try sipgate.co.uk
M
On Sun, Mar 2, 2008 at 8:21 PM, Mike [EMAIL PROTECTED] wrote:
hey Folks,
Just curious if anyone has suggestions on how one can get a near
FREE(I hope) DID number.
I am experimenting with asterisk, for home use.
thanks,
Mike, my friend, you have hit the nail on the head - and thanks for the
support - it's good to know I'm not alone with this issue.
I am working with 4 customer callcentre sites to resolve this problem. 3
sites are in Melbourne (Aus) and one in Auckland. The Auckland site is
dialing international
I am trying to automatically detect disconnected numbers when using the
outbound dialer I have written.
* Some numbers hang up immediately with a Cause Code 0 and no voice
treatment
* Some numbers get voice treatment with a PROGRESS indication and an
associated Cause Code 0
* Some numbers get
Does anyone on list have experience with Digium hardware in the following
servers:
Dell poweredge SC440
IBM xSeries x226
Have just had major hassles getting TE205P ISDN cards going in these boxes.
No joy so far.
Anyone managed to do it yet?
Thanks.
Mark
Probably find you have less hassle ditching the proprietary PBX's altogether
and just use the * boxes at each end of an IAX trunk. Probably be a cheaper
solution in the long run.
On 11/30/06, asterisk-robert [EMAIL PROTECTED] wrote:
Inital setup for testing will be 2-4 channels in order to
in there
though...
many thanks,
Mark.
On 12/1/06, Joe Dennick [EMAIL PROTECTED] wrote:
I've got a Dell SC440 running just fine with a Digium TDM-400 card in
it. It's running CentOS-64bit.
Mark Edwards wrote:
Does anyone on list have experience with Digium hardware in the following
servers:
Dell
have any suggestions or pointers here?
Cheers,
Mark.
Mark Edwards
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Are you seeing any IRQ misses
Cat /proc/zap/1 and let us know.
You might be experiencing some interrupt conflict
.
M
-Original Message-
From: David Gagnon
[mailto:[EMAIL PROTECTED]
Sent: Friday, 22 September 2006
3:37 PM
To: 'Asterisk Users Mailing List -
September 2006 6:56 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fax Detection on outbound call
On Fri, Sep 22, 2006 at 05:17:03PM +1000, Mark Edwards wrote:
I'm trying to configure my asterisk server to detect fax on an outbound
ZAP
call. The reason for this is that I have
, Mark Edwards wrote:
I'm trying to configure my asterisk server to detect fax on an outbound
ZAP
call. The reason for this is that I have a bunch of interviewers in an
outbound callcentre who don't like listening to fax machines and I want to
be able to detect fax on the outbound leg before
cat /proc/zaptel/1are you seeing any IRQ misses?are you also seeing any HDLC errors in your asterisk debug log?Mark.On 9/8/06, Xue Liangliang
[EMAIL PROTECTED] wrote:
Hi, I just installed a TE110P ina supermicro server with Intel 945Gchipset,the customer reportedthe system has random drop
I'm running both these cards in different boxes. MB chipset is the
same. I see consistently zero IRQ misses on the 205p and 9/10 timing
samples at 100%.
the te110p is a different story. No amount of optimisation can get the
timing up to 100% consistently. 1 out of 15 zttest samples will come
Joseph, I'm getting exactly the same issue as you. Periodically, I get
CHANUNAVAIL back from the PRI span and then we fail over to VoIP. There is
plenty capacity spare on the span.
If you get any further with figuring this out, please let me know. I'll do
likewise.
Cheers,
Mark
-Original
Yes. Me.
I don't have a fix unfortunately - like you I seek one, however I have had a
better experience by far though with the new 102x firmware branch.
I would definitely recommend it to you.
Mark
-Original Message-
From: Gareth Blades [mailto:[EMAIL PROTECTED]
Sent: Monday, 10
Hi,I have a single PRI span setup at present and need to dial a prefix number in order to suppress outgoing caller ID.I am about to have a second PRI Span set up on the same server, but I want to bring both spans into the one group. The second span will be from a different telco.
I want to
Try dtmfmode=info and see if that works.
Mark
-Original Message-
From: Dovid Bender
[mailto:[EMAIL PROTECTED]
Sent: Thursday, 9 March 2006 6:08
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] No DTMF
Some one was on my server
making changes to my
Youll want to check the docco
against the SetGroup and CheckGroup applications,
although I think these have been deprecated in favour of a variable
type approach now.
Regards,
Mark
-Original Message-
From: Marc Archer
[mailto:[EMAIL PROTECTED]
Sent: Friday, 3 March 2006
I had a similar issue here in Aus where I
was chasing crossover cables around. Eventually the cows actually did come home
and I called up the telco. They rebuilt (or reinitialized) the
ISDN service and everything worked a treat from there on in. Took a couple of
days to get to this point.
Ah! There you go - I knew Chuck Norris had something to do with it...
;-)
Mark
-Original Message-
From: Alexander Burke [mailto:[EMAIL PROTECTED]
Sent: Monday, 20 February 2006 11:17 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Asterisk on Solaris 10 (AMD
Hey Alex,
Please forgive the question, but what is the rationale behind using Solaris
over Linux as an asterisk hosting platform?
Cheers,
Mark
-Original Message-
From: Alexander Burke [mailto:[EMAIL PROTECTED]
Sent: Monday, 20 February 2006 3:45 PM
To:
First impressions telling me you want to check your phone settings. What
phone are you using and what are the config settings?
Mark
-Original Message-
From: Tomislav Parèina [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 14 February 2006 9:01 PM
To: Asterisk Users Mailing List -
Keep a watch here as I believe there is a bug brewing...http://bugs.digium.com/view.php?id=6147cheers,MOn 10/02/06,
Bartosz Jozwiak [EMAIL PROTECTED] wrote:
I have some more info about my deadlocks...It usually happens when you have a callwaiting and users are pressing flashbuttonon ZAP
, is this 4 parties you want to connect up? If so,
then I think the only way to do this is to use a meeting room as you will effectively
have a conference call between 4 parties.
Cheers,
Mark Edwards
www.switchnet.com.au
-Original Message-
From: Wai Wu [mailto:[EMAIL PROTECTED]
Sent
,
Mark.
Mark Edwards
http://www.switchnet.com.au/
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Hi.
I think it is something to do with your priindication= setting.
I may be wrong, but if I was you I would try priindication=inband
Cheers,
Mark
-Original Message-
From: Adam Rybak [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 6 December 2005 9:01 AM
To: asterisk-users@lists.digium.com
Firstly, try unplugging span 1 and plug it into spans 2,3,4 in turn to
check the port.
Secondly, try E1 crossover cables in the remaining red-alarm spans.
Regards,
Mark
-Original Message-
From: Diyanat Ali [mailto:[EMAIL PROTECTED]
Sent: Sunday, 25 December 2005 8:08 PM
To:
Hi.First off, the illegal instruction doesn't look at all pretty.The best way to start a new installation is to start asterisk thus:/usr/sbin/asterisk -vcwhen you get a clean start with no fatal or significant errors, you can close it out and then start it as a daemon
/usr/sbin/asteriskOnly then
Simple.GSM is a voice codec. It is not designed in any way to compress music. It is _only_ designed to work with voice. You will therefore never be able to hear good quality music over a GSM codec. No matter how much you try.
sorry.MarkOn 11/21/05, Christoph Rothe [EMAIL PROTECTED] wrote:
dell power edge, without usb i read some
thing about asterisk needing the usb for something internal
is this true,
will i have to change the hardware or add a usb card...
thanks again Mark
Phil
Mark Edwards wrote:
Phil.
You won't have much luck achieving this. The closest you
Title: Message
Can I
suggest a quick review of http://voip-info.org as the majority of your
questions will be answered with the information contained on this
site.
cheers,
Mark.
-Original Message-From: ram
[mailto:[EMAIL PROTECTED] Sent: Friday, 25 November 2005 4:54
Phil.
You won't have much luck achieving this. The closest you will be able to
get is to set the intel voice modem up as an FXO port. Was this what you
meant?
If so, take a look on the voip-info.org site as there is a page there
that will point you in the right direction.
Title: Message
Are
you also implementing the "ping" keepalive as part of your
app?
Mark
-Original Message-From: Bill Michaelson
[mailto:[EMAIL PROTECTED] Sent: Thursday, 24 November 2005 9:09
AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Re: manager
I've got an HT486 on my network which was configured to send DTMF over
RTP. _Was_ being the operative word because post my recent upgrade to
1.2, RFC2833 for DTMF just stopped working!
Works fine on my GXP2000, but no longer on the HT486. Got it going again
by configuring sip info mode.
Hi.
I'm having a problem building zaptel in rc2.
Seems to complete process, but lots of warnings...
[EMAIL PROTECTED] zaptel-1.2.0-rc2]# make
/lib/modules/2.6.11ac7/build
make -C /lib/modules/2.6.11ac7/build SUBDIRS=/usr/src/asterisk-1.2rc2/zaptel-1.2.0-rc2 modules
make[1]: Entering directory
You need to be using firmware 1.0.1.12 on the GXP2000
There is a known issue with feedback/echo on the GXP2000 with earlier
versions. It was fixed with .12 firmware and works fine.
Well mine does anyway...
Cheers,
Mark
-Original Message-
From: Shawn Iverson [mailto:[EMAIL PROTECTED]
:14 PM, Mark Edwards wrote:
I would recommend vorbis speex for this.
You can get windows drivers to read speex files directly.
Vorbis are the same bunch that develops ogg.
Ogg and mp3 are more suited to music rather than speech.
Speex is a much better fit for speech archiving.
Mark
Please clarify,
are you referring to a motherboard with an intel chipset or an Intel Motherboard?
Am currently putting together some desktop machines to run * on that are Asus, but with an intel chipset.
If you could clarify this, it would be a great help.
ta
MarkOn 11/9/05, George Pajari
I would recommend vorbis speex for this.
You can get windows drivers to read speex files directly.
Vorbis are the same bunch that develops ogg.
Ogg and mp3 are more suited to music rather than speech.
Speex is a much better fit for speech archiving.
Mark
-Original Message-
From: BJ
Title: Message
do you
mind posting your dialplan and your zapata.conf for this incoming
line?
Mark
-Original Message-From: Gary Li
[mailto:[EMAIL PROTECTED] Sent: Thursday, 3 November 2005
5:22 PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Response
This indicates that 602 is a dynamic host. It must therefore register
with the pbx so that the pbx knows where to send data.
In this state it is unregistered so it will be unlikely you can call it.
Regards,
Mark
-Original Message-
From: Ronald Wiplinger [mailto:[EMAIL PROTECTED]
Sent:
to drop it
programmatically.
If anyone could point me in the direction of some additional docco that
might help nudge me in the right direction, I would be very grateful.
cheers,
Mark Edwards.
http://www.switchnet.com.au
___
--Bandwidth and Colocation
in connecting the call, I want to drop it programmatically.
If anyone could point me in the direction of some additional docco that might
help nudge me in the right direction, I would be very grateful.
cheers,
Mark Edwards.
___
--Bandwidth and Colocation
Before I relate the actual problem, some context.
Callcentre environment, a few users testing a new digital dialer...
1. Agents are using Grandstream ATA HT486 and a small analogue dialpad with a headset.
2. SIP connection to Asterisk-1.2b1
3. IAX2 connection to ITSP provider.
The call is
number, next call.
I'll go chat to my ITSP next week.
cheers,
Mark.
On 10/18/05, Kris Boutilier [EMAIL PROTECTED] wrote:
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
]On Behalf Of Mark Edwards Sent: Monday, October 17, 2005 3:28 PM To: Asterisk Users Mailing List
Can you explain a little why you would want to do this?
MarkOn 10/3/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Since the search engine on voip-info.org is not working correctly with old
links, etc..I was curious if there is some hidden talent in the IAX2 outbound dialing?What I'm asking about
when dial via zap channel.Does anyone know how to solve it?thanks.2005/9/15, Mark Edwards [EMAIL PROTECTED]: Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan
Hi.
I, personally, would start this process by checking out a clean
version of CVS-HEAD and seeing if the problem still exists. If this is
too much of a jump all at once, at _least_ go to 1.0.9-Stable.
cheers,
MarkOn 9/20/05, Ka Lun Chan [EMAIL PROTECTED] wrote:
Hi All,
My Asterisk Server
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang up.
Am using the following dialplan macro to dial out.
[macro-advdial]
exten = s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds
Hi.
I'm dialling two numbers - one that's unobtainable, one that's busy.
${DIALSTATUS} is coming back ANSWER each time right before the channels hang up.
Am using the following dialplan macro to dial out.
[macro-advdial]
exten = s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum
Or is it a monitoring application that you need? for instance, do you
need the ability to monitor active channels on request? The
description below isn't clear around what you mean in regard to
'monitorin' and 'placing the others on hold'. Normally you 'place
someone on hold' after you have
concur that the best way around this is to perioically restart.
FWIW this is my restart script which I invoke from cron in the middle
of the night...
#!/bin/bash
ASTERISK=/usr/sbin/asterisk
RMMOD=/sbin/rmmod
MODPROBE=/sbin/modprobe
ZTCFG=/sbin/ztcfg
echo Stopping
$ASTERISK -rx stop when
speex is a codec.
it's not a network protocol or a service.
you need to be looking to be providing QOS for RTP data, over which
the speex encoded data is sent.
cheers,
Mark
On 8/8/05, Adam Robins [EMAIL PROTECTED] wrote:
Can anyone out there please tell me what ports Speex uses? I want to
set
http://bugs.digium.com/view.php?id=4631
FYI
I am experiencing the same, but due to lack of cooperation from ITSP
am not able to proceed with debugging it.
Feel free to pursue...
regards,
mark
On 8/8/05, Justin Richards [EMAIL PROTECTED] wrote:
I too have been having inbound dtmf problems
what does your voicemail.conf and sip.conf look like?
Mark
On 7/25/05, Mauro Zanin [EMAIL PROTECTED] wrote:
Hi everybody,
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
update your CVS - recompile and try again
cheers,
Mark
On 7/25/05, Alessio Focardi [EMAIL PROTECTED] wrote:
Hi,
I get this message after password request in voicemail app:
Unable to create lock file: No such file or directory
Anyone got a clue about fixing that problem ?
I can't
OK Angus
just start here
mv extensions.conf extensions.conf.old
and create a new extensions.conf
[default]
exten = _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten = _2XX,2,Hangup
just those 3 lines
do an 'extensions reload' in the CLI or just restart Asterisk
and see if it works
regards,
PS you would be better seeing this debugged with set verbose 5 in the CLI
regards,
Mark
On 7/25/05, Mark Edwards [EMAIL PROTECTED] wrote:
OK Angus
just start here
mv extensions.conf extensions.conf.old
and create a new extensions.conf
[default]
exten = _2XX,1,Dial(SIP/${EXTEN
Do you have module versioning support turned on when you compile the kernel?
On 7/23/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi List,
When I update my kernel the modules for my wildcard fail to load. When I
revert back they work fine. They only seem to work with the kernel I
can you give a little more detail here?
I assume * is sending the SIP messages. Where is is sending them. To the ITSP or the PHONE?
What does the SIP message contain? Is is MWI? is it 'OPTIONS' keepalive?
Try posting a Sip Debug output from * that contains the dialogue in question
cheers
Mark
Can we assume the following?
1. You have eliminated the possibility of a card fault (i.e. tested all ports on all cards in both a single and a dual configuration)
2. you have confirmed the support of the MB for PCI 2.2 in all slots
3. You have removed any other cards in the bus (that it is
Yes!
is Vonage SIP or IAX Terminated? I am experiencing the exact same issue and I have logged a bug
http://bugs.digium.com/view.php?id=4631
How helpful are Vonage? Unfortunately my provider in question has been somewhat unwilling to assist in debugging the problem (oztell). A few questions to
yup.
had exactly the same problem as you. been spending the last hour trying to figure out what I did wrong in my config.
guess how I fixed it?
cd /usr/src/asterisk
cvs update
make install
simple really! ;-)
I guess someone posted a bugfix a few mins ago and I just picked it up! ;-)
cheers,
Here is a log from a recent call made out on a ZAP channel from a SIP phone inside my network.
For some reason, CDR is billing time even though the busy tone was detected.
It's also logging the call as ANSWERED.
Is this normal behavior? Seems a little odd to me.
I have this as the first 3 lines
:
In article [EMAIL PROTECTED]
,Mark Edwards [EMAIL PROTECTED] wrote: Hi TonyI am having a similar issue to you - from the 'other' direction in that when I connect to * via IAX2 the DTMF is being ignored. I am running HEAD at
the moment.(and for the benefit of another subscriber so that they don't have
Thanks mate - I had my voicemail context set up wrong
cheers - works a treat for me too! ;-)
Mark
On 7/11/05, Peter Bowyer [EMAIL PROTECTED] wrote:
On 10/07/05, Mark Edwards [EMAIL PROTECTED] wrote:
anyone managed to get MWI going on the GXP-2000 with * CVS-HEAD? I have set up the mailbox
Hi Tony
I am having a similar issue to you - from the 'other' direction in that when I connect to * via IAX2 the DTMF is being ignored. I am running HEAD at the moment.
(and for the benefit of another subscriber so that they don't have to invoke their autoresponder I acknowlege that DTMF is
anyone managed to get MWI going on the GXP-2000 with * CVS-HEAD? I have set up the mailbox in the sip.conf entries but no flashing lights... SIP NOTIFY seems to be being sent out...-- regards,
Mark P. EdwardsFWD: 667917
___
Asterisk-Users mailing list
Hi.
I have the following line in the default context of all my internal extensions:
exten = 9876,1,Transfer(125)
When I dial extension 9876 from any sip phone, * dutifully transferrs it to extension 125, which is just what I want.
Unfortunately when I dial 9786 from my Zap connected analogue
Unless I'm very much mistaken you want to get rid of either the host=dynamic or the defaultip=something
host=dynamic indicates the device is getting an IP from dhcp and it will tell * what it is when it registers.
defaultip=something indicates that the device is staticip.
Devices like this are
Thanks guys - appreciate the comments. I understand that IAX does not support inband dtmf, but I still can't fathom why 9 times out of 10 my * box is ignoring DTMF's even though they are showing up in the IAX2 protocol debug output. The really annoying thing is that I can't consistently reproduce
Do you think this might have an impact on http://bugs.digium.com/view.php?id=4631?
Mark
On 7/3/05, Mohit Muthanna [EMAIL PROTECTED] wrote:
Right... that's the one. My mistake.On 7/1/05, Rob Scott [EMAIL PROTECTED]
wrote: I don't find this option in the Makefile. I find RADIO_RELAX which is
I hear you. background is in definitely in use in my extensions.conf here.
Hopefully this partially accounts for the 10% of times when it _does_ work! ;-)Mark
On 7/3/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Saturday 02 July 2005 19:56, Mark Edwards wrote: Thanks guys - appreciate
Hi.
Probably been asked before, but my IAX provider assures me its not their problem
I have a IAX connection to a peer providing a DID. I am dialing up
my number, seeing the DTMF tones come down the line, and the * IVR is
just ignoring them.
IAX debug output is:
Rx-Frame Retry[ No] --
Probably been asked before, but my IAX provider assures me its not their problem
I have a IAX connection to a peer providing a DID. I am dialing up my number, seeing the DTMF tones come down the line, and the * IVR is just ignoring them.
IAX debug output is:
Rx-Frame Retry[ No] -- OSeqno:
Does the registration show up?
try sip show registry at the CLI
also try sip debug peer sip_proxy and post the result.
Might be able to see what's going on there...
mark
On 7/1/05, David [EMAIL PROTECTED] wrote:
Hi,I have been trying to configure my Asterisk to use a Sip provider forout and
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