Shift + Page Up and Shift + Page Down. Leif Madsen told me this in 2005 when
I was new to Linux and Asterisk, at an Asterisk seminar in Mississauga.
Thanks Leif, it made my life easier to scroll through the logs.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-04 11:36 PM, bruce bruce
Dear All,
Is there anyway to put the call on Hold and Retrieve the call based
on external configurations through AGI? Please help me...
Regards,
Velusamy.
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On Sun, 4 Jul 2010, bruce bruce wrote:
And the 20k+ lines is where it's really hard to handle. The scroll bar is
too small and I was wishing there was an easy page up or page down function
maybe to it rather than using the mouse.
No-one's mentioned 'screen' yet.
Use putty to connect to a *ix
Hello Gareth,
echo also appears when making calls with a SIP phone. These are outgoing
calls.
Another site now also gives feedback on echo, telling they sometimes
also have echo on outgoing calls and if they recall right then sometimes
also on incoming calls (coming from a queue).
This
Dear Please send us, your iax configurations.
best
On Mon, Jul 5, 2010 at 7:10 AM, bruce bruce bruceb...@gmail.com wrote:
Hi guys,
I have two Asterisk servers (with FreePBX) connected together with IAX2
trunking. When I call from server A-B call connects but hangs up after 30
seconds. What
add the a2billing configurations to the sip.conf
best
On Thu, Jul 1, 2010 at 7:34 PM, bruce bruce bruceb...@gmail.com wrote:
Yes, you are missing a whole bunch of configurations from creating SIP
users to making sure they show as peers on Asterisk to making sure you use
dnid, etc.You
please send your extension.conf
2010/6/30 Anahi Ludueña a_ludu...@hotmail.com
Hi people,
we have some extensions which are included in the IVRs and/or queues.
Everything works fine, but the calls done from these extensions are hang up
after 30 o 35 seconds. If they are not included in the
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are
unable to URI dial our clients. We run a multi-tenant server and have set
sip.conf to forward calls to a public context based on incoming domain name.
This was all working before but not it is complaining of a loop
Has anyone mentioned Teraterm in this thread? I know it's very old but
I also know it worked well with XP. I preferred it over Putty, but I
haven't used Putty in years either. Nowadays, I use mostly Mac with
occasional virtual XP - and the OS X terminal is great. It's a little
surprising that no
Hello,
I'm trying to register to my provider sip trunk, I got from him an host IP
(a.b.c.d) to connect to and my provider recognize me based on the fixed IP
(x.y.z.w) he gave me (no need for username and password)
In the sip.conf I add:
[mytrunk]
type=friend
insecure=no
host=a.b.c.d
Hello
In case Asterisk is used in a private LAN behind a firewall while
allowing remote SIP clients to connect from the Net, we must open
UDP5060 for SIP and a range of UDP ports (as set in rtp.conf) so let
incoming voice packets.
Provided the user doesn't have access to the firewall
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that
we are unable to URI dial our clients. We run a multi-tenant server
and have set sip.conf to forward calls to a public context based on
incoming domain name. This was all working before
On Mon, Jul 05, 2010 at 12:09:30PM +0200, Randy R wrote:
Has anyone mentioned Teraterm in this thread? I know it's very old but
I also know it worked well with XP.
Teraterm only supports the old, insecure and much less capable ssh1
protocol, IIRC. Many recent SSHDs disable ssh1 support
Hi
I have a EMT-22 IP Phone. I need user name and password to access it from ip
address.
Thanks
Gsphull
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PS: http://www.ayera.com/teraterm/
I'm pretty sure there was a last update or patch or something because
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New to Asterisk? Join us for a live
On Mon, Jul 5, 2010 at 1:43 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Teraterm only supports the old, insecure and much less capable ssh1
protocol, IIRC. Many recent SSHDs disable ssh1 support nowadays. Don't
use it.
I'm pretty sure there was a last update or patch or something because
On Mon, 5 Jul 2010 14:05:09 +0200, Randy R wrote:
PS: http://www.ayera.com/teraterm/
I'm pretty sure there was a last update or patch or something because
For as long as I have used Asterisk I have used either the freeware
PuTTY or a commercial SSH/SFTP client called Private Shell.
hello
you must to do a configuration of yor sip.conf
like that
[the login of sip]
type=friend
context=default
secret=(the password of sip )
host=dynamic
dtmfmode=auto
disallow=all
allow=alaw
allow=ulaw
qualify=yes
Regads
2010/7/5 Pezhman Lali l...@lopl.net
add the a2billing
Hello, i just had some fax abortions because of some packet loss. so i
startet to examine in the pcap recording
from the res_fax_digium, if the T.38 EC mode redundancy was really
used. So i watched into it, and compared it
with a t.38 pcap from spandsp (same asterisk setup, but with app_fax)
and i
hello,
i had the same issue when using x-lite when i verify i found that the issue
is related to configuration of x-lite i change the value in x-lite option
and now there is no issue all function good
Hope it can help you
2010/6/30 Anahi Ludueña a_ludu...@hotmail.com
Hi people,
we have
Dear Folks,
I'm running Asterisk 1.4.31 server, on an Ubuntu 9.10 system. My scenario is
simple: connection to the PSTN directly via SIP, using g729 codec, and
connection to the softphones (X-lite 3.0 build 56125) trought local network,
using ulaw codec.
Sometimes, I got messages like:
[Jul 1
Hello,
I'm trying to use a SIP trunk service and the provider ask me to have the IP
address of the contact header as my public IP and not as my private one, how
can I do it?
See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w
is my public address:
sipINVITE
Have you tried setting
externip=
In the [general] of your sip.conf?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Goltzman
Sent: Monday, July 05, 2010 1:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that
we are unable to URI dial our clients. We run a multi-tenant server
and have set sip.conf to forward calls to a
On Mon, Jul 5, 2010 at 5:05 AM, Randy R randulo2...@gmail.com wrote:
PS: http://www.ayera.com/teraterm/
I'm pretty sure there was a last update or patch or something because
Whats different about teraterm compared to putty? I know back in the
day I used to send files to my linux box with
- Original Message -
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
that
we are unable to URI dial our clients. We run a multi-tenant server
and have
Yes, I tried and it did not solve the problem,
Thanks
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A.
Stapleton
Sent: Monday, July 05, 2010 9:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
By definition, correct values for localnet, externip and nat=yes for this
trunk should solve this problem.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-05 3:40 PM, Eyal Goltzman egoltz...@gmail.com wrote:
Yes, I tried and it did not solve the problem,
Thanks
*From:*
I'm having issues with T.38 on Asterisk 1.6.2.8. A few lines are received OK
and then I get only garbage. I'm using ReceiveFAX() provided by app_fax to
receive the faxes.
After talking to the engineers on the telco, they said Asterisk is sending a
REINVITE to alaw after the T.38 reception is
Hi,
Does anybody have experience working with Aculab groomer II, to convert
between ISDN E1 and non-ISDN T1, or anything similar. I am looking for
sample config files. We have asterisk as ISDN E1, but for testing we set it
up as regular T1 if we get sample config files.
Zeeshan A Zakaria
--
Hello all,
Does anybody know if is it possible to install dahdi on solaris 10?
I've only found a zaptel modified code for solaris at solarisvoip site.
I'd appreciate any comment or experience about asterisk + dahdi/zaptel on
solaris..
Best regards,
Caio
--
Hi Claudio,
As far as I am aware, dahdi is not able to compile on Solaris, although I've
not attempted to compile it. There may be others out there that may have
better experience than I with dahdi on Solaris.
Thanks
Bruce
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hello all Asterisk Users,
This is my first post here.
We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server
to Asterisk box.
Which card drivers do we need?
Please share experience if anyone have successfully configured Dialogic
JCT-T1 card with asterisk?
Only source proves
Hello all,
I am putting together an installation for our organization. My dialplan
has most users in context [inside], and a separate [users] context
includes the inside context.
My voicemail config file has these users in a [users] context.
I did this so I could get the name directory to work
On Mon, Jul 5, 2010 at 7:29 PM, AMARDEEP SINGH yahhod...@gmail.com wrote:
Please share experience if anyone have successfully configured Dialogic
JCT-T1 card with asterisk?
Your not going to find much; there is no channel driver for Dialogic.
--
Paul Belanger | dCAP
Polybeacon | Consultant
List,
Its been 2 weeks since my previous email and this time I am linking
all 97 issues marked 'Ready for Testing' [1]. Simply follow the link,
view the available patches, download, compile and install. Report
your result into the actual issue, we can them continue to triage the
issue.
The
Not sure if this is a bug yet, so I wanted to ask around to see if anyone else
was having this issue. I have pollmailboxes=yes set in voicemail.conf but
externnotify is not called. I know it isn't the externnotify script because if
the changes are done in asterisk then it is called properly,
On Mon, 2010-07-05 at 19:59 -0400, Paul Belanger wrote:
On Mon, Jul 5, 2010 at 7:29 PM, AMARDEEP SINGH yahhod...@gmail.com wrote:
Please share experience if anyone have successfully configured Dialogic
JCT-T1 card with asterisk?
Your not going to find much; there is no channel driver for
OK, feeling very stupid right now.
The test mailbox had delete=yes option set. All cleared up; sorry for
cluttering up the list.
Cassius
snip
Now, however, I don't get message waiting lamp to show up on the phones
and when the recipient of a voicemail tries to retrieve the message
Alyson says
What do you mean now that ABE is discontinued? My company payed thousands of
dollars this year for the product and the support it provides!
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 5, 2010, at 5:42 PM, Carlos Chavez wrote:
On Mon, 2010-07-05 at 19:59
On Mon, 5 Jul 2010 18:17:48 -0700, Jim Dickenson wrote
What do you mean now that ABE is discontinued? My company payed
thousands of dollars this year for the product and the support it provides!
Well, last year when I took my dCAP that is what my instructor commented.
Since Digium now
On Monday 05 July 2010 20:17:48 Jim Dickenson wrote:
What do you mean now that ABE is discontinued? My company payed thousands
of dollars this year for the product and the support it provides!
Those who paid for ABE support will continue to get it, and those who really
want ABE can still
On Monday 05 July 2010 19:17:00 Eric Hiller wrote:
Not sure if this is a bug yet, so I wanted to ask around to see if anyone
else was having this issue. I have pollmailboxes=yes set in voicemail.conf
but externnotify is not called. I know it isn't the externnotify script
because if the changes
I am writing to you privately because I am an asterisk consultant and if you
need any help I can help you for a fee. I have worked with dialogic cards
for several years, until I kicked them out my life when Intel bought
Dialogic J
Having said that however, these are my thoughts:
You have to
On Mon, Jul 5, 2010 at 9:03 PM, Kyle Kienapfel doctor.w...@gmail.com wrote:
Whats different about teraterm compared to putty? I know back in the
day I used to send files to my linux box with xmodem over ssh. Does
this newer version do that? :)
THe next time I turn on the XP box, I'll try to
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