On Thu, 19 Jun 2008 11:36:27 +0200, Vincent
<[EMAIL PROTECTED]> wrote:
>Will do, although it could be a problem in the Zaptel code, which is
>not written by the mfg. Thanks.
I also notice that I can't restart the driver:
# /usr/local/etc/rc.d/zaptel restart
zaptelkldunload: can't unload file: De
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:
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 -0
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
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
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"
Sent: Thursday, June 19, 2008 6:41:06 PM GMT -05:00 US/Canada Eastern
Subj
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, Activ
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, 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 s
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'v
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
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 no
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/1-2/5
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 head
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
_
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]
[mailto:
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 (p
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 (dynamic)
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 iss
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 n
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]
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
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 Prim
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 TE22
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 pa
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
loadzo
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
> extens
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 t
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 e
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
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 futu
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
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 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 _somethin
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 comi
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
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 c
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)
>>
>> http://www.voip-info.org/tiki-index.php?page=Ast
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 PROT
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 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
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 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, 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
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 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
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