as far as I can see
If you cat /proc/interrupts is anything else sharing with the TEs?
-Original Message-
From: Jim Gottlieb [mailto:[EMAIL PROTECTED]
Sent: 25 June 2004 20:31
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] panic() panic() panic()
Hi all. I've been trying to build some new systems, and
I'm going to modify the queue announcements to allow
for rounded seconds (e.g. we want to know to the tens of seconds. E.g.
Average wait 1 minute 20 seconds).
I'm going to add the optional announce of seconds to
the queue config and a rounding factor (e.g. 10 in our case).
The
1. Can't answer, sorry!
2. Yes, although you have to be clear about what you're proving, you're
proving that the cards work, not that asterisk is correctly configured and
will eventually talk exactly correctly to the entity that you'll connect it
to be a PBX or a telco. You'd configure one as net
Hi,
Is anybody in the UK using Telewest as
a PRI Telco provider?
Are you sending them caller ID?
I've been told by Telewest that:-
Oftel doesn't allow them to accept caller ID
(this is rubbish, and I replied pointing out where in the link to Oftel
that they sent me
-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On
Behalf Of Steve Hanselman
Sent: 30 June 2004 18:57
To:
'[EMAIL PROTECTED]'
Subject: [Asterisk-Users]
Providing Telewest in the UK with per extension outbound callerID
Hi,
Is anybody in the UK using Telewest
as a PRI Telco provider
call and detail
the number, if any, that is presented to theCalled Party.
HTH
Darren
--
Comgate
TelcoInternetBroadcast
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On
Behalf Of Steve Hanselman
Sent: 01 July 2004 09:57
It'll work, either as a SIP phone with the SIP image, or as skinny using
wither chan_sccp or chan_skinny (check the wiki).
Steve
-Original Message-
From: Matt Davies | MattDavies.Net [mailto:[EMAIL PROTECTED]
Sent: 02 July 2004 15:46
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
It's so you can have agents that are less likely to take calls (e.g. imagine
a sales queue, you'd have the sales people with no penalty, you might have
the receptionists with a penalty of 1 and us propeller heads in technical
support with a penalty of 2).
The technical support people would only
,ccs,hdb3 immediate = yes The line has not CRC activated (I have asked) Thanks!
Steve Hanselman
Brendata (UK) Ltd
Tel: +44 (0)1268 466111
Fax: +44 (0)870 1387283
Mob: +44 (0)7973 750993
The information contained in this email is intended for the personal and confidential useof
If the card plugs into the slot then it's not a power (3.3v/5v) issue as the
cards are physically different.
-Original Message-
From: asterisk [mailto:[EMAIL PROTECTED]
Sent: 09 July 2004 13:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dell 6450 / TE405p
I'm having some trouble
] On Behalf Of Steve Hanselman
Sent: 09 July 2004 13:01
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p
If the card plugs into the slot then it's not a power (3.3v/5v) issue as the
cards are physically different.
-Original Message-
From: asterisk [mailto:[EMAIL
red on
each channel.
Julian.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent: 09 July 2004 13:16
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dell 6450 / TE405p
Do the modules load (lsmod).
Are the interrupts assigned
controller: American Megatrends Inc. MegaRAID (rev 32).
IRQ 23.
Master Capable. Latency=32.
Prefetchable 32 bit memory at 0xf000 [0xf7ff].
[EMAIL PROTECTED] root]#
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Doesn't matter which span you use, but the Telco span should be set as clock
1 and CPE, the Meridian span should be set as clock 0 and NET.
Use only the channels you have been assigned on the Telco end, you as many
channels as you want on the Meridian end.
Take a look to see the order that
Sorry to bore you more with the clock issue, but have you check
/proc/zaptel/span to make sure it's not missing interrupts?
There's also an option to record the audio for the fax, you could listen to
that vs a recorded file that will receive correctly on a fax machine and see
whether there is an
, direct to a Telco pots
line or via an fxo card in asterisk?
Steve
-Original Message-
From: Stephen J. Wilcox [mailto:[EMAIL PROTECTED]
Sent: 14 July 2004 15:43
To: Steve Hanselman
Cc: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spa
ndsp
You can shut down the span in its entirety, or just exclude some channels in
zaptel.conf.
Steve
-Original Message-
From: Shawn Lawrence [mailto:[EMAIL PROTECTED]
Sent: 16 July 2004 08:03
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Flag Bad PRI Channel
Is there a way to flag a bad
Your priority assignment will probably cause that, if you have 2 people at
different priorities asterisk will only send to the other priority if the
best 1 is busy.
I'm guessing you really want a %age split? Whereby a is guaranteed 70% of
the calls and b 30%?
We used to achieve this on our old
That would certainly make sense, but I am not sure how to set an Agent's
priority. The only information that I have been able to find is setting
a QUEUE_PRIO value when queuing the calls (New as of July 2004).
Thanks,
Robert Jackson
-Original Message-
From: Steve Hanselman [mailto
They only get created as they are used and voicemail left, try leaving a
message and you should see that the structure etc is created.
Steve
-Original Message-
From: Steve [mailto:[EMAIL PROTECTED]
Sent: 19 July 2004 08:19
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Adding
In your dialplan for your voip routing you'd put a gotoif that jumped to
your PSTN context if it matched your criteria (e.g. EXTEN = faxextension)
Steve
-Original Message-
From: GIBERT Frédéric
To: [EMAIL PROTECTED]
Sent: 21/07/04 13:58
Subject: [Asterisk-Users] Caller based routing
Yes, you'd have a dialplan entry that set a value in the database, then
acted upon that.
You'd probably want some nice voice prompts
The system is currently in [Day/Night/Holiday] mode, press 1 to set to day,
2 to set.
Steve
-Original Message-
From: Massimo De Nadal [mailto:[EMAIL
You can have a refresh interval on the XML though which achieves the same
thing.
Also, you can do a push, there are examples in the developers kit available
on the Cisco web site.
-Original Message-
From: Wayne Sheppard [mailto:[EMAIL PROTECTED]
Sent: 04 December 2004 18:29
To: Asterisk
It looks like this is a splice between a couple of ISDN-30 lines and one or
more PBX's?
Are they both with the same provider, or with different providers?
We ended up adjusting the gain our ours as we would hear a distinct echo on
certain calls.
Other than that, you'll need to do the usual
PROTECTED] On Behalf Of Steve
Hanselman
Sent: 13 December 2004 12:13
To: '[EMAIL PROTECTED]'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Echo on one E1 line, but not the other
It looks like this is a splice between a couple of ISDN-30 lines and one or
more
You multiply to get the dollar price.
Careful where you go on holiday, it could be costing more than you think!!
-Original Message-
From: David J Carter [mailto:[EMAIL PROTECTED]
Sent: 11 October 2004 08:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Has anybody got any scripts for a lone worker system using Asterisk
before I write them?
Something along the lines of a regular phonecall with some kind of
random question (e.g. press 1 then 5) to provide monitoring of lone
workers with alerts?
Steve
The information contained in this
Discussion
Subject: Re: [asterisk-users] Lone worker system
On Mon, May 12, 2008 at 10:28 AM, Steve Hanselman
[EMAIL PROTECTED]
wrote:
Has anybody got any scripts for a lone worker system using Asterisk
before I
write them?
Something along the lines of a regular phonecall with some kind
Can anyone suggest how I can find the value of the call start time that
will be logged by CDR in the dialplan?
I've taken a look through the variables but I can't see anything that
seems to hold this?
The information contained in this email is intended for the personal and
confidential
/view/Asterisk+Detailed+Variable+List
Thanks,
Krunal Patel
On Thu, Oct 2, 2008 at 3:08 PM, Steve Hanselman [EMAIL PROTECTED]
wrote:
Can anyone suggest how I can find the value of the call start time that
will be logged by CDR in the dialplan?
I've taken a look through the variables but I can't
I think this is more related to the PRI, we've been seeing this for a
few weeks now, and our environment is bridged PRI-PRI on the same board,
Steve
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan
Zakaria
Sent: 10 May 2007 01:31
To:
, when dialing long
distance. So in this case, no PRI is involved. Its either the server, or
the network. Now I don't know how to find out what is it and why?
On 6/1/07, Steve Hanselman [EMAIL PROTECTED] wrote:
I think this is more related to the PRI, we've been seeing this for a
few weeks now
There seem to be two threads here that mention multi-second loss with
the common part being a PRI, certainly for my situation it's purely PRI
as the asterisk box sits in between the telco and another PRI enabled
PBX and the calls are bridged between the two.
There is no network traffic involved
We're also seeing the same thing, our calls are bridged zaptel calls between
ISDN30 PRI interfaces on a single TE410P.
We don't' appear to have any lost interrupts.
Same as stated, 2-3 second gaps in audio.
Steve
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
We're running 1.4.0 of asterisk
1.4.2.1 of zaptel
And kernel 2.6.20-1.2316.fc5smp
The information contained in this email is intended for the personal and
confidential use
of the addressee only. It may also be privileged information. If you are not
the intended
recipient then you are hereby
On a zaptel TE410p, when a call is bridged PRI - PRI how much involvement does
the processor have?
We're now seeing chunks of missing audio and I can't tell whether this is due
to a kernel upgrade or to a zaptel/libpri/asterisk upgrade.
I'm not seeing missed interrupts (from a cat of the
The setup.
Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum
updates applied), the TE410 lives on it's own interrupt.
Asterisk sits between our telco and a PRI enabled PBX.
These are the relevant versions installed:
Linux: 2.6.20-1.2316.fc5smp
Zaptel: 1:1.4.2.1-34.fc5
Asterisk:
occurring, that would help in fixing the problem.
Matthew Fredrickson
Software/Hardware Engineer
Digium, Inc.
On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote:
The setup.
Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum
updates applied), the TE410 lives on it's own interrupt
Digium, Inc.
On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote:
It probably did but we run in updates every week and nobody can state
exactly when the problem started only a few weeks ago - not very
helpful.
I can see that when I hear the issue the iowait time is high on the
processor.
Steve
switch's docs and
let us know what your FFR is and if you are doing any mirroring or link
aggregation.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Hanselman
Sent
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: 11 June 2007 09:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement?
On Mon, 11 Jun 2007, Steve Hanselman wrote:
This is the io wait
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Hanselman
Sent: 11 June 2007 10:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement?
I checked for BIOS upgrades
-Commercial Discussion
Subject: Re: [asterisk-users] Unable to match on CallerID in an include
block
What version of asterisk ?
Julian
Steve Hanselman wrote:
Is there any reason why I can't use the xxx/callerid format in an
include section?
It doesn't seem to work, but if I paste the lines
extension, I get the NoOp(Here)
If I dial 1234 from my 7701 extension, I get the NoOp(Hmm 7701)
Julian.
Steve Hanselman wrote:
Hi Julian,
Ah, a very good point, I put that in my first cut but had completely
forgotten in this one!
1.2.10
Steve
-Original Message-
From: [EMAIL
Fax2ps is what we use, works fine.
Yum tells me it comes from libtiff
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Artifex
Maximus
Sent: 08 September 2006 11:05
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Trouble with rxfax
Add the debug option to the rxfax line
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Danz
Sent: 15 July 2005 13:13
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Re:
SpanDSP rxfax, no tiff
Ive tried
both with and without Answer as
Are there any other GDK users out there with Asterisk?
Ive got all the integration working, except
voicemail.
Does anybody know a way of disabling the forward to
voicemail on a per extension or per DDI basis (I can disable the voicemail hunt
group but then I cant light the MWI
I think it's displaying the name of the line that the call is coming in on,
but you're expecting the name of the calling party (as I was!)
Steve
-Original Message-
From: Mark Johnson [mailto:[EMAIL PROTECTED]
Sent: 03 May 2005 16:44
To: Asterisk Users Mailing List - Non-Commercial
We do this, you need to ensure that you are allowed to control your
callerID (we had to request this from our telco)
You should then be able to use the SetCallerPres and SetCallerID to
control what (if any) number you give out.
Steve
-Original Message-
From: Robert Rozman
Jumping in very late to this thread...
Is the solution not to change the voicemail system to enable it to utilise
other entities as the store, e.g. a pop3 server or an imap server rather than
just flat files on disk (which should remain an option).
That way it doesn't matter where they listen
With call manager V4 and above it's extremely easy, just connect a SIP trunk to
*.
BTW Unity is the Cisco voicemail system, Call Manager (CCM) is the actual PBX
so your terminology may be confusing some people.
From: [EMAIL PROTECTED] on behalf of Simone
is for the PRI zaptel support?
Regards
Steve
Steve Hanselman
Brendata (UK) Ltd
Tel: +44 (0)1268 466111
Fax: +44 (0)870 1387283
Mob: +44 (0)7973 750993
The information contained in this email is intended for the personal and confidential useof the addressee only. It may also
I doubt they do, if they are marked as being there, but happen to be down then
the numbers would stay the same.
Sounds more likely that something happened with the clock source.
You'd need to reproduce it out of hours and look at the output of pri show span
x and cat /proc/zaptel/*
Might be worth asking the owner of voip-info.org if the mailing list
link can go on the left sidebar permanently?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: 28 June 2005 16:26
To: Asterisk Users Mailing List - Non-Commercial
It depends on your telco, in the UK on an analog line we can prefix it
with 141, so in that case yes, Asterisk can do it. You to find out from
your telco whether a caller with a standard handset can do anything to
control callerid with your telco.
Steve
-Original Message-
From: [EMAIL
And the UK although the PRI provider can either override or supply
it for you and you are normally limited (unless you've signed an
agreement) to DDI numbers directly provided by the PRI provider.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
A little off topic, but I'm on orange, what's the domain and what is the format
e.g. 07973 or +447973...
From: [EMAIL PROTECTED] on behalf of Wilson Pickett
Sent: Fri 01/07/2005 6:56
To: Mark Charlton; Asterisk Users Mailing List - Non-Commercial Discussion
Throwing errors
relating to utils.h:
/usr/include/asterisk/strings.h:264: parse error before
`__extension__'/usr/include/asterisk/strings.h:264: parse error before
`;'/usr/include/asterisk/strings.h:264: warning: type defaults to `int' in
declaration of
Sorry for the late reply but both of these are fine, we use spandsp to
print some faxes and email others.
We also route via a PRI to our other phone system to hylafax on an
analog modem and also to an analog fax.
So what you want to do is fine and will work.
Steve
-Original Message-
into an fxs port and call in to a spandsp extension the fax will be
received just fine, so I am left thinking it must be the PRIs, but if all PRIs
are this bad, how can anybody be using them?
Tom
On 5/19/06, Steve
Hanselman [EMAIL PROTECTED]
wrote:
Sorry for the late reply but both
Create yourself a crossover cable and loop the spans, set one to provide
clock and you should quickly see them come up, this will provide a very
basic test of hardware.
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 16
Try doing an strace on it and seeing what the last section shows you.
i.e. strace asterisk -vvvc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 31 August 2005 22:39
To: Asterisk Users Mailing List - Non-Commercial
/asterisk script
if I manually run
/etc/init.d/asterisk start
all's ok
if I manually run
service asterisk start
it says that it has started, but hasn't :)
Julian
Steve Hanselman wrote:
Try doing an strace on it and seeing what the last section shows you.
i.e. strace asterisk -vvvc
Can anybody see a way of detecting the current number of
retries remaining to a call file in the extension context that it is calling?
E.g. If I want to schedule a fax and I want to feed an email
back to the sender stating that the number is busy 2/5 retries remaining?
Steve
Probably easiest to set a variable to the number to be called and then
jump to an extension to do whatever you want to do?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruno
Voigt
Sent: 13 September 2005 23:37
To: asterisk-users@lists.digium.com
Our ISDN 30 is delivered that way, but we were also supplied with a balun
that takes the two balanced coaxs and turns them into a single RJ45, maybe
your telco needs to supply you with some extra kit?
Steve
-Original Message-
From: Kim Esben Jørgensen [mailto:[EMAIL PROTECTED]
Sent: 26
Delete it and cvs update will retrieve it.
-Original Message-
From: AJ Grinnell [mailto:[EMAIL PROTECTED]
Sent: 02 August 2004 17:33
To: Asterisk
Subject: [Asterisk-Users] App.c
Can someone tell me where I can get just app.c from. Mine somehow got
corrupted, and no updates or anything
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