Displaying 20 results from an estimated 10000 matches similar to: "How to Clear SIP Channels"
2007 May 23
16
WiFi SIP phones
Greetings list,
What are people's experiences with WiFi SIP phones?
When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices.
I assume things must have moved on somewhat since
2005 Jul 18
5
TDM04B - Takes long to initialize...
Hello All,
I got my TDM04B card installed and configured.
Everything works fine I can receive calls and route to appropriate
extensions.
The only problem I am facing is Slowness.
When I dial the PSTN number which is connected to Zap 1-1 after two
ring it answers and then run the AGI script. What I did was assign it
to a specific extension. So all inbound call on that PSTN number
should
2007 Aug 11
1
LumenVox Speech Recognition
Hello All,
While looking for solution to solve my Callback DTMF problem, I came
across LumenVox Speech Recognition software.
Has anyone tried out? Need some feedback before I purchase it... Please
help...
Cheers,
Nitesh
2007 Jun 24
3
Nokia N95 + Dial Plan
Hello All,
Recently I added some Nokia N95 customers and it worked pretty good.
Now the customers are complaining about the dialing rules...
They are used to dialing +12486543210 and +4479XXXXXX for long distance
calls.
Is there anyway to create a "+" sign dial plan which will allow them to
dial a number with "+" sign.
Cheers,
Nitesh
2005 Mar 26
5
Click-to-Talk with Asterisk?
Hi Nitesh,
Take a look at this
http://www.microappliances.com/site/html/index.php?section=Products&page
=clienthowto.php
I've never implemented it though so I would appreciate some feedback on
if it works.
Cheers,
Dean
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Saturday,
2005 Feb 23
4
Vonage <---> Asterisk Working Config!
Hi Nitesh, check out my config that I have for the Faktortel config in
the asterisk@home sourceforge forum, you'll probably be able to work out
how to set it up from there.
Cheers,
Dean
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Wednesday, February 23, 2005 4:12 PM
To:
2005 Feb 18
2
VONAGE <----> ASTERISK SIP TERMINATION?????
Has anyone out there successfully set up their * box to terminate their
VONAGE calls?
I (and I am sure lots of others) would love to hear how you did it.
I'd like to be able to get rid of the extra hardware I have hanging around
here and use the ASTERISK machine to handle the SIP termination instead of
needing to have a Linksys modem (w/phone) and an additional X100P card.
Thanks.
2007 Jul 19
2
Upgrade Procedure
Hello All,
I would like to upgrade my recently installed Asterisk 1.2.21.1 to
Asterisk 1.4.8?
My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6
05:25:07 EDT 2007 i686 i686 i386 GNU/Linux
Is there any detail step by step procedure to uninstall the current
version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons
1.4.2?
Cheers,
Nitesh
2007 May 23
3
What replaces SetCallerPres in 1.4
Hello
SetCallerPres function seems to be removed from Asterisk 1.4.
What function or application replaced it? Bit of a problem if you want to use CLIR on your PRI connections.
Jon
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2007 Jun 20
1
Asterisk RealTime
Hello All,
I manage to configure Asterisk RealTime and now it loads the SIP
users/peers from MySQL DB. The table I am using is of A2Billing DB
"cc_sip_buddies".
Now the only problem I am facing is incoming calls are failing... The
ATA which is assigned this DID number is behind NAT and according to
Olle's explanations he said "*there's no support for NAT keep-alives
2008 May 08
1
MOH and Licensed G729 codec
Hello All,
Recently, I build three Asterisk 1.4 box and installed licensed copy of
G729 codec. Before installing the G729 codec I tested the MOH on all
three Asterisks box and it was working fine. So I install G729 codec and
retested MOH and it was all wavy... Meaning the music was going up and
down and missing bits and pieces and choppy...
Any idea what did I do wrong? The MOH files are the
2007 Oct 03
1
Asterisk Keep Loosing Registration
Hello All,
For some odd reasons my Asterisk is keep on loosing registration of my
SIP devices. On the SIP device it shows I am RESISTED but when I do "sip
show peers" it shows my sip endpoints are "UNREACHABLE". And it keeps on
flapping "Peer '9099993456' is now UNREACHABLE!" and "Peer '9099993456'
is now REACHABLE!"...
I changed my
2007 Jun 14
4
Que on A2Billing
Hello All,
I got one quick question on A2Billing.
Specs: -
- A2Billing v1.3
- OS CentOS 4.5
- Asterisk 1.2
- Zaptel 1.2
Did the installation and everything is working as it suppose to...
Using the A2Billing documentation, I created the RateCard, SIP Trunks,
and SIP Customers. I was also able to login using XLite Dialer and was
able to call out to my SIP Trunk also.
Now how can I remove the
2009 Jan 10
3
Asterisk/GXW410x IP Analog Gateway
Hello All,
I am trying to setup a small system where Nextone Softswitch will send
traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog
Gateway but for some odd reasons the call are flashed back from
Grandstream to Asterisk and creating a Black loop...
I did follow the instructions provided by Grandstream support but it
doesn't seems to be working...
2007 May 17
2
Blacklist
Hello All,
I was wondering where does Asterisk stores the blacklist numbers?
I looked into the dialplan and it shows that it
*"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB?
hyperion*CLI> show dialplan app-blacklist-add
[ Context 'app-blacklist-add' created by 'pbx_config' ]
'1' => 1.
2008 Jan 18
2
SAY TIME + PHPAGI + Timezone
Hello All,
Is there any way to change the timezone on the fly? I have this little
time clock program running on Asterisk system developed using PHPAGI.
Currently, whenever user logs in, Asterisk will prompt the current
system time using "$agi->say_time();" which executes "SAY TIME". Now the
current timezone set on the system is "PST", and I have a request to
2008 Feb 22
2
AGI / Voicemail Que
Hello All,
I have my own AGI script running and I am trying to push the call to
voice mail when Busy, Unavailable and Not Answered.
Everything is working fine but the only problem is voice mail greetings
for Busy and Unavailable is not played. By default only "Temp Greetings"
voice mail greetings is played. I am passing the correct parameters for
Busy => 'b', Unavailable
2006 Jan 06
7
Fax, txfax -bizarre thing
Hi,
I've been struggling with this for a quite long time. Maybe I am not the
first asterisk user with this problem, (I try to search on google, but I
didn't find anything good). My point is:
I try to set up * to work as a fax server. Each incoming fax (from PSTN)
should be received on email. Luckily it works. I didn't notice any problem
with receiving faxes on email, so rxfax works
2010 Mar 14
2
dahdi-linux-complete-2.2.1+2.2.1 failed to compile
Hello All,
I'm trying to do a fresh installation on Ubuntu Server 9.10 (Karmic)
64-bit but I am getting error when "make config" is trying to install
the init script... Here is the output: - Can anyone help me please...
Thanking in advance...
Cheers,
Nitesh
###################################################
###
### DAHDI tools installed successfully.
### If you have not done
2007 May 12
2
zonedata.c
Hi,
Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly.
Thank you.
Jad Wauthier
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