For those that are interested, I was just out on the Cisco site and
noticed that they had released firmware 7.4 as of March 11th for the
7940/7960 phones.
Doug
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An article posted on the The Register:
http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/
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Joseph wrote:
Anyone try the new Cisco firmware for the 79xx sip phones?
In my test it seems to work fine for a little and than soon the phone
looses its time. At first the status shows clear, and then it appears to
get confused about the ntp time source and the time goes away on it.
I've
Kevin P. Fleming wrote:
Joseph wrote:
I don't have that problem on the 10 or so phones I've updated to 7.4.
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This happened on my 7940
Doug
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[EMAIL PROTECTED] wrote:
change the sntp_mode: from directedbroadcast (the
default) to unicast. This will cause the phone to poll
your NTP server. This solved the problem for me.
This fixed my problem as well! Thanks
Doug
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Eric Wieling wrote:
C F wrote:
Now consider this (this works with the cisco 7960, even if you put a
7914 with it, it will still use all 20+ plus buttons this way, if CW
is disabled on the phone):
I thought the 7914 does not support SIP. Is that incorrect?
The 7914 does not support SIP from what
Tom wrote:
Configuring VLAN 100 seconds
TFTP SIP loads a few seconds
back to Configuring VLAN the rest of the time.
Roughly the same there here as well. 7940 boots faster, but not by much.
Doug
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Andrew Kohlsmith wrote:
Also, carrier access has an incredible support site that they do not charge
for. You do need to register with them but that's free.
Just for a follow up to this statement,
I recently purchased an Adit 600 via eBay, tried to gain access to
Carrier Access's website.
Mohamed Farid wrote:
I did create a Voice Mail Boxes for Some Cisco Phone Sets ,,
I can now record a Voice Mails , and I can hear them ,, but I am not
able to configure the Voice Mail Button on the Phone Sets to directly
listen to the Voice Messages ..
The 7940/7960 series phones, the option is
Chris Lee wrote:
On Mon, 14 Mar 2005 08:06:20 -0800, Scott Laird [EMAIL PROTECTED] wrote:
Has anyone else upgraded to 7.4 and found that the date time no
longer appears on the phone?
Chris,
As someone pointed out earlier, change your sntp_mode to unicast in your
SIPmacaddress.cnf as such:
cmisip wrote:
far asterisk seems to use 1:10 while all other traffic uses 1:102. How
does one packet shape RTP?
Thanks for any help.
# +-+
# | root 1: |
# +-+
# |
# ++
# | class 1:1 |
#
cmisip wrote:
I initially used that script without modification. However, I noticed
that all traffic was going through class 1:102 regardless. Seems as if
all the children of 1:20 are set with a prio of 0 by default even if
1:20 is specifically set to prio of 2. I used
My setup is a
Greg - Cirelle Enterprises wrote:
-- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221
Greg,
Completely unrelated to your current query. Your logs show that your
BT100 is running 1.0.5.20 firmware. Is this correct? The last I knew,
they were at 1.0.5.18
Doug
Erick Perez wrote:
any others not so expensive? does these 3com sip phones work with * ?
http://www.grandstream.com
The BT102 has 2 ethernet ports.
Doug
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Mike Dent wrote:
Whilst on the subject of BT's, do the callers and called buttons function?
they dont seem to do anything on mine?
Yes, but the hand set needs to be off hook.
Doug
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Paul Fielding wrote:
I've seen lots of stuff go around about Grandstream firmware levels
(in my case specifically the BT101/102). I'm just wondering what the
currently accepted 'best' firmware version is to use? After seeing
stuff going around about buggy firmware I want to know what I'm
Just got my (10) BT102 phones, flashed them to 1.0.5.20 and all work.
No duds at all. Not a bad little phone at all.
Doug
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To
Remco Barende wrote:
What would be the best / easiest VPN software solution. I would like
to install vpn software on the * server for roadwarriors to connect to
with laptops running windows. Ideally the vpn solution will not
require any additional software on the client side but will use IPSEC.
to the G3.
Incoming shows the trunk info setup by our phone admin.
Happen to have a link that you could point me to on this setup?
Thanks again!
Doug Lytle
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Pat Delaney wrote:
AS a proof of concept experiment, I want to try and integrate Asterisk
with my Lucent Definity G3 switch. I dont have an available T1 port
on the G3 but I can round up 4 analog ports off the G3. What I thought
I could do is create a Hunt group on the G3. Lets say I configure
Ken Godee wrote:
We have been able to get our Definity G3R working with Asterisk via a
T100P card and a TN767E card, works very well! But, I'm a little
stuck on how to get the DID info from the G3 and ext/ext info to the
G3. Incoming shows the trunk info setup by our phone admin.
With no
Pat Delaney wrote:
Thanks for you comments. I have the one port card now. I plan on
purchasing the TDM400. My only question is whether or not the Dell
optiplex has pci 2.1 (I think)
Depends on the model, check Dells website. A quick googling show:
Specifications: *OptiPlex* GXi. *...*
Mark Johnson wrote:
I also have loaded POS3-07-3-00 and hitting any numbers does nothing.
I am using the default dialplan.xml file and a really basic
SIPxxx.cnf file. This is the same on a couple of phones I am
trying. Any ideas?
Other then, when I did the flash, I only included the
Chris Wade wrote:
Doug Reid - Stormcorp wrote:
To the Dougs,
This is turning into a me too, but my phones, about 25 of them, don't
let me dial without picking up the handset, pressing the speaker
Thats the part I didn't catch before. No, I am pressing the speaker
button first. On my 7912, I
Mark Johnson wrote:
This may be OT, but I can't seem to find how to do this. I have
7940/7960's with Skinny on them. When you start pressing numbers on
the dialpad, you start building a number to dial. When I install SIP,
that functionality goes away. You have to hit the speaker button, or
Jose Cruz (Branders IT) wrote:
But how about the config files (SIP...) that needs to be inside the tftp
server, where can I get a sample of that?
That's where the images for the firmwares of the ip phones come from, on
boot right?
Jose,
Under Mandrake, to install the tftp program is, urpmi
Michiel van Baak wrote:
Hi,
Do you happen to know if those image will work on a cisco 7905g ?
I have chan_sccp now but SIP is what i want to do.
Michael,
SIP040406A is labeled on Cisco's website for the CP-7905G
Doug
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no experience with the OS
whatsoever. 30-40 hours to get basics running for Asterisk at your
current level of experience is not really a good estimate.
I would suggest picking a distro, getting at least comfortable on it's
install before even considering this project.
My 2 cents.
Doug Lytle
Derek Conniffe wrote:
Hi everyone,
I'd say this question has come up and been answered before but I
haven't been able to find it.
I have a Cisco 7940 that I've upgraded to SIP firmware (currently
P0S-3-06-3-00 - for some reason there was a failure when trying to
upgrade to V7 so I left it at
Matt Schulte wrote:
one phone you say? We upgraded all of ours in our office to 7.3 without
a problem.
The 7.3 firmware has 2 files associated with it, P003-07-3-00 and
P0S3-07-3-00, the OS79XX.TXT file didn't have them both listed. I put
mine as such:
[OS79XX.TXT]
P003-07-3-00
Michael J. Tubby B.Sc. wrote:
Gents,
Following from a previous posting I've switched my 7060G at home
from SCCP/Skinny to SIP but am stuck as I can't get it to register with
Asterisk.
Mike,
This is from my configuration with a working 7960 under Asterisk 1.05, I
have a 7940 under 1.03 with
Shaun Ewing wrote:
You need to do an incremental upgrade - eg: SIP 2.3 - SIP 4.4 - SIP 7.3.
The ealier images can be obtained from Cisco if you have a valid CCO login.
Actually, you need to go from earlier version to version 5. This is the
version Cisco started using an encrypted firmware.
Christian Moller wrote:
Hi all,
settings, including the settings password. It was reset back to the
factory default. Well, then I decided to enter my TFTP settings again,
but I can't even do that! I unlock the settings but I can not enter
anything on that screen. I get the Phone unprovisioned
Sascha E. Pollok wrote:
Did you switch DHCP off?
I didn't need to.
Doug
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Noah Miller wrote:
I'll call bullshit on that. I know for a fact that Debian does NOT
allow root logins except from console. Hell Debian isn't allowing
root logins
from X anymore due to the likely hood for you to try and use root for
more than administration.
I'm sure that's true nowadays. I
Michael Di Martino wrote:
What is the unsubscribed address?
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dean collins wrote:
1.0.5.22 is available for downloading here
http://gs-firmware.gratissip.dk/
I dont know why these are available if Grandstream dont update their
webpages to indicate newer versions are available.
Because, it's still in beta. It can be found on Grandstream's website.
Razza wrote:
Trying the get back to position of a running * PBX, I tried to install
the zaptel drivers, using the following process -
CD zaptel.
Make linux26
Make install
When I modprobe zaptel I get the following errors -
[EMAIL PROTECTED] zaptel-1.0.4]# modprobe zaptel
FATAL: Error
Razza wrote:
When I run 'cat /proc/version' I get the following -
Linux version 2.6.8.1-12mdk-i586-up-1GB ([EMAIL PROTECTED])
(gcc version 3.4.1 (Mandrakelinux (Alpha 3.4.1-3mdk)) #1 Fri Oct 1
12:36:44 CEST 2004
I'm running 2.6.8.1-12, but not the 1GB version under an AMD XP 1800+
The
Andreas Sikkema wrote:
[EMAIL PROTECTED] wrote:
(side note: If you havent bought their hardware and are using
Asterisk for free them again you should expect even less
assistance imo)
Right, so I have to buy hardware I don't need?
No, but you have to do research on your own. If, after
Razza wrote:
I have added 66-zaptel.rules file as you suggested...do I not need
to ref this in a conf file or does udev simply look for any '*.rules'
file in /etc/udev/rules.d/ ?
That would be correct, udev will look for them there.
Which to me whould show some form of mismatch, or is this
[EMAIL PROTECTED] wrote:
Any assistance on gettting bi-directional calling going would be great...
We got it working in SIP but it won't register hence calls going to the phones
don't even start..
If the phones are behind a NAT, make sure the option on the phone for
NAT is set to yes.
Doug
Victoria Alexandru wrote:
Since Mandrake cooker (10.2beta3) witch I'm using is using kernel
2.6.10, I wander if I have a chance to install it properly.
All I have are the 3 installation CDs and it looks I'm missing some
packages (bison, and associated -devel, zlib, and associated -devel).
Luca Bariani wrote:
I tried to compile source tarball on my mandrake 10.1 but I got compilation
error, I think because mdk 10.1 has a too newer gcc compiler
Luca,
I run Mandrake 10.1 Official with current updates and compile without issue.
Doug
___
asterisk asterisk wrote:
Hi
I confiured Gasnstream phone 100. Firmware ver:Program--1.0.5.16
Bootloader--1.0.0.21HTML--1.0.0.41 ï VOC--1.0.0.7.
The message button under 1.0.5.16 was broken, go to 1.0.5.18 or newer to
fix.
Doug
Doug Meredith wrote:
Peter Bowyer [EMAIL PROTECTED] wrote:
Indeed - the bug was fixed in 1.0.5.18 (Nov 14 2004) according to the changelog.
Is this a beta version of the firmware? The main download page only
has 1.0.5.16.
Yes, the current is 1.0.5.22, but I've found it to be very
Fred Skrotzki wrote:
Is Fedora Core 3 Supported?
Fred,
I've just installed FC3 on a new box and will be installing Asterisk
today. I've done it a couple times and had no problems with the compile
and install. Just starting to learn *. I haven't gone beyond the
compile/install and play
Craig Guy wrote:
About once an hour the phone displays '403' on the display for about 10
I noticed the same thing.
Doug
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difference.
context = sip
mailbox = 5574
disallow=all
allow=ulaw
allow=alaw
callerid = Doug Lytle 5574
[5558]
type = friend
host = dynamic
auth=md5
username=5558
qualify=300
reinvite=no
canreinvite=no
nat=yes
dtmfmode = rfc2833
context = sip
mailbox = 5558
disallow=all
allow=ulaw
allow=alaw
callerid
Wilson Pickett wrote:
hangup hangs up the channel, thats why you get a busy sound on your phone.
you have to phsically hang up the phone.
Which is basically happens when someone hangs up on you, no? ;)
Not really, no.
On an analog line, after 30 or so seconds, yes this is the norm. On a
**[rooms]
;
; Usage is conf = confno[,pin]
;
conf = 1234
Should be:
conf = 1234,
And in *extension.conf:*
**
exten = 8600,1,Meetme,1234
Should be:
Doug Lytle wrote:
Call extension 8600, enter the pin of and off you.
hahaha, should of read, and off you go.
I'm getting old.
Doug
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Dave Cotton wrote:
I would have:
exten = 8600,1,Meetme(1234,M)
just to have music on hold if there's only one person in the conference.
Very cool! I'll give it a try, thanks!
Doug
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Jeremy SALMON wrote:
I have an PSTN line (X100P) and 10 grandstream budge tone phone.
Jeremy,
Receive call, press flash, call other party, wait for answer, press
transfer, hangup.
I believe that is what I saw on an earlier post.
Doug
___
Jason Lixfeld wrote:
Nov 30 00:43:40 WARNING[138185728]: chan_zap.c:757 zt_open: Unable to
open '/dev/zap/pseudo': Device not configured
Nov 30 00:43:40 ERROR[138185728]: chan_zap.c:6665 chandup: Unable to
dup channel: Device not configured
Nov 30 00:43:40 WARNING[138185728]: app_meetme.c:227
[EMAIL PROTECTED] wrote:
Hello,
I am using Mandrake 10.1.
Howto to install asterisk.
I have downloaded tarball.
I have not installed any hardware yet.
Is it possible to install ?
Yes,
http://www.voip-info.org/wiki-Asterisk
Doug
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Greg - Cirelle Enterprises wrote:
Hi,
Is there an * configuration that will allow the BT100 to
display the numeric callerid instead of the broken
text?
exten = extension,priority,SetCIDNum(${EXTEN})
Doug
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Kim Lux wrote:
I'm new to VOIP. We are thinking of setting up a VOIP system between a
couple remote offices. I've been lurking on this group for a while.
What is the consensus on these phones:
http://www.netvoice.ca/grandstream/budgetone101.htm
Cheap, but useable. I'd go for the 102 model
Stojan Sljivic - Pamet wrote:
Hi all,
I have a problem starting the ztdummy. Here is what happens:
I have used following command to make the ztdummy:
make clean
make linux26
make install
I use Fedora Core 3.
You need to read the udev.README file in the zaptel make directory.
Doug
Brian Capouch wrote:
Derek Conniffe wrote:
It's a zipfile, but it's in a hidden URL on their website with the
phrase BETA in it. I don't have the whole URL saved anywhere, but I
think I got it off the list, so you should be able to find it by an
archive search.
, but they don't list prices.
Thanks,
Doug Lytle
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Ken Godee wrote:
Yes, a TN767E will work and actually a TN464 may not,
depending on how the G3 is setup. If I remember right
the TN464 needed a different clock set up then we had
on our system.
Ken,
I printed that out to give to our phone Admin. Thank you very much!
Doug
Paul Brock wrote:
Gents,
Just a passing thought... is there any reason why the ability to search the
past posts on here isn't switched on?
Paul,
I went to the following link and was able to download the arcives in
mbox format. Pulled them into Mozilla for searching, worked very well:
David Hajek wrote:
Actually, I got the display flashing when I have a new message. But it is
possible to get the Grandstream's Message button working? My goal is to
pickup earphone and press Message button to retrieve my messages.
David,
I have both the message button and the MWI working under
David Hajek wrote:
Doug,
thanks for the info. I'm curious where you get the BETA from? ;-) I sent a
notice to Grandstream support anyway.
http://www.grandstream.com/BETATEST/
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Doug Reid - Stormcorp wrote:
HI
How would I get the MWI working on the Grandsreams?
Thanks
Doug (Yip another one!)
Doug,
Currently, my voicemail is on extension 5700, so under the GS web
interface, under Voice Mail User ID, I put 5700
Now, when pressing the message button, I get the
Ken Godee wrote:
I'm currently playing with a Digium T100P card and 2 Grandstream
phones, things are working well. I wanted to move on to linking our
Definity G3R Rev 8.2 to the T100P. Everything that I've read so far
shows that you need a TN464 to accomplish this. We have a TN767E
Ken Godee wrote:
More than happy to try and share my set up, but
I'll be out of reach to my system and notes till
first of the year.
Thanks for the offer Ken, hopefully we'll get it figured out. Have a
good time, and I'll leave a message as to how it went.
Doug
I just joined this list. I would be interested in this information as well.
Thanks!
Doug Lytle
Damon Estep wrote:
I am trying to determine what hardware/distro combinations others have
had success with.
My goal is to find a reliable entry level server where the manufacturer
supports a Linux
Arun Kumar wrote:
Hi,
Congestion() if no of calls in this group are more then 3. But my
provider says he is not getting any busy signal from my side and he
says for all incoming numbers (30) he is getting back only one number
from asterisk box(4340).
exten = 4340,16,Congestion()
Try
randulo wrote:
On 7/29/07, Don Kelly [EMAIL PROTECTED] wrote:
Note that some of us newbies have posted the same question two or three
times because we didn't see our own post (let alone a reply) in a timely
manner.
True. I could swear that when I post to biz, I get a post
Stephen Bosch wrote:
Doug wrote:
Kewwl! How do you get the .wav files into the Polycom?
If it's not obvious, I'd be interested in this information too.
Most people seem to think you can't change the ringtones on the Polycom
sets.
This is the info I used:
MOSBAH ABDELKADER wrote:
Hello,
Is the OpenVPN the ideal solution to set a tunnel between two asterisk
servers or there is a better solution.
Mosbah,
We use OpenVPN between 3 facilities and Asterisk between them. We have
OpenVPN on their own systems and Asterisk sets behind them.
I don't
Kate Kretz wrote:
OpenVPN is very good in NAT (if one of your boxes is behind NAT).
otherwise, OpenVPN seems to be a bad choice, it's complicated,
non-standard (there'n no RFC on OpenVPN).
It's complicated? No more so then Asterisk.
We use, it was quite straight forward to set up and the
Peder @ NetworkOblivion wrote:
That's great, now say you have 5 or 6 AA's and each one has 10 different
parts that you want to record (thank you for calling... for Steve
press
This is what I do. I found it some place on the wiki, it lets you
record many prompts.
exten =
Gordon Henderson wrote:
On Fri, 17 Aug 2007, Andres Jimenez wrote:
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,VMAuthenticate(${me})
exten =
Gordon Henderson wrote:
On Fri, 17 Aug 2007, Doug Lytle wrote:
XOR in dialplan :)
9 lines of code vs. my 7 (which include a validation) though ;-)
Ooops! Missed that line.
One thing I noted recently is that phones sometimes do weird things with
*'s and #'s )-: The Siemens C460IP
Andres Jimenez wrote:
In the latest version (see below) I added some playback that will say
if the phone is lock or unlock, before and after locking/unlocking it.
Just a note,
You will want to make sure that (911/999) calls are handled properly
when the phone is locked down.
Doug
--
Russ Price wrote:
On my Asterisk installation, I've had to roll back to Zaptel 1.2.19.
When I use 1.2.20, I get very bad echo problems.
You should be trying 1.2.21.1
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
Ed Pastore wrote:
since half the time a new one comes out, a fix for it comes out the
next day.
So... that said, what's a good version to linger on? I don't *need*
Until 1.4 improves, I'm staying with 1.2
I do know that I'm running some version of 1.2, and am also not sure
if I
Tzafrir Cohen wrote:
On Fri, Aug 24, 2007 at 07:33:21AM -0400, Steve Totaro wrote:
stability problems) with 1.0, ahve already migrated to 1.2 or 1.4, and
now swear (by?) 1.2 or 1.4.
My decision based on what I've been reading in the bug tracker and
people commenting on how they've had
Joshua Colp wrote:
I'm going to end this email with a question myself... how many people
have Asterisk on a development/staging server before deployment, test,
and isolate the issues they may have in their specific scenario?
I do, but with limited resources for testing (2 polycoms, no
Eric ManxPower Wieling wrote:
Sangoma cards are complicated to set up, have a history of kernel (and
zaptel) VERSION issues. i.e. It seems like the zaptel or kernel version
I'm running on a machine is always something newer than is supported by
the Sangoma drivers. Never had any issues
shadowym wrote:
Well there are a couple fine examples of FUD if I do say so myself. Just do
a search and see what cards the 'serious' companies out there are using.
Nuff said.
How do you consider that FUD? It's happened to me several times.
Doug
--
Ben Franklin quote:
Those who would
Jerry Geis wrote:
However, if I call in and get connected then a second call comes in
they also get connected.
I was expecting them to get a busy signal or something...
Your dialplan needs to take this into account. I do the following:
;
Christian Peter wrote:
Can anybody help me with this issue. Please no switch to Hylafax
mails, because I'm very happy with SpanDSP, it integrates nicely and
It just show you how many people on this list are pleased with HylaFAX+
Doug
--
Ben Franklin quote:
Those who would give up
Bruce Reeves wrote:
multiple source options, it is worth checking, I had problems with
poor audio quality using the sound card with asterisk.
I did as well using the built-in sound on the motherboard that I was
using, switched to a Sound Blaster Live Value card and that problem went
away.
Mike Hammett wrote:
Pong
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Jeremy P wrote:
Basically I've taken an HP thin client workstation which is all solid
state and loaded Debian and Asterisk on it (well, Asterisk-GUI too,
but just to prove I could make it appliance-worthy). I'd be
interested in any feedback on how to improve it, specifically on how
to
satish patel wrote:
Dear all
I have asterisk 1.4.11 i am new in asterisk i want
to see online call list how it is possible to see how man call
currently active is there any command or tool to see online call ??
from --- to
Flash Operator Panel is what you'd want to look
Ken D'Ambrosio wrote:
- Dial by name. Has anyone made it so it can be first or last?
Yes
- Jump to voicemail; you used to have to actually dial the voicemail,
whereas most voicemail systems allow you to go to your mailbox when you
hear your voice prompt. Any chance this has been
Russell Bryant wrote:
Jeremy Wadhams wrote:
Yep. In fact, it was one of the first patches I ever wrote for Asterisk. :)
And under 1.2 it can be easily bypassed. After the password is changed,
if the user hangs up, the next time they call into the voice mail
system, it doesn't
Russell Bryant wrote:
Doug Lytle wrote:
I'm ... sorry? However, it does behave exactly as is documented. It
specifies
No need to be, I was just making an observation.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety
Brian Alexander wrote:
Have any of you connected two asterisk machines by t1 crossover using
pri_net/pri_cpe signaling? I am completely stumped and would love to
know that some had done this and what their
What does your cable pin in/out look like. I haven't connected two
Asterisk
Brian Alexander wrote:
Thanks for all of the feedback. I really appreciate the help! :)
My cable is
1--4
2--5
4--1
5--2
I believe you need to change your cabling.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
Tilghman Lesher wrote:
Uh, why? That is the correct pinout for T1 crossover.
Then why does mine work? Is it a straight though cable that is needed then?
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
Tilghman Lesher wrote:
A straight-through cable to used for connecting from a QuickJack to a T1
card, but a T1 crossover is needed for card-to-card (or card-to-channel bank,
for that matter).
I have a Sangoma 102u port 1 to port 2.
And, my Micoscanner is showing
1 -- 5
2 -- 4
4 --
Brian Alexander wrote:
At this point I do not think the problem is the wiring. What else
should I try?
Have you confirmed that the failing card is working correctly? Maybe
the card is at fault.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Brian Alexander wrote:
On 9/24/07, *Doug Lytle* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Have you confirmed that the failing card is working correctly? Maybe
the card is at fault.
All of the cards have been confirmed to work by themselves.
The only other suggestion I
Brian Alexander wrote:
On 9/24/07, *Doug Lytle* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
The only other suggestion I have would have would be to use IAX
instead
of PRI for inter-machine communications.
LOL Yeah, normally that is what I would use. Unfortunately
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