All,
I have not found/seen a resolution to the issue where my TDM400P seems
to cause problems, as outlined in the mISDN (HFC-S) and TDM400P -
mISDN: ISAC XDU no TX_BUSY thread.
I have also not found/seen a simple 'how to' on patching DAHDi with
ZAPHFC as outlined in the HFC-S card thread.
Do I
On 23 February 2010 13:16, Razza razz...@gmail.com wrote:
On 23 February 2010 12:58, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Have you managed to install those zaphfc drivers?
Those are basically the same ones from http://code.google.com/p/zaphfc/
Hi Tzafrir. I checkout out
On 22 February 2010 14:07, Razza razz...@gmail.com wrote:
I'm using CentOS5.4, can anyone advise how I can make DAHDi work with
a generic HFC-S card?
On 22 February 2010 15:12, Pedro Santos pnlsan...@gmail.com wrote:
I´m using centos 4.8 server, and i don't now how integrate zaphfc with dadhi
On 23 February 2010 12:58, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Have you managed to install those zaphfc drivers?
Those are basically the same ones from http://code.google.com/p/zaphfc/
Hi Tzafrir. I checkout out that but there were no instructions.
--
On 22 February 2010 10:26, Per Jessen p...@computer.org wrote:
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if
that helps.
/Per Jessen, Zürich
Not meaning to hi-jack this thread, I’m not
On 22 February 2010 13:02, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Nowadays (as of Asterisk 1.6.0) BRI support is included in Asterisk. The
zaphfc driver, though, is still not included in DAHDI. It's maintained,
though. The version included in the Debian packages is taken from
On 19 February 2010 15:16, Razza razz...@gmail.com wrote:
I had Asterisk 1.6.2.2 running fine with a mISDN using a HFC-S based card. I
installed my TDM400P into the PC, it's really slow to boot now, when it
finally does I gets stuck in a loop of reporting isac xdu no tx_busy.
Anyone able
I had Asterisk 1.6.2.2 running fine with a mISDN using a HFC-S based card. I
installed my TDM400P into the PC, it's really slow to boot now, when it
finally does I gets stuck in a loop of reporting isac xdu no tx_busy.
Anyone able to assist?
Thanks in advance!
--
Hi all,
I have been running Asterisk for years (CVS-HEAD on 2005-08-24) with no
problems save a failed harddrive. I have decided to build a new box and have
Asterisk 1.6.2.2 playing nicely with mISDN after lots of changes to dialplan
syntax etc. I am struggling with SIP trunks to sipgate.co.uk
Seems the only option is to give Elastix a go.
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Thanks for your response. I gave loads of info in my original mail, surely
someone can help without jumping distro?
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I'm running centos, so tried a yum upgrade but nothing was marked for
upgrade. I've reinstalled bluez-libs.i386 0:3.7-1.1.
I've tried a different dongle, but still get the same message.
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I've been failing to get chan_mobile working, so am looking to the experts
to help :o)
I have followed this guide -
http://www.voipphreak.ca/2008/10/30/installing-and-configuring-chan_mobile-for-bluetooth-presence-support-in-asterisk-16/
and this guide -
Hi all, does anyone know of an application that will run in Windows (in my
case users PC's) and behave in a similar fasion to chan_mobile? I'd like the
app to register with asterisk, then talk to a (or a number of) mobiles over
bluetooth thus creating an FXO port? I'm not interested in SMS etc.
I have quite an old version of Chameleon Mail, currently the prompts played
when leaving a message are –
-- Executing VoiceMail(SIP/209-3b0e, u5) in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/5' (language 'en')
-- Playing 'vm-isunavail' (language
Thanks kindly works a treat :o)
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2009/3/6 Kevin P. Fleming wrote:
It was just merged into Asterisk SVN trunk, on its way to becoming part
of Asterisk 1.6.2. It's new and still has some limitations, but progress
is being made and we welcome additional testing!
Excellent, checked out trunk and indeed it's in menuselect :o)
Are
Hi all, I’ve read that meetme works at G711 (ulaw), so asterisk would
down-mix a number of parties using G722, is that still correct?
If so, i’ve also read that Joshua Colp was/is working on a replacement
(conf_bridge?) that works with G722. If this is this still in active
development are there
Thanks Klaus. Putting both in the same context solved my issue!
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Hi all,
I have two sipgate accounts (numbers), if I have both accounts register only
one will work for incoming calls (which is all i'm interested in). However
if I disable either account the other account will work perfectly. Am I
missing something obvious?
Thanks in advance,
Ray.
Excerpts from
2008/12/30 Jeff LaCoursiere wrote:
Oops - I take it back: http://www.audiocodes.com/gpl-lgpl
Looks like they are at least attempting to comply... did you follow these
steps?
Thank goodness you posted that, it's saved weeks of self flagellation after
reading the earlier posts. ;o)
So now we
Please see below Console Messages, Pertinent section of SIP.CONF and
AudioCodes Config.
*Console Messages:*
Dec 28 18:14:45 NOTICE[19109]: chan_sip.c:9808 handle_request_register:
Registration from 'sip:2...@192.168.10.4 sip%3a...@192.168.10.4' failed
for '192.168.10.4'
Dec 28 18:14:45
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk also, I
want all the channels to register with asterisk.
I have the FXS channels working fine, I cant acheive that with the FXO
channels, does anyone have any advice or possibly sample configs.
Thanks in advance :)
For interested parties
It would appear I didn't need to had anything between the
loadInformation/loadInformation tags in XMLDefault.cnf.xml and/or
SEP{MAC Addr}.cnf.xml to force the upgrade, although the correct version of
firmware needs to be present between the tags once it is converted.
On 31/03/2008, Greg Oliver [EMAIL PROTECTED] wrote:
For a 7965, you might try loadinformation to be 335.. I have had to
match up CCM tk.prod values to match on newer phones in the past to be
what cisco uses in their internal database before I could get them to
work. Although, leaving those
I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S).
Does anyone have a valid XMLDefault.cnf.xml they could share?
I have tried the version at
voip-infoinfo.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIPview_comment_id=14768#Troubleshootingfor
the
On 31/03/2008, J. Oquendo [EMAIL PROTECTED] wrote:
YMMV Change to reflect your firmware (e.g. P003-07-4-xx)
8 SNIP 8
I removed the following lines:
loadInformation8 model=IP Phone 7940P003-07-4-00/loadInformation8
loadInformation7 model=IP Phone
Imagine, repairing an engine of your brand new car you just bought?
Imagine restarting your TV because it just froze? What if your shoes
have just changed colour to blue screen?
It will just not pass, will it? ... You will DEMAND a service for your
car/TV,shoes or you may return it or
On 11/03/2008, Senad Jordanovic [EMAIL PROTECTED] wrote:
I would suggest to you learning how to use text emails and quoting first
then you may have some responses that may be your worth while.
Clearly, you keep responding!
Oh and move out of the dark ages and get a decent mail reader.
On 10/03/2008, Matt Riddell [EMAIL PROTECTED] wrote:
Has anyone done any integration with this?
All I know so far is that it appears to use some non standard form of SIP.
Any pointers?
If you are looking to use Enterprise Voice (voice breakout between the OCS
and PBX environment) as opposed
On 18/02/2008, Razza [EMAIL PROTECTED] wrote:
Thanks every one, having looked at
http://www.cisco.com/warp/public/788/products/1750-vic-issues.html and
http://www.cisco.com/en/US/products/hw/routers/ps259/products_tech_note09186a00800e73f6.shtml
It appears to support a single BRI I need
Is anyone using a cisco router as an ISDN gateway with Asterisk?
As you might have seen from a couple of my threads, I have been looking at
Fritz! and Cologne cards, both of which require development against a
specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive
and causes a
On 18/02/2008, Tuukka Laurikainen [EMAIL PROTECTED] wrote:
Just make sure you have the dsp's necessary installed for the simultaneous
calls
on the router you're planning to use.
Thanks every one, having looked at
http://www.cisco.com/warp/public/788/products/1750-vic-issues.html and
On 17/02/2008, Jaap Winius [EMAIL PROTECTED] wrote:
Interesting. I still have several AVM Fritz!Cards, but I stopped using
them after I upgraded my server because I could no longer get them to
work. I used to compile the fcpci module for kernel 2.6.8, but it
doesn't work with 2.6.18. AFAIK,
Hi list, i'm keen to move to Asterisk 1.6, so really need to update my
system which is running Mandrake 9.2 although it has been solid for years,
fo Fedora 8.
I have a Fritz! card for ISDN BRI, I have installed the drivers/kernel
drivers from ATrpms
On 13/02/2008, Raj Jain [EMAIL PROTECTED] wrote:
SIP over TCP is included in 1.6.
http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co
Thanks all! :o)
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On 13/02/2008, Bill Andersen [EMAIL PROTECTED] wrote:
Has anyone tried to used VB6 to communicate with the Asterisk Manager?
If so, would you be willing to share some basic code showing your
approach to getting connected and parsing results?
Bill
I wrote some very very basic stuff ages ago
When I first set up asterisk, I had similar, fortunately not with the old
bill!
It basically boiled down to not enough delay between seizing the line and
dialing the digits, for example the following would have dialled the coppers
012*99 9*12345, as 012 would have been missed.
I'm guessing this
I am aware there is a SIP over TCP patch. Will this ever become part of
a release, if so are there any timelines?
Thanks in advance.
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On 10/10/2007, Reggie Payne [EMAIL PROTECTED] wrote:
Hello All! I am new to the list. Does know how to record a call on
demand? What I would like to do is setup something that during a call
someone can hit a button a the call is recorded the after the call is over
the recording is sent to
I second calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc.
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EPIA 5000 wit 512MB RAM, TDM400P (1xFXO) and FritzCard for ISDN. Running
Mandrake 9.2 as later distros I have tried (Fedora) wont play nicely!
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On 27/09/2007, Eric B. [EMAIL PROTECTED] wrote:
For starters, what is the difference btwn the 1.2 and 1.4 branches of
Asterisk? I can't seem to find a document that describes the changes.
Anyone?
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On 29/08/07, Razza [EMAIL PROTECTED] wrote:
On 29/08/2007, Patrick [EMAIL PROTECTED] wrote:
On Tue, 2007-08-28 at 20:51 +0100, Razza wrote:
HI all,
Has anyone succesfully installed an AVM Fritz! card on Fedora 7 using
the drivers at ATrpms recently? http://atrpms.net/dist/f7/fcpci
On 29/08/2007, Patrick [EMAIL PROTECTED] wrote:
On Tue, 2007-08-28 at 20:51 +0100, Razza wrote:
HI all,
Has anyone succesfully installed an AVM Fritz! card on Fedora 7 using
the drivers at ATrpms recently? http://atrpms.net/dist/f7/fcpci/
I tried with a clean F7 build on my EPIA 5000
HI all,
Has anyone succesfully installed an AVM Fritz! card on Fedora 7 using the
drivers at ATrpms recently? http://atrpms.net/dist/f7/fcpci/
I tried with a clean F7 build on my EPIA 5000 yesterday, after modifying
/etc/capi.conf (removing the coment # in front of fcpci line) I received the
Nope, it only has chan_capi. I don't have any experience with AVM Fritz
cards so I'm afraid I can't help you with it. I think there is an
article on voip-info.org that explains howto use a Fritz card with
Asterisk.
Regards,
Patrick
Patrick thanks! I guess my question should have been
What problem did you have? I did not have any problem. You can find the
chan_capi SRPM here:
http://www.laimbock.com/asterisk/
Regards,
Patrick
Hi Patrick,
Does the '*chan_capi-1.0.1-8.el5.lc.src.rpm'* include the driver for the
AVM Fritz Card (AKA BT Speedway card) ?
I cant for the life
Is it possible to bridge a media stream, lets say created by VLC on to an
Asterisk channel?
What I would ideally like to do is - When mobile dial into my Asterisk
server, follow through some security/prompts then, through the dialplan
launch VLC as an external application. VLC would connect to an
Kevin P. Fleming on 12 July 2007 19:52 wrote:
8 SNIP! 8-
we haven't yet found the root cause of
the delays, but it does appear to be a large number of subscriber
addresses that fail to resolve via DNS (but 'soft failures' (timeouts),
not hard failures).
As we get through the
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Hi all, is it possible to to dumb down a FRITZ!Box Fon ata (
http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_ata/index.html
http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_ata/index.html##)
and have the two FXS ports AND the ISDN interface register with
Asterisk. In much the same way a
trixter wrote:
do you have a url for an online demo?
http://www.scansoft.com/speechworks/realspeak/demo/default.asp
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Or even . http://www.asteriskwin32.com/
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Alchaemist Wrote:
Now... if you have dynamic IP in the asterisk... things change
because
Asterisk must know in sip.conf the external IP.
I think I read in this list, that the best (only?) way to get arround,
is to
place a script that detects the external IP when it changes,updates
sip.conf
Colin Anderson Wrote:
Assuming your Asterisk mail client is set up correctly, cron this twice
a day on your Asterisk box:
ifconfig eth0 | grep inet addr | mail -s My Asterisk IP address
[EMAIL PROTECTED]
Check your mail before you call and verify that the IP address has not
changed, if it has,
Jens Wrote:
Who needs that when there's dyndns and similar free services which
are even supported by many routers? I have a dyndns hostname and my
router is configured to contact the dyndns site whenever the IP on
the public side changes. Works very well for my Asterisk setup at home.
I'm
Ray Wrote:
I'm sure if you use a DNS in SIP.CONF for your external IP this is
only
resolved when loaded?
Jens:
This might be true - for me there's only other Asterisk servers
connecting from the outside using IAX, and that works fine.
IAX is different! So back to sip, assuming if externip
Colin Anderson wrote:
The one I like:
http://www.rhetorical.com/cgi-bin/demo.cgi
is toast. I think they went broke or got aquired by someone. Also, is
there a Festival voice that sounds as good as Rhetorical or the AT
T
stuff?
According the UK Companies House, they are still going.
Derek,
You said -
Needless to say when I don't have any NAT settings on the SIP phone I
don't get any registration with the * server (this confuses me too - I'm
not sure why I only get registration when I set the * server to be the
outbound proxy? Maybe its because the SIP phone sends its
Could you send me a copy of your script?
%- SNIP! -%
Wilson Pickett wrote:
What I did was to have the ip checker write a one line file called
externip.conf containing the line:
externip = nnn.nnn.nnn.nnn ; this is the new ip address
then in sip.conf,
#include externip.conf ; replace
Angus wrote:
But the systems are sold in this configuration. There is a fan option.
I chose the fanless option.
Not sure this is really a debate for this group? I think they sell
fanless options as there are cases capable of sinking the heat without a
fan or for custom oem development.
My
:o)
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Title: Message
Did
you burn the iso properly or as a file on the cd?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ChandsSent: 30 August 2005 19:20To: 'prashant yadav';
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE:
Title: Message
I'm assuming no apps/scriptsexist which completes
this?
Can
someone please confirm thatif I use a FQDN in sip.conf for my external IP,
the FQDN is only resolved at the time of loading, therefore if my IP changes
after sip is loaded, I will have to manually reload
Title: Message
I'm
assuming no apps/scriptsexist which completes this?
Can
someone please confirm thatif I use a FQDN in sip.conf for my external IP,
the FQDN is only resolved at the time of loading, therefore if my IP changes
after sip is loaded, I will have to manually reload
Title: Message
I have a standard BT
home DSL, which meansI cannot have a static IP address, therefore i'm
forced to use NAT,I subscribe to a DDNS service and have written a VB app
which polls the router every 10 seconds and updates the DDNS if appropriate.
This is fine but I
need to be
--% SNIP! %---
I have a standard BT home DSL, which means I cannot have a static IP
address, therefore i'm forced to use NAT, I subscribe to a DDNS
service
and have written a VB app which polls the router every 10 seconds and
updates the DDNS if appropriate.
There are ready
--%
On Wednesday 24 Aug 2005 09:44, razza wrote:
I have a standard BT home DSL, which means I cannot have a static IP
address, therefore i'm forced to use NAT, I subscribe to a DDNS
service and have written a VB app which polls the router every 10
seconds and updates the DDNS if appropriate.
Ditch your ISP
Title: Message
All,
wondering if you can
help, I had a perfectly working Mandrake 9.2 box running on a via Mini ITX
5000/classic. Asterisk (zaptel and libpri)was built from CVS head around
22nd July 2005. I decided now was a good time to ghost it upalthough
humorous for you all suffice
Title: Message
All,
beforeI start,
I have read the readme and realise the dial syntax has changed since
3.5.(and I really wish it hadn't) anyway can somone please help me round a config
issue?
Previously I had a
capi.caonf as follows. This config allowed me to have multiple MSN's
Title: Message
Is any one willing to make their TFTP server
available and try this upgrade for me?
Razza26 May 2005 18:43'Status messages' simply shows SEPMACaddr.cnf
for info, 'Firmware versions' shows
'Application Load ID' P003G302 and 'Boot Load ID' PC030301
Ray
Christopher
I have tried the suggestions received but no joy :o( can I assume this
is now 'an expensive paper weight'?
I originally wrote:
Was in the process of upgrading a 7960 to SIP and in advertently applied
a skinny image (P003G302.bin), now no matter what i put in OS79XX.TXT
and OS7960.TXT it simply
Title: Message
I have tried numerous versions and other
skinny images, the phone takes OS79XX.TXT from my TFTP server (seemingly ignores
it) then asks for SEPMACaddr.cnf and SEPDefault.cnf, despite whats in
OS79XX.TXT.
%-- SNIP! --%
what version of SIP are you trying to
It's not asking for SEPMACaddr.cnf.xml but SEPMACaddr.cnf?
Anyway created SEPMACaddr.cnf with the following lines -
# Image Version
image_version: P0S3-03-0-00
# Proxy Server...etc..etc..etc..
My TFTP server receives the error - Timeout error sending
SIPmacaddr.cnf?
I modified the
It's not asking for SEPMACaddr.cnf.xml but SEPMACaddr.cnf
Anyway put the lines below in my SEPMACaddr.cnf and no joy :o(
My TFTP server reports either -
Timeout error sending SEP00115C0F70F4.cnf to 192.168.10.63
Or
Transmit error while sending to 192.168.10.63 : The connection is reset
by remote
Title: Message
Sorry meant SEP not SIP!
I have tried P0S3-05-3-00 and P0S3-06-3-00
in the formats below in the SEPMACaddr.cnf file -
# Image Versionimage_version:
P0S3-0x-3-00
and just -
P0S3-0x-3-00
The phone takesthe files with no
errors but nothing happens as far as upgrades are
Title: Message
Don't have P0S3-04-X-XX.bin, will have to
find a copy!
-Original
Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher
KennaSent: 26 May 2005 15:10To:
asterisk-users@lists.digium.comSubject: RE: [Asterisk-Users] Cisco
7960 Firmware help
Title: Message
Ok hopefully V4 will solve all my problems -
thanks for your time and help!
Ray
Christopher
Kenna 26 May 2005 15:27when i received
all my phones, they all had v3. Once I went to v4.x, i was able to jump to 5
then 7.
-C
[EMAIL PROTECTED] 5/26/2005 10:16 AM
Don't have
I'm sure it would running a sip image, but I stupidly loaded a skinny
image (P003G302.bin) which is where this all started from and can't seem
to move anywhere :O(
Maybe it is requesting an xml file but not from my TFTP server (port
69).
Ray
-Original Message-
From: [EMAIL PROTECTED]
Title: Message
Ok tried P0S3-04-0-00 in the in the formats
below in the SEPMACaddr.cnf file -
# Image Versionimage_version:
P0S3-04-0-00
and just -
P0S3-04-0-00
The phone takesthe files with no
errors but nothing happens as far as upgrades are
concerned?
Razza26 May 2005 15:43Ok
for? Does it
ask for OS79XX.txt?
Razza wrote:
I'm sure it would running a sip image, but I stupidly loaded a skinny
image (P003G302.bin) which is where this all started from and can't
seem to move anywhere :O( Maybe it is requesting an xml file but not
from my TFTP server (port 69).
Ray
Title: Message
'Status messages' simply shows
SEPMACaddr.cnf
for info, 'Firmware versions' shows
'Application Load ID' P003G302 and 'Boot Load ID' PC030301
Ray
Christopher
Kenna26 May 2005 18:30in the phone
under status, what are the error messages?
-C
[EMAIL PROTECTED] 5/26/2005
Title: Message
Was in the process
of upgrading a 7960 to SIP and in advertently applied a skinny image
(P003G302.bin), now no matter what i put in OS79XX.TXT and OS7960.TXT it simply
wont upgrade. The phone is pulling down OS79XX.TXT from my TFTP server but then
goes on to repetedly ask for
Discussion
Subject: Re: [Asterisk-Users] Cisco 7960 Firmware help please.
On Wed, 2005-05-25 at 18:34 +0100, Razza wrote:
Was in the process of upgrading a 7960 to SIP and in advertently
applied a skinny image (P003G302.bin), now no matter what i put in
OS79XX.TXT and OS7960.TXT it simply wont
Patrick Worte:
The problem is indeed unique to the TDM400 FXO daughter board. I can
confirm that the X100P and clones do correctly detect hangup on the BT
network, but are plagued by echo problems due the impedance mismatch
with the UK phone network.
Ok I might be beingd really dumb here, but if
Title: Message
Hi all, looking for
some advice on a good FXO solution for Asterisk, living in the UK brings some
issues (as Asterisk and associated hardwareappears to begeared to
the US/Canadian market) such as calling line ID, impedance settings and echo
cancellation (as a result of the
Title: Message
This
maybe an out of place comment but it would appear Digium show little to no
interest in non-North Americanimplementations, do we know if they are ever
going to resolve this issue? or indeed how much it would cost? Based on my
experience I'm sure there are a number of UK
Title: Message
Got a
feeling that the same card I bought from goods2world.co.uk, which gave me
terrible echo problems, due to the impedance mismatch of the US telco network
(600ohm) versus the BT network.
When
using that card all seemed to disconnect fine, I assume the issue lies with the
Subject: Re: [Asterisk-Users] Good FXO for UK use.
Razza wrote:
Got a feeling that the same card I bought from goods2world.co.uk,
which gave me terrible echo problems, due to the impedance mismatch of
the US telco network (600ohm) versus the BT network.
I think Echo's one of those oddities
: [Asterisk-Users] Good FXO for UK use.
I find the voice quality fine on the 3000. Have you set the line Z etc
correctly and used ulaw?
Chris
- Original Message -
From: Razza [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, April 26, 2005 12:36 PM
Subject
] On Behalf Of Ed
Greenberg
Sent: 16 April 2005 14:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Sipura 3000 FXO with Asterisk
Hi Razza,
I don't know what country you are in, or what your country's telephone
numbers look like, but it seems from your
Greg,
Have you checked the 'PSTN Dialing Delay:' setting under the 'PSTN Line'
tab, I suggest this is at least 1 (second), just to let things
stabalise.
% -- SNIP -- %
Greg Wrote:
Lucky you, my spa-3000 likes to dial 911. So far the local cops have
been nice about it though. (my mobile
]
% -- SNIP --- %
[101]
;PSTN
type=friend
regexten=101
username=983
secret=razza
context=sip_home
port=5080
host=dynamic
nat=no
canreinvite=no
disallow=all
;allow=alaw
allow=ulaw
[budget1]
type=friend
regexten=105
username=budget1
secret=razza
context=sip_home
callerid=Kitchen 105
host=dynamic
nat
Pete wrote:
The comments about it being an ugly hack arent really correct. The
Sipura is really built for standalone useage wiht a sip provider
however it does work well with asterisk.
Follow this thread
http://voxilla.com/forum-viewtopic-t-1335.html
it works and it works **VERY** well :-)
Sipura Support said when I asked the same question.
The SPA-3000 product is very feature rich and has configureable
parameters, including caller id settings. The default configuration is
set for usage in US, but with few adjustments -you can use the device
anywhere. For usage in UK, try
Chris Blake wrote :
-%-
If anyone can help I`ll send the call file to you, or is it ok to
clutter the list with it ?
-%-
'Clutter' the list I'd be interested and at least it is pertinent to *
;o)
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Subject: Re: [Asterisk-Users] mini atx and asterisk (EPIA and the like)
See my comments inline
Razza wrote:
I run mandrake 9.2, one FXO (x100p clone), 5 sip phones, MusicOnHold,
voicemail, etc. off my EPIA Classic/5000 with 512MB memory (I know 512
is totally OTT but had a spare SD stick lying
Title: Message
Is there a
knownproblem with the latest CVS and the UK CLID patches or have new
commands been introduced to asterisk around musiconhold?
I have just updated
my machine following the instructions at h##p://www.lusyn.com/asterisk/patches.htmland
MusicOnHold stops workingwhen I
Thanks to Dave Cotton, Adam Goryachev, Bob Goddard, Doug Lytle and Craig
Guy for their help on this, I think I am going to revert to Mandrake 9.2
and try the CAPI route again - my annual leave is running out!
Ray
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