Javier Chia ha scritto:
The phone is now logged in but can´t place nor receive
calls. It keeps giving Busy tone when I try to dial a
number.
it does happen when there are no matching extensions for the number you
are dialing
internal context is ok?
you can dial just internal context
Hi,
does anybody has working this konfiguration? For me app_rxfax start receiving,
fax start sending, but after few seconds at begining of the page it stop with
error 400.
My HW PBX configuration is:
ISDN PRI - AVAYA S8300 - H.323 channel - * with app_rxfax
My extensions.conf is:
'7406211'
Does anyone have any ideas? Any magic words to give to the people at
McLeod to get this running?
You might ask the carrier to take a careful look at the mapping for the
d-channel in their DACS equipment and perhaps even ask them to try
re-mapping it for you. If that does not get things moving,
Hi, all
Sorry for not exactly on-topic question
I got Polycom SP300 phones. Somehow they did not come with software. I will
call them on Monday, but in the meantime, I would like to get them going.
I need Polycom configuration template files (phone.cfg, sip.cfg and whatever
else they supply).
Hi,I met the same problem as this mail,
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg101451.html
***Hello,
I want to recieve the output from astmanproxy in a php script.
Is that possible ?
I made a simple php script:
PRE
?php
i dont how to edit the time for ringing 3ms to
4ms when it displayed on console Nobody picked
up in 3 ms and its very short time for ringing .
please if anyone can help me do it please.
Sell on Yahoo! Auctions
Tzafrir Cohen wrote:
On Fri, Jul 08, 2005 at 11:12:37PM +0200, Zoltan Szecsei wrote:
Is this how the modprobes are supposed to respond??
gl0:/home/zls # modprobe zaptel
gl0:/home/zls # lsmod | grep z
Module Size Used by
zaptel239620 0
crc_ccitt
http://www.freedomphones.net/polycom/files/
On Sat, 2005-07-09 at 17:03 +1000, Rudolf Ladyzhenskii wrote:
Hi, all
Sorry for not exactly on-topic question
I got Polycom SP300 phones. Somehow they did not come with software. I will
call them on Monday, but in the meantime, I would like
Hi,
I'm using AMP and its dialparties.agi as most important script in system.
I'd like to port configuration to more embedded system, where I don't have
Perl available.
So I'd like to implement dialparties.agi functionality as closest as
possible with dialplan language.
Are there any
wassim darwish wrote:
i dont how to edit the time for ringing 3ms to
4ms when it displayed on console Nobody picked
up in 3 ms and its very short time for ringing .
please if anyone can help me do it please.
This is now a joke, right?
B.
Hi,
I'm not sure if DTMF is convenient solution for user that has cellular on
his ear
Regards,
Rob.
- Original Message -
From: Dean Collins [EMAIL PROTECTED]
To: Ed Greenberg [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
On 09/07/05, wassim darwish [EMAIL PROTECTED] wrote:
i dont how to edit the time for ringing 3ms to
4ms when it displayed on console Nobody picked
up in 3 ms and its very short time for ringing .
please if anyone can help me do it please.
Didn't any of the 5 answers you got to
Hello,
I finally arrived to convince a cellsocket for Nokia phones to work with
a X101P card in an asterisk v1.0.7.
The problem I have now is that cellsocket usually resets after receiving
a call in the mobile. If asterisk by luck notices it, it issues an error
message Detected Alarm on
- Original Message -
From: Richard Koch [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, July 08, 2005 4:38 PM
Subject: [Asterisk-Users] Speech Recognition
Ed,
Check this out:
http://turnkey-solution.com/asterisk-sphinx.html
That got me up in running in no
Still looking for cheaper (under $250,-) alternative to cisco 7940 with
features needed for corporate use, mainly:
- shared phone book (e.g. via LDAP or XML browser in phone)
- in-line power
- missed/dialed/received numbers
- integrated switch (voice VLAN support)
I found only aastara/sayson
Thanks, Mark. I've changed several of my old Set entries, but
totally spaced out on that one. Done now though. :)
The new format is:
exten = _1NX,1,Set(CALLERID(number)=4025551212|a)
exten = _1NX,2,Set(CALLERID(name)=NPI|a)
Thanks,
Rudolf
- Original Message -
From: Scott Kamp [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, July 10, 2005 1:00 AM
Subject: Re: [Asterisk-Users] Polycom SP300 config files
I am running Asterisk 1.0.9 with chan_bluetooth (kernel 2.6.11)
I have managed to get the module loaded and it connects to my phone and
dials (Nokia 6310i) okay, but once the call connects I hear no sound on
either end.
It is not who I am calling since it works with any other bluetooth headset
If you use 2.4, consider 2.6, as its ztdummy works better. If you use
2.6, you may be using udev, and need to read README.udev .
Bingo!
I had read the README.udev, but had not noticed any make-time udev
related messages so chose to ignore its contents.
Bad, bad boy - naughty
On Sat, 2005-07-09 at 13:52 +0100, Shaun Orchard wrote:
See inline
I am running Asterisk 1.0.9 with chan_bluetooth (kernel 2.6.11)
1.0.9 with kernel 2.6.12
I have managed to get the module loaded and it connects to my phone and
dials (Nokia 6310i)
Nokia 6680
but once the call connects
Can anyone tell me how to register users in oh323.conf ... i m currently using Netmeeting SJPhone n i can call from/to them without creating user accountsproblem is that my Netphone KU1120(IP phone) uses uid and password for authentication... is there any way to define users like
On Sat, 2005-07-09 at 06:21 -0700, Adeel Ali wrote:
Can anyone tell me how to register users in oh323.conf ... i m
currently using Netmeeting SJPhone n i can call from/to them without
creating user accountsproblem is that my Netphone KU1120(IP
phone) uses uid and password for
wassim darwish wrote:
how to edit the time of ring 3ms to 4ms in
astcc since it displays this on console Nobody picked
up in 3 ms when nobody picked up the phone in
3ms and then it hangup.
please help i have been asking this question from long
time and no body answered me yet.
wassim darwish wrote:
how to edit the time of ring 3ms to 4ms in
astcc since it displays this on console Nobody picked
up in 3 ms when nobody picked up the phone in
3ms and then it hangup.
please help i have been asking this question from long
time and no body answered
Title: Re: [Asterisk-Users] editing ring time
I am using the auto-dial-out
feature to play recordings. I create the call files, place them in the
outgoing directory and off they go.
The problem is that the number I am dialing
does not get stored in CDR. One suggestion was to put this
Rich Adamson wrote:
wassim darwish wrote:
how to edit the time of ring 3ms to 4ms in
astcc since it displays this on console Nobody picked
up in 3 ms when nobody picked up the phone in
3ms and then it hangup.
please help i have been asking this question from long
time and
what if a phone is a H323 phone???
On 7/9/05, Guillermo Salas M [EMAIL PROTECTED] wrote:
On Sat, 2005-07-09 at 06:21 -0700, Adeel Ali wrote:
Can anyone tell me how to register users in oh323.conf ... i m
currently using Netmeeting SJPhone n i can call from/to them without
creating
On Saturday 09 July 2005 10:33, Zoltan Szecsei wrote:
I too have noticed that on my thread sometimes my postings take over an
hour to pop up
(Maybe this list server engine is clever enough to know when someone
hogs too much bandwidth ;-) )
It's a mailing list, not a realtime interactive
Hi again,
Well, thanks for the
details steps. But before I received your mail I had already installed [EMAIL PROTECTED] v.1.3 and updated it with OH323
add-on. It is a zip file which when you install you get all the libraries
installed and compiled for you.
Now, one last step
for me which
This is becoming a waste of time and bandwidth. He doesn't know the
dial-command, he can't use google and he can't read email... I don't
think he'll be around here much longer. I would say by ignoring his
posts we're only replying in kind.
-Original Message-
From: Eric Wieling aka
Ha! yes - we are getting there - hopefully soon you will allow yourself
some time for anything other than me.
see inbetween - and then at the end.
Rich Adamson wrote:
Now we're getting there. In one of your previous emails, you indicated:
8) IAX username - still left blank
9) IAX password -
PJ,
You should check out the Polycom 500/501/600. I'm quite sure it has all
that (although I don't use all of what you listed).
- Dan
Pavel Jezek wrote:
Still looking for cheaper (under $250,-) alternative to cisco 7940
with features needed for corporate use, mainly:
- shared phone book
We have a McLeod T1 and they told us specifically that it was a PRI, ended up
being em_wink. Make sure they really have it setup right.
--
~Andy Brezinsky
On Friday 08 July 2005 5:45 pm, Kristian Kielhofner wrote:
Hello everyone,
We have recently turned up a new T1 from McLeod
On Sat, 2005-07-09 at 22:48 +0800, IM.Nobody wrote:
what if a phone is a H323 phone???
You need gnugk to register H.323 phone.
You must have to include your SIP extensions in your oh323.conf, in
example:
;-
; Configure H.323 aliases, prefixes and
;
He was wanting to edit the dialtime in astcc. I have sent him a patched
copy and I think the issue has been resolved.
Darren Wiebe
[EMAIL PROTECTED]
Jay Milk wrote:
This is becoming a waste of time and bandwidth. He doesn't know the
dial-command, he can't use google and he can't read
Hello:
I dont know, if is my question to do hier, or in the dev-list, but anyway:
I 've installed Asterisk (head, development because I need Realtime),
but when I try to apply the patch I 've got many errors, reason why I
wrote myself the apps/Makefile.
(Of course, first, I compiled spandsp,
I have found out that there are 2 modules.conf in [EMAIL PROTECTED].
One in /etc/asterisk/modules.conf and another in /etc/asterisk/default/modules.conf
Also skinny.conf is located in /etc/asterisk/default/skinny.conf, however sccp.conf is in /etc/asterisk/sccp.conf
Should I copy sccp.conf to
Go into the file astcc.agi and find the exec dial line and edit it.
- Original Message -
From: Darren Wiebe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, July 09, 2005 8:40 AM
Subject: Re: [Asterisk-Users]
On Sat, Jul 09, 2005 at 01:15:16PM +, Leonardo F. Bauchwitz wrote:
Hello:
I dont know, if is my question to do hier, or in the dev-list, but anyway:
I 've installed Asterisk (head, development because I need Realtime),
but when I try to apply the patch I 've got many errors, reason why I
On 7/9/05, Dan Perik [EMAIL PROTECTED] wrote:
PJ,
You should check out the Polycom 500/501/600. I'm quite sure it has all
that (although I don't use all of what you listed).
IIRC, the 500's browser is crippled. I think you have to go up to the
600 to get that functionality.
-Brian
Hi,
I have uploaded the all the .conf files and screenshots of the log and Xlite.
http://www.amsystems.cc/7910/files.zip
Please unzip them and check what is wrong.
Thanks,
JavierSergio Chersovani [EMAIL PROTECTED] wrote:
Javier Chia ha scritto:The phone is now logged in but can´t place
I have an issue with silent calls when an agent gets a call from the queue
What happens is
- The system dials a call (agent call)
- The caller picks up
- Asterisk sees the person picked up
- Transfered to an agent
- Agents phone automaticly picks up (sjphone auto accept on)
-The user
Asterisk/phones work perfectly within our LAN. Asterisk box has a public IP -
no NAT or firewalls. When I take the phones to a remote location (again,
public IP - no NAT or firewalls that I know of) the outgoing audio does not
work. I can hear the other party, my phones ring, I can dial
I use the distinctive ring detection for our front door intercom, and
I've noticed it's not 100% effective. If this is a business type line,
I think I might try to find another solution if it's important that it
works 100% of the time.
-Mishehu
Andrew Kohlsmith wrote:
On Friday 08 July
Based on last nights breakthru this mornings fiddling, I have
minimised iax.conf filled in everything on the phone itself.
Hallelujah! (I'm sure Rich Carlos will agree) :-)
I'm still not ringing the other phone, but that is now surely a dialplan
issue - extensions.conf has been
About once a day I have noticed a phantom incoming call with a caller ID of
[EMAIL PROTECTED]cut off. When I answer the call there is a dial tone
and the call is disconnected. Any clues?
David Koski
David and List,
I am having the same problem.
I have an * box at my house with 1 zap (pstn on
thank you Brian,
but seems, that Polycom phones are not very good option for general
corporate use and even not for use with asterisk (* explicitly
unsupported!), look:
from voipsupply.com:
Please Note: Polycom phones are not supported under Asterisk Open Source
PBX.
from Polycom FAQ:
Can
Many telcos do an automated once a day or once a week or ?? line test,
which can appear as an incoming call to some devices.
If you unplug your telco line and the events disappear, perhaps that is
what is happening?
John Novack
John Millican wrote:
About once a day I have noticed a
Pavel Jezek wrote:
thank you Brian,
but seems, that Polycom phones are not very good option for general
corporate use and even not for use with asterisk (* explicitly
unsupported!), look:
from voipsupply.com:
Please Note: Polycom phones are not supported under Asterisk Open
Source PBX.
Hi:
Astcc is working fine, except for one thing. It
doesn't give the called party enough time to answer
the phone. If nobody picks up in two rings, astcc
reports back no answer and hangs-up. The only instant
NOANSWER value was mentioned in astcc.agi script is:
elsif ($res eq NOANSWER) {
Javier Chia [EMAIL PROTECTED] :
I have uploaded the all the .conf files and screenshots of the log and
Xlite.
well let's start from extensions.conf
; Cisco 7910
replace
[121]
with
[sccp]
because in your sccp.conf the context is sccp
exten = 121,1,SetCalledParty(PRUEBA121)
exten =
Are you sure that the video is set up correctly? If you have a cheap webcam
you have to turn off video hardware acceleration.
Cheers.
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Blake Krone
Sent: Friday, July 08, 2005 5:53 AM
To: Matt Riddell
Hi:
Please save the bandwidth if your answer is going to
be go to google or read the wiki.
Regards;
Chawki
--- Jay Milk [EMAIL PROTECTED] wrote:
This is becoming a waste of time and bandwidth. He
doesn't know the
dial-command, he can't use google and he can't read
email... I don't
If you searched the archives you might find the answer from the past
couple of days. I'll resend it.
In astcc.agi there is are lines similar to this:
$dialstr = IAX2/$res-{path}/$phone|30|HL( . ($maxtime * 60 *
1000) . :6:3);
change the 30 to however many seconds you want.
You have to change the sip.conf and set context=sccp for x-lite to be
able to dial 121
http://www.voip-info.org/tiki-index.php?page=Asterisk+Dialplan+Introduction
Sergio
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
It is by no means cheap, and it works doing an echo test, just doesn't
work when I try to transmit to another side.
On 7/9/05, Storm D. J. Petersen [EMAIL PROTECTED] wrote:
Are you sure that the video is set up correctly? If you have a cheap webcam
you have to turn off video hardware
I have an extension setup in my extensions.conf for hold music. ext. 600.
If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600
I hear the hold music playing. If I call another extension and pick it
up and put the call on hold with the hold button on the phone I hear
I think that it might have to do with the codec that is being used. I had a
problem trying to get the hold music to work with calls that went over our
trunk. I can't remember which one did not work but hopefully this will give
you a direction.
Rick
-Original Message-
From: [EMAIL
Hi,
I have downloaded
asterisk-oh323-0.6.6.tar
pwlib-Janus_patch4-src-tar
openh323-Janus_patch4-src-tar
pwlib and openh323 compiled fine as instructed.
When I tried to compile asterisk-oh323
I am getting this and anybody know howto fix this?
[EMAIL PROTECTED] oh323]# cd
Zoltan,
If you don't mind, can you explain me a little more the ztcfg problem. My
experience with that is null, but I need to setup a box for testing purposes
only, and I don't want to but a TDM card only for make it work.
I know people that has asterisk running on SIP that don't use zaptel
On Sat, 2005-07-09 at 17:59 -0500, CM Rahman Jr. wrote:
Hi,
I have downloaded
asterisk-oh323-0.6.6.tar
pwlib-Janus_patch4-src-tar
openh323-Janus_patch4-src-tar
pwlib and openh323 compiled fine as instructed.
When I tried to compile asterisk-oh323
Try this link:
Some way you should have a udp filter between you box and your phones. I see
that before.
Can you call those phones?
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ross
Overstreet
Sent: Saturday, July 09, 2005 2:04 PM
To:
Yes, I can call the phones, they ring, etc, and call call out, just no outbound
audio. Is their any difference in the inbound outbound audio streams in
Asterisk that could cause it, e.g., different ports, protocols,
connection/discovery methods, etc?
Thanks,
Ross
-- Original
sounds like the inbound (out from the phone, into the local net) RTP
packets are getting dropped..
just a guess here.. whats the output of iptables -L -v
Ross Overstreet wrote:
Yes, I can call the phones, they ring, etc, and call call out, just no outbound
audio. Is their any difference in
Your post is just as superfluous as mine... or this very post. But if
you scroll down a bit, you'll see a I gave a proper reply yesterday.
However, the OP doesn't seem to grasp some basic internet principles,
such as... waiting for a response before re-posting, reading a response
when it occurs,
About once a day I have noticed a phantom incoming call with a caller ID of
[EMAIL PROTECTED]cut off. When I answer the call there is a dial tone
and the call is disconnected. Any clues?
David Koski
David and List,
I am having the same problem.
I have an * box at my house with 1 zap
I have a conference set up through MeetMe and I
can record each call coming in with the Monitor command. What I would like to
move away from is having to then generate multiple files for the final output of
these calls.
On voip-info.org, there is an 'r' option to record
the conference.
Hello Tzafrir:
Tzafrir Cohen wrote:
On Sat, Jul 09, 2005 at 01:15:16PM +, Leonardo F. Bauchwitz wrote:
Hello:
I dont know, if is my question to do hier, or in the dev-list, but anyway:
I 've installed Asterisk (head, development because I need Realtime),
but when I try to apply the
Never used. $250 + shipping (your choice of method).
- a
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I just install a Digium TDM04B card. I created 4 separate Zap channels and
one outbound routing containing zap channels from 1 to 4. If a phone line is
plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating
fail) the system keep dialing out on Zap/1, even with no
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