Just installed and so far it works fine for a 30VIP,
only issue is the Speed dials
they dial out OK, but the cisco phone doesn't break dial tone even
when the other party has answered.
Do you have any suggestions
Regards
Robb
___
Asterisk-Users
Hi
I have updated from CVS about a week ago and got the externip working
with FWD for outbound calls., but I'm having problems with inbound
calls, I don't think they are even reaching the Asterisk box even though
I have forwaorded 5060 and the rtp range specified, another thing I have
Has any one seen or heard of the lastest developments fo the Farfon IAX
phone? the web site
Thanks
Robb
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even print them 6 times instead of 3..
Ie if it prints
DEBUG: 327
DEBUG: 0
DEBUG: 0
then that ring is 327,0,0 ( but to be safe lets round it to 5's) and make
it 325,0,0 only have to hit within 10 +/- for it to match.
bkw
On Thu, 13 Nov 2003, Robert Boardman wrote:
Thanks again Brian, one
I have been trying to start asterisk all night after a reboot
I keep getting this error scrolling up the screen
ouch: error while writing audio data broken pipe
when I go to another console there are 4 instances of mpg123 running
and when I do TOP they are taking 100% CPU between them
I
I have been trying to start asterisk all night after a reboot
I keep getting this error scrolling up the screen
ouch: error while writing audio data broken pipe
when I go to another console there are 4 instances of mpg123 running
and when I do TOP they are taking 100% CPU between them
I have
Tim Sailer wrote:
On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote:
I have been trying to start asterisk all night after a reboot
I keep getting this error scrolling up the screen
ouch: error while writing audio data broken pipe
when I go to another console there are 4
Hi
Has anyone go the 30VIP phone to work with asterisk?
If so how good us the usability of the Cisco 30VIP phone with asterisk either
using chan_sccp or Chan_skinny?
Thanks for your Help
Robb
--
Robert Boardman
Tel:01617737929
FWD:86263
John Fraizer wrote:
Robert Boardman wrote:
Tim Sailer wrote:
On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote:
I have been trying to start asterisk all night after a reboot
I keep getting this error scrolling up the screen
ouch: error while writing audio data broken pipe
Hi
I have had distinctive ringing working before the patch was applied to
the CVS tree, now it doesn't work,
Could anyone point me in the right direction to debug distinctive ringing?
Thanks
Robb
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Hi All
I have been working in getting 2 Cisco 12 sp phones working, they work
fine over the lan, but I cannot get it to work across the Internet
I only have one way voice
Do es anyone have any advice
Thanks for your help
Robb
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--
Robert Boardman
Tel:01617737929
FWD:82623
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Asterisk
Hi
Just one question
do any of the Digium T1/E1 cards do DPNSS signaling?
Robb
--
Robert Boardman
Tel:01617737929
IAXTel:17007737929
FWD:82623
--
Robert Boardman
Tel:01617737929
IAXTel:17007737929
FWD:82623
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[EMAIL
Steven Critchfield wrote:
On Wed, 2004-03-03 at 06:33, Robert Boardman wrote:
Hi,
With the current CVS as of last night 20:00GMT
I was testing a asterisk with the e100p card using a PRI analyzer to excerise
the 30 channels over and over, just going directly to voice-mail.
Basically, I don't
Hi
Just got a brand new Box Cisco VIP30 off ebay, the standard phone
functions work fine, just a couple of questions,
1) how do I program the other buttons not on the standard keypad part..
2) When I hang up the display doesn't clear and keeps the numbers just
dialed on screen, can this be
Hi
I have an asterisk voicemail system connected directly to a pri, there
are no extensions connected to the asterisk box,
anyway my questions is, can I get asterisk to call an associated phone
number when a voicemail box has a message?
Thanks in advance for your help
Robb
I wonder if anyone could post a how-to for the chan_sccp, I've
downloaded and compiled the code, but I don't know where to go from here,
any help would be appreciated
Thanks
Robb
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In the UK we have a service that if you dial 1471, the last 6 calls are
read out to you and you can pick which one you want by pressing 3,
this means that 1471 shows in the cdr, has anyone created a script or an
application that will read out the last callers and then dial the
number? ( that
Hi Just recieved the above phone
Does anyone have sip.conf and extension.conf example for the SIP phone working
with the FXS w100p and the FXO tdm400d
any help would be appreciated
Thanks
Robb
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Hi
Firstly could I thnk everyone who has helped me so far,
I just have a couple of queries
I have not had chance to debug this much yet
but When using the tdm40p all extesions busy themselves out, and * cannot rint
the extensions for incoming calls
is this because I don't have a hangup
Hi All
when I modprobe the wcfxs drive and do a cat /proc/pci, it is sharing irq with
my AGP and USB, I think this is causing the card to stop working, it would work
for a couple of days or a couple of hours but then stop, I'm a complete linux
newbie, how can I force the wxfxs driver onto
Hi
My Motherboard cannot disable the IRQ sharing, can I specify on with modprobe,
the IRQ to be used with a particular module?
Robb
Quoting Emanuele Pucciarelli [EMAIL PROTECTED]:
On Mon, Jun 16, 2003 at 12:05:34PM +0100, Robert Boardman wrote:
for a couple of days or a couple of hours
I want to limit my sons phone useage, by setting a 30min limit on out going
calls from his room
is there a simple way of doing this with asterisk?
Thanks for your help
Robb
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When I was looking for a psu the only site I found for 5V ( and a decent price)
was CPC, but I cannot remember the www address
robb
Quoting WipeOut . [EMAIL PROTECTED]:
Hi,
Quick question to all the electronics gurus out there..
I unpacked my second GS phone yesterday (had it for about
Hi
I have looked through the wiki and search the mailing list, but I cannot
find a way to intrude on a call, can asterisk do this feature?
if so how?
Thanks for your help
Robb
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I have an ISDN TA that has 2 POTS interfases (FXS), can these be used
with asterisk?
Thanks in advance
Robb
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Hi
I really need caller id to work in the UK, I understand that the X100p
uses a US chipset,two questions
1) is that a product that converts UK to US caller id in line
or
2) would it be possible to have modem that supports CID in parallel
with the line and the x100p.The modem reads the line
Hi
I have a problem with a new DECT phone I have bought
The key pad works like a Mobile phone where you dial first then pick up
the line, but it seems to dail too fast or spuriously, ie 012826736464
show on thew Asterisk console as 0012282677, could any one offer advice
how to fix?
Also when
Hi
I've just signedup for Distinctive ringing on my PSTN line in the UK, could
anyone explain what I need to add in the conf files to detect and route based
on in comming Distinctive ringing
Thanks in advance for your help
Robb
___
Asterisk-Users
Does asterisk know when each ring comes in or just the first ring, ie
so the cadence can be worked out? say over two rings?
Robb
Martin Pycko wrote:
The X100P together with asterisk does not support the distinctive ringing
detection on the line. Asterisk however can generate the distinctive
Hi
I'm having a problem where the recall button doesn't work
If i press recall before I dial numbers it disconnects me which is what
I would expect, but during a conversation if I want to transfer the TDM
400 just ignores the recall
Any advice would be gratefully received
Thanks
Robb
with asterisk)
line 792
#define ZT_MAXPULSETIME (150 * 8)
I moved it to (20 * 8)
be sure not to set it under ZT_MINPULSETIME, that's (15 * 8)
Matteo
Il ven, 2003-09-19 alle 22:04, Robert Boardman ha scritto:
Hi
I'm having a problem where the recall button doesn't work
If i press recall before I dial
Could someone point me in the right direction for setting up the mysql
cdr function
Thanks
robb
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Thanks that worked
Robb
Brancaleoni Matteo wrote:
mmh... have you enabled
threewaycalling = yes
transfer = yes
in zapata.conf ?
matteo.
Il sab, 2003-09-20 alle 10:51, Robert Boardman ha scritto:
Thanks for the advice Matteo but it didn't work, anthink else I may of
missed?
Robb
How would I compile asterisk for the Athlon XP arch, would there be any
advantage doing this?
Thanks for your Help
Robb
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Thanks for your reply
The system is a small installation but I was thinking about optimizations and
wondered if there would be any particular benifit
anyway thanks for the reply, your comments are very useful
robb
Quoting Alastair Maw [EMAIL PROTECTED]:
Robert Boardman wrote:
How would I
As you my be aware the X100p cannot collect uk caller id,
now I have a modem and a perl script that creates a
file /etc/asterisk/callerid.txt on each incoming call in the format
number,date,time,name
over writing each time a new call comes in
I can asterisk read this file and populate the
Hi Dave
I've not completed the script yet,
But you may not like this but I've had to use a win98 box for the zoom 3025C
(important its the C model), the zoom modem is the only internal one I've found
that can do uk caller id (but its not supported on the linux driver), and is
still
I have two questions about incomming ring and extension ringing
1) When an incoming call is detected by asterisk it takes 2 or three
rings before the internal phone ring does anyone know how I can fix this?
2) All internal phone ring on an incoming pstn call but after the call
is answer all the
Hi I'm trying to install but I think I have a problem!!!
Would I be correct in saying if I don't have the jp graph libs, the
links on the form would be followed but nothing would be displayed
Areski wrote:
I made an Update, now don't need register_globals on anymore...
By the way, I fix some
Yes php sysinfo say gd is complied inb
any other clues?
Robb
Areski [EMAIL PROTECTED] said:
Do your php support GD ?
You can simply check it with a phpinfo !
More info about gd (configuration, installation) :
http://www.php.net/image
On Wed, 2004-03-24 at 21:12, Robert Boardman wrote
, 2004-03-24 at 21:12, Robert Boardman wrote:
Hi I'm trying to install but I think I have a problem!!!
Would I be correct in saying if I don't have the jp graph libs, the
links on the form would be followed but nothing would be displayed
Areski wrote:
I made an Update
Hi,
I trying to get agi with perl to stream a gsm file , and wait for a
digit , the agi gets to the stream but doesn't play back, could some one
explain how this works
here is a snip it of code
open(DAT,/etc/asterisk/1571.log) || die(Cannot Open File);
while( $sth-fetch() ) {
print DAT in
Hi
I'm trying to compile the Zaptel Drivers, but I seem to be getting an error
zaptel.c:131: warning: data definition has no type or storage class
zaptel.c:132: error: parse error before
config_must_be_included_before_module
zaptel.c:132: warning: type defaults to `int' in declaration of
Hi Victor
I'm currently working in a Linux Distro, it is being internal alpha
testing by my self and a couple of me my colleagues, over the next
couple of weeks I'm hoping to release a beta version to the asterisk
community., I'll keep you posted via asterisk users, about its features
as it
Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box?
Thanks in advance
Robb
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First of all thanks for the patch it works great,
but i think it breaks the distinctive ringing,
I have 2 incoming numbers in one x100p in contexts home1 and home2 but
'default' is always chosen has anyone else seen this?
if you need any more info just ask
Robb
Tony Hoyle wrote:
David J Carter
change it. There are probably unneeded lines
above.
Regards
Peter
-Original Message-
From: Robert Boardman [mailto:[EMAIL PROTECTED]
Sent: 09 October 2004 21:40
To: Whisker, Peter
Subject: bt communicator`
Hi Peter
I have been following your post but didn't see the other emails about
Cesc Santa wrote:
Hi,
I am trying to use an Avaya 4602 phone, which I just updated from a
very old SIP software to the latest I could find on avaya's site
(032207). The upgrade went fine and it gets registered on the Asterisk
server.
Now, a couple of glitches, though.
- The phone's web
Hi All
I'm having problems posting to this list, no bounces the mails just
dont show
any advice how to get the postings through is there filtering?
robb
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asterisk-users mailing
me too
being a patch newbie how do you apply the patch
and
are the three comma seperated values equivalent to the dron and drof on the modems?
I ask because the dron and droff, using my modem arent always say 5, sometimes there 4
Robb
--- Original Message ---
From: John Vozza [EMAIL
Thanks for your help Brian
how would you come the the values required for the distincive ring?
Robb
Brian West wrote:
cd /usr/src
patch -p0 file.diff
bkw
On Thu, 13 Nov 2003, Robert Boardman wrote:
me too
being a patch newbie how do you apply the patch
and
are the three comma
it 325,0,0 only have to hit within 10 +/- for it to match.
bkw
On Thu, 13 Nov 2003, Robert Boardman wrote:
Thanks again Brian, one more question if i may ( soory for the hand holding)
I've added the line below should the information show up when I am in
asterisk gc,
what do I have to do
will this port sort out UK caller id?
--- Original Message ---
From: Mark Spencer [EMAIL PROTECTED]
Sent: Wed, 19 Nov 2003 17:58:01 -0600 (CST)
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 4 Port FXO cards
We *are* making progress, and i have a running prototype, however the
production
Hi All
Maybe this would be a beter solution, but you may have to buy directly from them
http://www.artech.com.tw/html/gx100e/gx100e.htm
Robb
--- Original Message ---
From: David Luyens [EMAIL PROTECTED]
Sent: Mon, 24 Nov 2003 14:14:10 +0100
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Hi
I'm trying to configur a grandstream BT101 to connect to asterisk, both
behind different NATs, I realise that a double Nat is a problem, I have
tried using fwd forwarding to iaxtel as a solution but cannt seem to
get them to connect as I think there is a codec problem as IAXTEL
doesn't
Peer Oliver schmidt wrote:
Nicolas Bougues wrote:
On Wed, Jan 07, 2004 at 01:42:30PM +, Andreas Anderson wrote:
Hi Guys,
is there a client which can be used on the SonyEricsson P800/P900...?
IAX would be cool, but i take anything that can connect (via bluetooth)
to an asterisk-server ;-).
Hi,
I have a load of files recorded with MixMonitor that are out of sync ie
one leg of the call is 2-3 seconds behind the other,
is this a bug in Asterisk 1.4.18, or am I possibly doing something wrong
Is it possible to edit the file and re sync the a/b leg?
Thanks for your help
Robb
Hi All
I'm having problems with outboud ISDN calls,
They setup OK , and ring the other end OK, but when the call is answered
I get a disconnect cuase 17 with an error message in the console of
[Apr 15 08:06:13] DEBUG[4361] chan_zap.c: Found empty available channel 0/31
[Apr 15 08:06:13]
I have been trying to sort this out for a while now but with no luck
I have isdn - asterisk- pabx on a te205
incoming calls work fine
outgoing calls seem to work fine but the call is dropped when answered
I think it is to do with the line
[May 8 17:51:55] WARNING[4762] channel.c: Unexpected
Hi
I'm trying to get the status of an extension that has DND set using the
service code, or trying to disable the service codes altogether so that
I can do them in the dialplan if needed
any advice wout be appriciated
Thanks
Robb
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Olivier wrote:
2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hi,
1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP
it is mentioned MWI is now working.
In my testings with lastest 02123 firmware, MWI is blinking when
missed calls but
Olivier wrote:
2008/10/5 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Kevin P. Fleming wrote:
Olivier wrote:
2. R Hook-flash key is now available to transfer calls.
In s450IP web management server, its defaults
Hi
I have a requirement to bridge Digital ISDN call through an asterisk box
but no matter what I setup in the dial plan the second leg of the zap
bridge is always set to Transfer Capability of SPEECH, I wondered if any
one has come across this and managed to fix it?
Thanks in advance for
yes and it is still set to speech
I've even tried to port the old patch here
http://bugs.digium.com/view.php?id=6251 to the system with no luck
robb
Melcon Moraes wrote:
Have you tried:
exten = s,n,SetTransferCapability(DIGITAL)
?
[]'s
MM
-Original Message-
From: robert
seen the bearer capability in asterisk or is the call nat
working? I've seen that a digital call shows up as speech.
You are using Zap? Or are you using mISDN? Cause there you have to set
an extra parameter in the dial statement.
chris...
robert boardman schrieb:
yes and it is still set
Krishna Sumanth Chava wrote:
Hi * Users,
I ran into a problem when I was trying to communicate an avaya IP
Office talk to asterisk with SIP Trunking. I had successful calls from
asterisk to Avaya but not from avaya to asterisk.
Can someone provide me insight on how to address it or
the Asterisk and avaya talking to each other.
Thanks
Krishna
On Fri, Nov 7, 2008 at 2:59 PM, Robert Boardman [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Krishna Sumanth Chava wrote:
Hi * Users,
I ran into a problem when I was trying to communicate an avaya
Sriram wrote:
Hi
below are my configs:
pstn(e1)---asterisk (span1)-legacy pbx(connected via
span2)- legacy pbx analog extensions.
my dial plan is like callers dial into asterisk(span1) , hear an IVR
option and they are connected to the agents via the legacy pbx (which
is in
Hi All
Just been looking at stats for one of my sites, and I'm conserned about
the number of error cause codes being returned from the telco
for example
12000 calls processed
131 are cause code 31* normal. unspecified.*
139 are cause code 28 * invalid number format (address incomplete).*
-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Boardman
Sent: Thursday, November 20, 2008 4:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN Cause codes
Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Boardman
Sent: Thursday, November 20, 2008 5:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN Cause codes
Hi All
Just been
of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Boardman
Sent: Friday, November 21, 2008 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi All
I cannot seem to find a way to stop atserisk inercepting DTMF tones and
regenerating them even on a zap to zap bridged call
is this possible?
Thanks
Robb
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zap channel on one card to zap channel on another
Robb
Alex Balashov wrote:
You mean a zap-to-zap call hairpinned into the same adaptor, or another one?
Robert Boardman wrote:
Hi All
I cannot seem to find a way to stop atserisk inercepting DTMF tones and
regenerating them even
thanks
Found that but sometimes I need to detect dtmf ie when playing back a
recording
Robb
Philipp Kempgen wrote:
Robert Boardman schrieb:
I cannot seem to find a way to stop atserisk inercepting DTMF tones and
regenerating them even on a zap to zap bridged call
is this possible
Hi All
I Have an ISDN 30 circuit passing through an asterisk box to a legacy
pbx, all is working well but I have had a problem that modems do not
work, I thought of turning off echo cancelation but I cann t seem to
find the ial switch do do it, could someone point me in the right
direction to
Hi all
Been messing about with the single port cards for a number of years, but
never got good results, I was thinking of giving them another go over
Christmas and was wondering if anyone would share there recent
experience, as to which driver works best MISDN BRISTUFF etc with the
latest
Here is just one example of a warning when compiling asterisk on Ubuntu 8.10
manager.c:1760: warning: ignoring return value of âreadâ, declared with
attribute warn_unused_result
is this anything to worry about?
can i safely ignore it?
Thanks
Robb
On 04/02/2009 00:24, Mark Michelson wrote:
Robert Boardman wrote:
Here is just one example of a warning when compiling asterisk on Ubuntu 8.10
manager.c:1760: warning: ignoring return value of âreadâ, declared with
attribute warn_unused_result
is this anything to worry about?
can i
Hi
Had an issue today where all channels connected to the telco when dialed
returned
WARNING[15366] chan_zap.c: Call specified, but not found?
in the logs,
when I removed the isdn cable and reinserted everything was fine
any ideas?
software Versions
asterisk-1.4.21.2
zaptel-1.4.12.1
Jon Morgan wrote:
Hi All,
We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge
calls, as follows:
ISDN Provider --- Span 1(pri_cpe) --- Span 2(pri_net) Phone
System
The company that looks after our internal phone system can no longer dial in
using their data
Hi
trying to record calls using mixmonitor, but I'm having problems with call
quality
the call seems OK but then it drops frames with silence ( for less than 0.5
seconds) then call continues
All I'm doing is bridging two zap channels and recording no transcoding or
changes to the channels
Hi All
BT are providing a SIP gateway for PSTN through the BT communicator with
Yahoo Messenger, I have done an ethereal trace and found that the BT
Communicator side of the software is using SIP, so in theory I could add
more PSTN lines to Asterisk for BT using SIP, but I am having problems
gARetH baBB wrote:
On Fri, 20 Aug 2004, Robert Boardman wrote:
BT are providing a SIP gateway for PSTN through the BT communicator with
Yahoo Messenger, I have done an ethereal trace and found that the BT
Communicator side of the software is using SIP, so in theory I could add
more PSTN
should this work with the x101p? or just the tdm400?
Thanks for your help
Robb
Edward Eastman wrote:
Brilliant - thanks, took me half an hour but it's working now.
Just for the record, settings as follows:
The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009
.
Dan
On Fri, 2004-09-10 at 17:38, Robert Boardman wrote:
should this work with the x101p? or just the tdm400?
Thanks for your help
Robb
Edward Eastman wrote:
Brilliant - thanks, took me half an hour but it's working now.
Just for the record, settings as follows:
The patch on http
Just done this for a client using an E1 Pri card in the avaya box and a
sangoma a102, using qsig , works fine, I wouls recommend this to any
oneits been up and stable for two months now
Regards
Robb
housi mueller wrote:
The main goal is that any extension from the Avaya PBX can make long
hi vincent,
In the UK you can have multiple pots lines with the same telephone
number. but you would need more fxo lines for this.
Regards
Robb
Vincent Li wrote:
Hi Lists,
I have one box with two FXO and two FXS ports, it is running asterisk
inside. I have one sinle POTS line connected
Hi All
I have been asked if it is possible for an external application to be
aware of the position of the playbcak of a file with control playback
ie a file is playing and the user hits the fast forward button , is
there a manager event that show how far into the file it has been played?
Tzafrir Cohen wrote:
On Wed, Jan 02, 2008 at 07:41:40PM +, Russell Brown wrote:
Don't you just hate it when something was working and when you come to
use it in anger it's broken :-(
Something in the, fairly, recent series of Asterisk updates has broken
DIGITAL call passthrough.
Tzafrir Cohen wrote:
On Thu, Jan 03, 2008 at 12:24:38AM +, robert boardman wrote:
I have an outstanding problem with this,I have found that if you set
overlapdial to no on the internal leg ie connected to the pabx it works,
but you will have to set the pabx to dial en-block ie send
Hi
I have had a Centos 5 installed with asterisk and zaptel for a couple of
weeks, I had to reboot eh machine today, and when it rebooted it got
stuck at Starting udev if I remove thew tdm400 it boots OK, but no zaptel
has anyone seen this , and can offer any advice?
Thanks Robb
Tzafrir Cohen wrote:
On Sun, Jan 13, 2008 at 05:33:58PM +, robert boardman wrote:
Hi
I have had a Centos 5 installed with asterisk and zaptel for a couple of
weeks, I had to reboot eh machine today, and when it rebooted it got
stuck at Starting udev if I remove thew tdm400 it boots
thanks for the reply
I'm already on 1.4.7.1
regards
Robb
Ed Nunez wrote:
I had the same issue and updated my Zaptel drivers to version 1.4.17 and
it's rebooting fine now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of robert
boardman
Sent
Nearly got an SPA922 phone working behind a NAT,
the phone registers, and I can dial out and have two way speech,
on an incoming call the SPA922 rings
I answer and the SPA922 shows Anwsering but never does and the far end
continues ringing until the voicemail answers,
this then show as a
and/or you’re timing out. Also sip.conf and user.conf would be helpful as
well as Asterisk release.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *robert boardman
*Sent:* Friday, August 07, 2009 9:01 AM
Do you have to set aside kines for the data channels or can you have dynamic
data channels, for example ISDN dialup internet backup?
Robb
2009/9/1 Tim Nelson tnel...@rockbochs.com
- Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote:
On Monday 31 August 2009 21:59:28 Tim Nelson
Hi All
I having an intermittent problem with the above mobile gateway and would
appriciate some advice
basically 1 in 10 calls fail at some point during the call, the duration of
the calls ate completely different
call progression
Call comes in from Zap channel and dials a mobile number on the
I Have a home line connected to a tdm400p with 3 extensions and a siemens
sip-dect , it seems to work fine but during a call there is always a digital
squeal every so often does anyone know what this could be?
Robb
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