Hello !
I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18
The things work up to 80%, I can transfer the call by BLF button and I can
see the green (free) status and red (busy) status.
What I cannot do is to accept the call when someone rings a remote
extension. The BLF
On Wed, 18 Jun 2008, Mark Hamilton wrote:
Hi,
I have a website where customers enter their phone numbers to be called. I'd
like them to have to put in information and 'schedule' a call.
1) Call Immediately
2) Call in the next _ minutes
3) Call me tomorrow, same time.
Jan Prunk wrote:
Hello !
I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18
The things work up to 80%, I can transfer the call by BLF button and I
can see the green (free) status and red (busy) status.
What I cannot do is to accept the call when someone rings a
On Thu, 19 Jun 2008, Jan Prunk wrote:
Hello !
I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18
The things work up to 80%, I can transfer the call by BLF button and I can
see the green (free) status and red (busy) status.
Firstly, make sure the GS phones are of a
On Wed, Jun 18, 2008 at 08:21:06PM -0400, OCG Technical Support wrote:
I've tried a few approaches to making the multimedia keys on my kbd play
nice with myth, but all have lead to dead ends.
One such dead end is to post this question to the Asteris Users mailing
list, I guess :-(
Wrong list?
On Thu, Jun 19, 2008 at 09:22:04AM +0100, Gordon Henderson wrote:
Reading the replies so-far... Cron jobs, databases, shell scripts... Ye
Gods... Try reading the manual (or at least the wiki)
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Scroll down to the bit
On Wed, Jun 18, 2008 at 02:02:28PM -0700, Robert McNaught wrote:
Trying to install zaptel-1.4.11 on a Supermicro SuperServer with Centos
5, and getting the following error trail on make. Googling the issue
has found one user who tried:
seems that commenting out typedef int bool; in
19 jun 2008 kl. 00.34 skrev Tom Browning:
To send calls into a custom SER implementation, I need to be able to
add some items to the URI that Asterisk will then send as part of
the INVITE
Asterisk dial SIP/[EMAIL PROTECTED]
needs to become
Asterisk dial SIP/[EMAIL
On Wed, 18 Jun 2008 12:47:04 -0500, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Please call the reseller from which you bought the card or the manufacturer
for support.
Will do, although it could be a problem in the Zaptel code, which is
not written by the mfg. Thanks.
On Thu, 19 Jun 2008, Tzafrir Cohen wrote:
On Thu, Jun 19, 2008 at 09:22:04AM +0100, Gordon Henderson wrote:
Reading the replies so-far... Cron jobs, databases, shell scripts... Ye
Gods... Try reading the manual (or at least the wiki)
Try adding you context in that the phone is subscribed to. I had some issue
with this because if you do not specify the context it defaults to “default”
and has trouble finding the phone correctly. If you watch your debug very
closely I you should see it try to pick the phone up in the wrong
On Wed, Jun 18, 2008 at 02:02:28PM -0700, Robert McNaught wrote:
Trying to install zaptel-1.4.11 on a Supermicro SuperServer with Centos
5, and getting the following error trail on make. Googling the issue
has found one user who tried:
seems that commenting out typedef int bool; in
LOL, I agree, it _did_ sound a little complicated than to just schedule a
call in the future. I apologize for not being able to find this on the wiki
earlier when I searched.
The other cron jobs and everything probably bring _something_ to the table.
I wonder what.
Either way, please keep 'em
On Thursday 19 June 2008 07:57:07 Mark Hamilton wrote:
LOL, I agree, it _did_ sound a little complicated than to just schedule a
call in the future. I apologize for not being able to find this on the wiki
earlier when I searched.
The other cron jobs and everything probably bring _something_
Hello Gordon,
On Thu, 19 Jun 2008, Jan Prunk wrote:
Hello !
I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18
The things work up to 80%, I can transfer the call by BLF button and I can
see the green (free) status and red (busy) status.
Firstly, make sure the GS
On Thu, Jun 19, 2008 at 08:05:59AM -0500, Tilghman Lesher wrote:
On Thursday 19 June 2008 07:57:07 Mark Hamilton wrote:
LOL, I agree, it _did_ sound a little complicated than to just schedule a
call in the future. I apologize for not being able to find this on the wiki
earlier when I
On Thu, Jun 19, 2008 at 9:57 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Jun 19, 2008 at 08:05:59AM -0500, Tilghman Lesher wrote:
On Thursday 19 June 2008 07:57:07 Mark Hamilton wrote:
LOL, I agree, it _did_ sound a little complicated than to just schedule a
call in the future. I
Is there any reason that the SIP INVITE URL shouldn't conform to the
same syntax as RFC3986 standard URLs
( http://en.wikipedia.org/wiki/URI_scheme#Generic_syntax ), as specific
to SIP according to RFCs 3969 and 3261? That would be, according to
Hi Asterisk Users,
my apologizes for cross posting.
I'm trying to make the next scenario in Asterisk DialPlan: Alice calls Bob,
Asterisk executes Dial application with G(context^exten^pri), after that Bob
answers the call, Asterisk transfers Alice to pri, Bob to pri+1. It should
be possible for
In article [EMAIL PROTECTED],
Alexander Olekhnovich [EMAIL PROTECTED] wrote:
I'm trying to make the next scenario in Asterisk DialPlan: Alice calls Bob,
Asterisk executes Dial application with G(context^exten^pri), after that Bob
answers the call, Asterisk transfers Alice to pri, Bob to
On Thu, 19 Jun 2008, Jan Prunk wrote:
You might want to try:
exten = _**.,1,Pickup(${EXTEN:2})
exten = _**.,n,Hangup()
Ok I have tried adding these 2 lines, and the error which I get when calling
01 5863165, which then rings extension 65, and I try to accept the call on
extension 70 by
I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1
crossover, and I'm currently stuck. Anyone have any thoughts on what I
can do to get past this?
Asterisk side
Digium TE220B w/ green LED (using port 2)
Zaptel.conf
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
Try underscore _ rather than dash -
Thanks,
Steve T
On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole
[EMAIL PROTECTED] wrote:
I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1
crossover, and I'm currently stuck. Anyone have any thoughts on what I can
do to get past
try pri_cpe instead of pri-cpe
On Thursday 19 June 2008 12:51, Eve-Ellen Cole wrote:
I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1
crossover, and I'm currently stuck. Anyone have any thoughts on what I
can do to get past this?
Asterisk side
Digium TE220B w/
The underscore helped, but didn't resolve the real issue. Now I get the
following messages:
[Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think
we're the CPE, but they think they're the CPE too.
[Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available!
Using
pri_net usually when connecting to a legacy system.
Thanks,
Steve T
On Thu, Jun 19, 2008 at 1:38 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote:
The underscore helped, but didn't resolve the real issue. Now I get the
following messages:
[Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on
Agreed. It looks like you've tried to tell the Avaya to be the network
side but it doesn't seem to be acting like the network. Do what Steve
suggested and see if you get a different result...
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
On Wed, Jun 18, 2008 at 05:27:04PM -0500, Tilghman Lesher wrote:
Here are the details:
If caller enters only three digits/letters:
Jane Smith, Extension 123, If this is the person you are looking for...
If the caller types in more than three letters, the person's name is not
spoken,
This will happen if the other side is configured the same as the
Asterisk side. i.e. PRI CPU mode on both ends or PRI NET mode on both
ends. This can also happen if the line is in loopback mode at the far end.
Eve-Ellen Cole wrote:
The underscore helped, but didn't resolve the real
Just about 30 minutes that I can´t get real information from my Asterisk
box. All agents seem to be available but is not true:
QUEUE_01 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4,
C:0, A:0, SL:0.0% within 0s
Members:
Local/[EMAIL PROTECTED]/n with penalty 1
Hi all,
I need to setup call using particular timeslot on one of E1's. I've
looked into
http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and
it says that:
exten = TestTrakt,1,Dial(ZAP/1-2/517255333)
exten = TestTrakt,2,hangup
should work and force call setup via span 1
Right again, getting a little closer (babysteps) ... no alarms on either
system, but when I check the pri status in the CLI, I get PRI span 2/0:
Provisioned, Down, Active. I've searched for clues, but am not coming up
with the next step.
-Original Message-
From: [EMAIL PROTECTED]
You'll probably need to turn on pri debugging for this span and then
capture the output from when you connect the T1 cable. That might yield
some clues, like whether or not any activity is happening on the
d-channel and if so, if there are any errors that might tell you what's
going on.
-MC
List,
Could anybody speak to the status of development in 1.6 branch? I
know support for SIP over TCP is pretty new / experimental but it
seems active development of it has slowed or stopped in recent months.
Is that a correct statement? Is SIP over TCP more a community project
or something
In article [EMAIL PROTECTED],
Marcin J. Kowalczyk [EMAIL PROTECTED] wrote:
I need to setup call using particular timeslot on one of E1's. I've
looked into
http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and
it says that:
exten = TestTrakt,1,Dial(ZAP/1-2/517255333)
On Thu, 19 Jun 2008, Eve-Ellen Cole wrote:
Right again, getting a little closer (babysteps) ... no alarms on either
system, but when I check the pri status in the CLI, I get PRI span 2/0:
Provisioned, Down, Active. I've searched for clues, but am not coming up
with the next step.
It's not
On Thursday 19 June 2008 13:38:05 Jay R. Ashworth wrote:
On Wed, Jun 18, 2008 at 05:27:04PM -0500, Tilghman Lesher wrote:
Annoying that people aren't following the directions and only entering
3 digits, but we've had some high level meetings here with a string of
clients coming through
On Thu, Jun 19, 2008 at 03:49:01PM -0500, Tilghman Lesher wrote:
On Thursday 19 June 2008 13:38:05 Jay R. Ashworth wrote:
On Wed, Jun 18, 2008 at 05:27:04PM -0500, Tilghman Lesher wrote:
Annoying that people aren't following the directions and only entering
3 digits, but we've had some
On Thu, 2008-06-19 at 15:50 -0400, Paul Belanger wrote:
List,
Could anybody speak to the status of development in 1.6 branch? I
know support for SIP over TCP is pretty new / experimental but it
seems active development of it has slowed or stopped in recent months.
Is that a correct
On Thu, Jun 19, 2008 at 10:06 PM, Chento Arohuanca [EMAIL PROTECTED] wrote:
Just about 30 minutes that I can´t get real information from my Asterisk
box. All agents seem to be available but is not true:
QUEUE_01 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4,
C:0, A:0,
On Thu, Jun 19, 2008 at 4:11 PM, Steve Edwards
[EMAIL PROTECTED] wrote:
On Thu, 19 Jun 2008, Eve-Ellen Cole wrote:
Right again, getting a little closer (babysteps) ... no alarms on either
system, but when I check the pri status in the CLI, I get PRI span 2/0:
Provisioned, Down, Active. I've
The d-channel on the Avaya would be 01C1424. The rest of 01C14 would be the
b-channels.
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, June 19, 2008 6:41:06 PM GMT
On Thu, Jun 19, 2008 at 6:41 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Thu, Jun 19, 2008 at 4:11 PM, Steve Edwards
[EMAIL PROTECTED] wrote:
On Thu, 19 Jun 2008, Eve-Ellen Cole wrote:
Right again, getting a little closer (babysteps) ... no alarms on either
system, but when I check the pri
Primary d-channel set to 01C14. Why doesn't it say 01C1424 then?
On Thu, Jun 19, 2008 at 7:48 PM, Eve-Ellen [EMAIL PROTECTED] wrote:
The d-channel on the Avaya would be 01C1424. The rest of 01C14 would be the
b-channels.
___
-- Bandwidth and
Wrong listsorry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: June 19, 2008 4:38 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Mapping multimedia keys: pressed key not
recognized
On Wed, Jun 18, 2008 at 08:21:06PM
All,
I've put a new asterisk server at another location and can't seem to get
it working. What's the best strategy to debug connections?
I'm doing inbound SIP only and have installed the server in the same way
as I did on my DEV server. Running an nmap on localhost shows the port
listening:
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