fix it).
Its really strange how the other 3 lines work perfectly.
Does anyone have any ideas?
thanks,
Derek
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DID Number: 01 440 1806 (International: 00 353 1 440 1806)
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Answer(Zap/93-1, ) in new stack but it doesn't really answer.
This problem wasn't here in asterisk 1.0.X and although I upgraded to
1.2.5 I fairly well kept the configuration files the same.
Has anyone seen this happen before?
thanks,
Derek
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DID Number: 01 440
this?
Thanks very much,
Derek
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DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Local) 01 440 1800
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a configuration problem with the asterisk server or, by change, the same
bug with both softphones.
Has anyone else come up against this?
Thanks,
Derek
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shaping rather than trying to do it myself.
Does anyone out there know the correct answer to this? I'm sure many
people must have come up against this before?
Thanks for any help!
Derek
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Ireland
Conniffe
Rivertower Ltd
DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
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Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823)
Fax: 01 201 0085
it to also incorporate the Manager API too?
Thanks,
Derek
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Hi Gabor,
I'm not sure about USB ISDN adapters but I'm using an AVM Fritz ISDN
PCMCIA card with asterisk and chan_capi very sucessfully. The notebook
is in a remote location and is solar powered.
You'll find that these cards are cheap on Ebay - its a German card and
you'll probably find
.
Thanks for any help!
Derek
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DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
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Derek Conniffe Mobile: 086 856 3823
Hi,
Is there a command to start simpleswitch from an extension? For
example it would allow me to dial in to my * box and get a dial tone to
make an outgoing call.
Thanks,
Derek
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Ireland: (Local) 01 440 1800
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,
Derek
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DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
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Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823)
Fax
-init.conf
- start the cards: /etc/init.d/misdn-init start
- start your * and check if chan_misdn.so started properly
Hope it helps
Cheers
Jörg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Derek Conniffe
Sent: Friday, September 30, 2005 10:39 AM
/misdn-init start
- start your * and check if chan_misdn.so started properly
Hope it helps
Cheers
Jörg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Derek Conniffe
Sent: Friday, September 30, 2005 10:39 AM
To: [EMAIL PROTECTED]; Asterisk Users
this on, and gives this
message for, every channel.
Its a bit weird because I have not changed the configuration of asterisk
at all and cables were not even unplugged.
Does anyone know why this is happening?
thanks,
Derek
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DID Number: 01 440 1806
7940s with the native Skinny firmware and
chan_sccp by Sergio Chersovani. Chan_sccp not only supports call
forwarding but a lot more and only with the Skinny image, you can use
all the features, the phone have.
You can read a bit more at http://chan-sccp.org/
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Rivertower Ltd
Ireland
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-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Derek Conniffe
Sent: Thursday, September 15, 2005 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] USB ISDN
In a remote location (solar powered) I have an old Gateway
333mhz notebook
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Derek Conniffe DDI: 01
be inband ? but doesn't matter -
still NAT SIP isn't working
[EMAIL PROTECTED]
disallow=all
;allow=ilbc
;allow=ulaw
allow=g729
;allow=ulaw
;allow=all
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]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Conniffe
Sent: 13 September 2005 12:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Nat Sip Pain
Hi everyone,
I decided to have a look at SIP NAT again and I've been at it for a
[quite a] few hours but typically
be done but would be painful.
2. Never played with a STUN server, sorry just another point to break in
the chain?
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___
-Original Message-
From: Derek Conniffe [mailto:[EMAIL PROTECTED]
Sent
the
problem is when the far away server caches a domain name to an IP and
then the IP changes.
Derek
Michiel van Baak wrote:
On 11:36, Tue 13 Sep 05, Derek Conniffe wrote:
Really? I've found that a dynamic domain name -- IP linkup didn't work
with Asterisk - it seemed to caches the IP
loud - like comfort noise).
I upgraded to the latest firmware on the phone - Wj.00.10
Has anyone else had this problem?
Thanks,
Derek
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0180
Tel: (Ireland) 01 201 0146 / (International) 00
PROTECTED] On Behalf Of Giles Scott
Sent: 23 May 2005 19:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ZyXEL Prestige 2000W - cant make a call?
make sure you have 'canreinvite=no' in sip.conf
Cheers
Giles
- Original Message -
From: Derek
://lists.digium.com/mailman/listinfo/asterisk-users
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Fax: (Local
a
configuration problem and that the error message on their end is Remote
Frame.
I have this span setup just like the others. Does anyone have any idea
what RECOVERING and/or Remote Frame means?
Thanks very much,
Derek
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Thanks Peter - I'll try a bit of cable swaping checking - I think it
sounds like a cable fault.
Derek
Peter Svensson wrote:
On Tue, 19 Apr 2005, Derek Conniffe wrote:
But I'm having problems with one E1 line (span # 4). When I cat
/proc/zaptel/4 I get one of two messages in the first line
-108,110-124
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Hi Tim,
I tried that - it didn't seem to make any difference.
Thanks,
Derek
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Robinson
Sent: 18 April 2005 19:14
To: asterisk-users@lists.digium.com
Subject: [Fwd: Re: [Asterisk-Users] Only one PRI out
Hi Andres,
That sounds very hopeful - I had assumed that I could control all E1s
through a single D channel But maybe not.
Is it just a case of multiple group= lines with channel= lines
underneath in /etc/asterisk/zapata.conf or is defining the E1s individually
more complex?
Thanks again,
Hi Andres,
You hit the nail on the head.
I've defined 4 trunk groups and set an E1 for each trunk with logical span
0 for each E1 in each trunk - it seems to be working fine now!
Again - thanks very much,
Derek
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
= _011353.,102,Busy
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Fax: (Local Ireland) 01 201 0085
=no
callwaiting=no
restrictcid=no
threewaycalling=no
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Subject: Re: [Asterisk-Users] Cisco 7940, Voicemail DTMF
Derek Conniffe wrote:
Would anyone know why Voicemail in * doesn't get the DTML keypresses
from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to
do with dtmf_avt_payload: 101 setting in SIPDefault.cnf in the tftp
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Main Line: (Local Ireland) 1890 45 70
Would anyone know why Voicemail in * doesn't get the DTML keypresses
from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do
with dtmf_avt_payload: 101 setting in SIPDefault.cnf in the tftp server?
Thanks for any help!
Derek
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DDI: (Local Ireland
on the telephone) it doesn't detect my DTML
keypresses so when I press 1 for new messages it just ignores it.
Otherwise DTMF dialing is working perfectly.
Does anyone know why this is happening?
Thanks very much,
Derek
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Derek Conniffe
Rivertower Ltd
DDI: (Local Ireland) 01 201 0146 (International
Hi, I had no problem revering to an older firmware by downloading the
firmware I needed and installing a tftp server (the same server as my *
server) - when I updated the phones with the local tftp ip address and power
cycled them they loaded the older firmware.
Derek
-Original
.
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is interested in getting this project going again please
contact me off list.
Mark
Derek Conniffe wrote:
I did this - I'm in Ireland and needed sounds like Euro and Hash
rather than Dollars and Pound. I typed up the script of what was
needed, recorded it a number of times on semi-professional
as a WAV with
Audacity then chopped it up into the phrases.
If anyone else is interested in getting this project going again
please contact me off list.
Mark
Derek Conniffe wrote:
I did this - I'm in Ireland and needed sounds like Euro and Hash
rather than Dollars and Pound. I typed up
I meant wav - gsm there - too used to typing web I suppose :)
Derek
Derek Conniffe wrote:
sox will easily convert between the usual formats (like web - gsm)
- you just need to make sure you include gsm when running configure
for sox.
Derek
Daniel Eboa wrote:
There are good soft that do
Hi Doug,
Thanks very much - sorry I didn't see this in the wiki. The details you
quoted worked perfectly and I'm now on firmware V7. I still have the no
DTML tones recognised in voicemail but thats another problem.
All the best!
Derek
Doug Lytle wrote:
Derek Conniffe wrote:
Hi everyone,
I'd
connected to a remote telephone system that wants DTML
keypresses it works fine too (even though I'm using ilbc and IAX2
trunking for VOIP). It's only voicemail that is the problem.
Thanks for any advice!
Derek
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DDI: (Local Ireland) 01 201 0146 (International) +353 1 201
Hi - I found the trunkfreq directive in iax.conf so I've put the directive
line into the iax peers section (along with trunk=yes) - I'm sure you
meant iax.conf rather than zapata.conf ?
Thanks for the help,
Derek
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
00 as the International prefix but US
people (and VOIPJET) like 011 as the international dial prefix)
I'd be very keen to hear from anyone who has knowledge on using IAX2 to
route multiple outgoing calls.
Thanks very much,
Derek Conniffe
PS I notice that if three international calls are being
Yes, you can buy a clone. Yes, it may work currently (although I wouldn't
want to guess for how long). Also, the ??
actual cards that the X100Ps are based on have stopped being produced by
Intel, so you're out of luck as far as a
replacement goes in 6 months time.
I though that the X100P
the conversations
elsewhere over an IAX2 link.
Thanks for any advice,
Derek Conniffe
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Fax
I meant to say downgrade to 1.0.5.11
Derek
Derek Conniffe wrote:
It sounds like you might be using the broken BT firmware - are you
using 1.0.5.16? If you are you'll find that the message button and
the auto-dial feature do not work. You can downgrade to 1.0.5.16 but
I've read that the newer
"downgrade to 1.0.5.11 Derek Derek Conniffe wrote:
It sounds like you might be using the broken BT
firmware - are you using 1.0.5.16? If you are you'll find that the
message button and the auto-dial feature do not work. You can
downgrade to 1.0.5.16 but I've read that the newer
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]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch
Sent: 08 December 2004 07:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to play messeage when user picks up the
phone
Derek Conniffe wrote:
I'm doing this in a call centre with Budgetone 100
For my local test asterisk server I'm using an old HP Netserver E800 (only
one processor running as I had trouble with running two) - I have 7 PCI
cards (3 X x100p, 1 X ISDN, 1 X SCSI, 2 X NIC). The board also has its own
NIC 2 SCSI buses. They all have their own Interrupts (very high looking
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I notice that you've put the msns in as the msn field and have the
incomingmsn as a * character. I have lots of msns too and they all work
just fine (SuSE 9.1 AVM Fritz chan_capi) *BUT* I have the msn field as
my OUTGOING MSN for others to see and my incomingmsn as the list of comma
separated
Hi Philipp,
I have a notebook running in a remote location (powered by Solar) and
connected to an ISDN line with an AVM Fritz PCMCIA ISDN card (and chan_capi)
- this works very well for me. This may not be helpful to you at all as you
are specifically talking about a POTS line.
Derek
Chan_capi works fine on a 2.6 kernel for me (2.6.8 and SuSE 9.1) but I'm
using a AVM Fritz PCI V2 BRI card and I firstly installed the AVM fritz PCI
capi driver and then I installed chan_capi - everything went very smoothly
and I've installed with different AVM Fritz cards a few times now.
Derek
Hi everyone,
I'd like to try to use * ASTCC to create a pre-paid public call card
solution. In testing is looks good. I've been talking with Telecom
suppliers about supplying me with E1 Primary ISDN lines (probably 4 to start
with) and I'd purchase digium TE405Ps to connect everything up.
: [Asterisk-Users] chan_capi on 2.6 - impossible?
On Tue, 2004-11-30 at 13:50 +, Derek Conniffe wrote:
Chan_capi works fine on a 2.6 kernel for me (2.6.8 and SuSE 9.1) but
I'm using a AVM Fritz PCI V2 BRI card and I firstly installed the AVM
fritz PCI capi driver and then I installed chan_capi
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I think the USB IP Phone adaptor is a S100U - I found the TigerJet
website/products by reading the chip inside a S100U that I purchased at
digium and they look identical - but dont trust me on this - I didn't
buy one from TigerJet direct.
Derek
Michael Vogel wrote:
Derek Conniffe schrieb:
Re
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Hi,
There is a lot of talk about Cisco phones, SIP firmware and Contracts to
download same.
Does using a 7940/60 or other with SIP firmware offer better
features/compatibility with Asterisk over using the [default?] Call-Manager
firmware and chan_sccp? A lot of people here must have started
://lists.digium.com/mailman/listinfo/asterisk-users
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Main Line
on implementation.
I think this wouldn't be such a problem with a 7960 because a speeddial
button could be used to easily access voicemail.
Derek
Derek Conniffe wrote:
Hi everyone,
I have a Cisco 7940 and I'm using chan_sccp with it (chan_skinny does
work fine but it seems to be very featureless
.
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Main
I have read that many people have problems with Caller Id in the Uk.
What kind of surprises me is that incoming caller Id workes for me with an
X100P and Asterisk (pre 1.0 release) without any modifications.
I know that in Ireland we use the same wiring and phone connectors as the US
(RJ11 with
mobile numbers start with 08 followed by 8 digits).
The cellsocket now is working very well and I'm saving myself money
routing daytime mobile calls through a mobile phone rather than through
land lines.
Derek
Derek Conniffe wrote:
Hi,
I have not been successful in working out how to dial
-users
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idea how to fix this?
Thanks,
Derek
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hear no
echo) ?
Regards,
robert.
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, October 29, 2004 1:12 PM
Subject: Re: [Asterisk-Users] Echo in CAPI channels
On Fri, 29 Oct 2004, Derek Conniffe
by having Asterisk in the system (I'm running Asterisk on a
Pentium III 500Mhz with SCSI disks and the server doesn't do anything
else in life) this could be it.
Thanks,
Derek
Peter Svensson wrote:
On Fri, 29 Oct 2004, Derek Conniffe wrote:
I've been wondering about this too. I've now got two
such as a softphone or a hard ip phone you will have at an absolute
minimum an additional 40ms round trip time, but more realisitcally twice
that at least. That moves the echo from the sidetone to actually being
perceived as an echo.
Peter
Derek Conniffe
Rivertower Ltd
DDI: (Local Ireland) 01
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