> Hi,
> I am using a 30 VIP phone and a 12 SP+ phone with
> Asterisk. When I complete a call outside through the
> ZAP device, the phone does not go back to dial tone,
> even after I hang up. The line gets disconnected as
> per Asterisk console. But the phone stays in the same
> state like it
On Jul 18, 2005, at 7:25 PM, Bill Wesson wrote:
Hello all,
Is there an advantage to purchasing a VOIP hard phone with a
display for use
with Asterisk versus a hard phone without a display? Are there some
creative
uses of the screen or is it pretty generic?
Would anyone have a URL referen
Here is a snippet from my remote voicemail application where a user
needs to enter a code which is then matched against the db
;
exten => s,1,Wait(1)
exten => s,2,Answer()
exten => s,3,NoOp(${CALLERID}) ;just so I can see who
called, may wish to save sometime
;exten => s,4,noop()
ex
On Jul 21, 2005, at 5:50 PM, Paul Belanger wrote:
Ryan Williams wrote:
I understand how CID works and how you must set CID when dialing
out on
a PRI and how the phone company sets the name.
I was wondering how this works in regards to inbound calls. I have
a pri
and I get the number t
It would be helpful to capture a complete ISDN call setup.
On the cli type "pri debug span 1"
Then place a call and turn off debug with "pri no debug span 1"
You will then have a complete listing of the signalling between your
co and your * for this time period.
Good Luck
On Jul 22, 2005,
Was this an exmple of your incomplete calls?
From the trace it appears that you issued the disconnect while the
call was in process.
On Jul 22, 2005, at 9:32 AM, JOAO CARLOS MOURA wrote:
My debug
Thank you for help.
Verbosity is at least 5
-- Accepting AUTHENTICATED call from
>
Perhaps a sip debug peer will shed some light? Never had an issue
like this myself and I am installing 25 more today so I hope not;-)
On Aug 2, 2005, at 7:10 AM, Rich Adamson wrote:
Hello everyone, I have just received 3 brand new Polycom
SoundPoint IP
600 from "voisupply.com" and I have
You are attempting to put the second registration on the backup
server for the first registration. Move farthur down the file and
setit up on the second line. Works great. My IP600 on my desk has 3
different registrations right now to different servers. Great debug
tool.
On Aug 2, 2005, a
Sounds like the two devices are not agreeing on the signalling.
double check the channel bank settings against the zaptel ones. You
did run ztcfg after changing the zaptel file right? zttool should be
helpful here.
On Aug 2, 2005, at 1:11 PM, David Sampson wrote:
Hello –
I have a Premi
Yes you can install an asterisk server to feed a legend. You may use
either em/did trunks or PRI depending on your software level. I quit
installing Legends several years ago but di just start feeding a
customers from asterisk via a T1 configured as PRI on both ends. Net
on * and cpe on Le
yes it sounds very familier. I would suggest not tweaking settings
for gain on a T1, it should work fine at 0db.
I got tired of chasing down these issues with our/other carriers
because the software echo cancellor on the system is just not capable
of performing this task. I suspect the new
I do not ftp the file to this phone instead I use the web interface
as I constantly make changes. If you are putting your registration
infto phone1.cfg then here is a sample with multiple registrations
for 3 seperate entries with 2 line appearance each. The server info
in this case is in si
Right track, but it can be simplified even more
exten => 720,1,macro(sipexten,${EXTEN})
On Aug 4, 2005, at 5:25 PM, Tarpo, Louie wrote:
We handled it by creating a macro which dials the exten, then sends
the call to voicemail.
You could create it where each extension is handled seperately
Have to answer your inbound call first - I suspect
On Aug 4, 2005, at 5:28 PM, Tim King wrote:
[macro-dialout-trunk]
exten => s,1,GotoIf($[foo${ARG3} = foo]?3:2)) ; arg3 is pattern
password
exten => s,2,Authenticate(${ARG3})
exten => s,3,Macro(record-enable,${CALLERIDNUM},OUT)
exten =>
Easy to send a call to two phones
[exten-context=here]
exten => 100,1,Dial(sip/user1&sip/user2)
sip.cfg
[user1]
...
...
[user2]
...
...
However you will need to transfer the call from one user to another
if the wrong party picks up
On Aug 3, 2005, at 3:31 AM, Dean Collins wrote:
Kevin
Actually Adits are pretty much self configuring for fxs anyway - I
can install and only have to set ip info
On Aug 2, 2005, at 6:38 PM, Doug Lytle wrote:
David Sampson wrote:
Hello –
I have a Premisys Slimline Channel Bank connected to a Digium
TE110P. I am not able to call the FXS ext
l 720 as there is not other
extension
defined. As a result, all calls would go to 720. ${EXTEN} would
always be
720.
I don't follow your logic.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jj
Sent: Thursday, August 04, 2005 4:00 PM
To: Asterisk
Probably complaining about the dialed number. You say you are dialing
the pstn - and I assume in north america.
What is the number 91713545 supposed to dial? Last time I checked
pstn calls were either 7 or 10/11 digits.
perhaps you forgot to strip the 9 off?
Perhaps the pstn is returning an e
Each button on a Polycom can monitor an individual mailbox. The light
will light if any line has a message, an envelope icon will appear
next to the button with the message. to retrieve use the message
button and it will then let you select the line you wish to retrieve.
Very easy.
On Aug
Use NI2 anytime it is availabel. It will deliver calling name. NI1
will only deliver calling number. Also most COs will support NI2 with
no tweaks much better than NI1 or any of the others.
NI1 was created to solve configuration issues between systems. did a
pretty good job. But as new feat
Did you try it?
On Aug 9, 2005, at 10:25 AM, Deon wrote:
I have an ATA186, a tech just told me to set UID0 and UID1 to the same
username, and PASS0 and PASS1 to the same password. In my mind, this
would seem to have the unit registering twice under the same account,
which Asterisk wouldn't sup
What does pri debug span 1 show?
On Aug 9, 2005, at 5:02 PM, Panitaxx wrote:
Hello,
I have an ISDN PRI E1. For some reason I am not receiving the did
number so every call can only go to s exten. I have tried using _X.
exten. Also I have immediate=no in zapata.conf. Any hint?
thanks in advance
lling q931_hangup, ourstate Active, peerstate
Active
Protocol Discriminator: Q.931 (8) len=9
Call Ref: len= 2 (reference 32840/0x8048) (Terminator)
Message type: DISCONNECT (69)
[08 02 81 90]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the l
sip show register will display the sip registrations the server has
performed to other peers, not other peers to it
this is also true for iax
not sure why you split the registrations into 2 instead of using
friend, friend works fine for me and I have not heard of any issues
of using it for
Hi everyone, i need some advice, i have 3 offices with
4 telephone line in each using a normal telephone
system( rather old systems )all connected via a
clarkconnect vpn (broadband). so i have been looking
at asterisk and like all the features so i was
wondering if i could get all the telephone ca
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