Hi
I want that all of my call should be asked for a code .
And then all call should go to a fixed extension.
My application will be running there that will differentiate stream of
calls.
like
person A enters 1234
person B enters 2345
both call will be directed to extension say 101, and from there
Hi,
I changed my sip.conf and added call-limit. At first I thought it works ok,
since i tried calling a cellphone that is currently busy(phone answers 1st
softphone, then another softphone calls the same number, it now returns INUSE).
But then, i tried calling a different number while the
Dear Asterisk-Users,
I have this Asterisk Box I run in my house, I need to terminate and originate
remote calls through the box via internet (SIP), the problem is in Nigeria most
ISPs would not provide you with Public Addresses, all they provide is dynamic
Natted addresses which change each
Try http://www.dyndns.com/ that should solve your problem with dynamic IPs.
Regards,
Jonathan
On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide
ayodeleabej...@hotmail.com wrote:
Dear Asterisk-Users,
I have this Asterisk Box I run in my house, I need to terminate and
originate remote calls
hi my friend
would ou say what did you do for solving the problem? because i use a
digium te121p and have many problems.
thanks in advance
On Mon, Oct 25, 2010 at 4:50 PM, Flavio Miranda
flaviormira...@hotmail.comwrote:
Sorry, thats right!!
I the nest email I will post here what I did
Hello list,
I have this problem with dropped calls on Asterisk.
The setup is SIP internal extensions (Grandstream GXP-2000), two
internal analogue DAHDI extensions and IAX2 trunk lines. IAX2 trunks use
ulaw/alaw. The Internet connection is ADSL. Asterisk is 1.6.1.6
Everything worked fine
thanks i would check it up
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
Date: Tue, 26 Oct 2010 12:52:30 +0200
From: jonathan@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
Try http://www.dyndns.com/ that should solve your problem with
Hello Jonathan,
The solution would work only if the ISP has one public address, but in my
solution they have a pool of public address, any other possible solution?
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
From: ayodeleabej...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 26
hi,
So, I think it depend of what environment are you setting up your link . In my
case, E1 R2 Digital Brazil standard (Variant=br), I needed to change
dahdi-channels parameter,chan_dahdi.conf , system.conf as well.
If you need I can send you such configuration.
good look!
Att,
Flavio
On 10/26/2010 06:38 AM, Administrator TOOTAI wrote:
I installed 2 HB8 cards each of them with a Quad Bri modules in a HP 360
G6 running Debian Squeeze. Here is an output of dmesg wafter server has
booted:
[9.784123] wctdm24xxp :0b:08.0: PCI INT A - GSI 31 (level, low)
- IRQ 31
I am trying to configure a channel bank with 24 ports of FXS., but appear to
be hitting a roadblock? This worked on v1.4.xx but now just get
SimpleSwitch and immediate=no/yes don't seem to make a difference?, no
matter if under top section, under channel, etc.
Chan_dahdi.conf:
Nevermind, figured it out.
Immediate=yes on top part of chan_dahdi.conf
And in extensions.conf
Exten =s,1,disa(no-password,internal)
William Stillwell
Systems Architect
MDT Personnel, LLC.
Ph. Coming soon.
Fx. Coming soon.
Cl. 727-638-6208
From:
Thanks Leif,
Forgot I could do a db lookup for the ddi.
Dan
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Have you contacted Sangoma regarding their card configuration?
I have found them always very knowledgeable and helpful
I would certainly go there first.
John Novack
William Stillwell (Lists) wrote:
I am trying to configure a channel bank with 24 ports of FXS., but
appear to be hitting a
Hello,
has anyone experience with auto provisioning IP-phones on different
locations through a central public provisioning server ? You use http or
https ?
Is there a danger that one uses a different MAC-address in the
provisioning link to obtain SIP username / password settings ?
Kind
You can provision over a WAN and access-lists or iptables can limit
the networks allowed. Define what level of security you need first.
For further security you can use an inbound proxy and check the http
headers for agent identification. This can also be faked.
Practice layers of security...
I havent had much auto provisioning experience, however, what about just
using IPTables to create an access list essentially for known IPs to connect
via HTTP/HTTPS and block all other addresses. This would only work if the
phones are coming from a Static IP, but I figured i'd give my 2 cents to
On Tue, Oct 26, 2010 at 12:31 PM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Hello,
has anyone experience with auto provisioning IP-phones on different
locations through a central public provisioning server ? You use http or
https ?
Is there a danger that one uses a different MAC-address
On 10/26/2010 05:40 PM, Matt Desbiens wrote:
I havent had much auto provisioning experience, however, what about
just using IPTables to create an access list essentially for known IPs
to connect via HTTP/HTTPS and block all other addresses. This would
only work if the phones are coming
Hello,
many SIP phones offer you the possibility to provisioning them over a FTP
connection (with username and password).
Regards
- Bakko
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
With the new phones with VPNs you can also do a stepped provision
One provisioning service for the vpn and another for the sip that can
only be reached with the vpn. This is advanced stuff so take your
time and learn about the tech.
--
Think about limiting geographically or use a CDN with good controls.
Thank you for your input, but IP-addresses will change, so this would
then become an administrative and time-consuming job...
Jonas.
--
_
--
On 10/26/2010 05:41 PM, Andrew Latham wrote:
You can provision over a WAN and access-lists or iptables can limit
the networks allowed. Define what level of security you need first.
For further security you can use an inbound proxy and check the http
headers for agent identification. This can
On 10/26/2010 05:52 PM, bakko wrote:
Hello,
many SIP phones offer you the possibility to provisioning them over a FTP
connection (with username and password).
Regards
- Bakko
In this case I will want to use Snom phones. TFTP is available, but no
FTP (with indeed then a username and
On Tue, Oct 26, 2010 at 12:06 PM, Jonas Kellens
jonas.kell...@telenet.be wrote:
On 10/26/2010 05:52 PM, bakko wrote:
Hello,
many SIP phones offer you the possibility to provisioning them over a FTP
connection (with username and password).
Regards
- Bakko
In this case I will want to use
snom phones can do http digest authentication...
In this case I will want to use Snom phones. TFTP is available, but no
FTP (with indeed then a username and password). FTP would be great...
Jonas.
--
_
-- Bandwidth and
Hi!
In this case I will want to use Snom phones. TFTP is available, but no FTP
(with indeed then a username and password). FTP would be great...
You could also consider to use the SNOM Redirection Service for
provisioning:
http://wiki.snom.com/PROVISIONING
Remark: TR-69 provisioning
On 26 Oct 2010, at 16:31, Jonas Kellens wrote:
has anyone experience with auto provisioning IP-phones on different locations
through a central public provisioning server ? You use http or https ?
What handset? That's rather what controls your options. Some support HTTPS with
client
What I am needing to do is to trim the 1 from beginning of the RDNIS and
I have tried using the CUT function but cannot seem to make it work for
me. What we have is a phone number like this, 18881232342 and want to
make it like this 8881232342. I appreciate any help that you guys can
give.
pstn pstn
asterisk link between avaya pbx
both systems tied together by 2 pri's
both have trunks out to the pstn
want to get rid of the avaya pstn trunk and send thru my asterisk box
avaya still has inbound
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Terrell
Sent: Tuesday, October 26, 2010 1:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] need to be able to pass a call to the pstn
Since Google Voice (GV) doesn't let us connect diretly via SIP, IAX2, etc.,
for outbound calls, it acts basically like a fancy click-to-call application.
So...
You need Asterisk to login into GV, and initiate the call. GV will dial
the number you tell it to, then connect it to one of your
Hi all,
I seem to be having a strange problem with a sip trunk.
On a fairly frequent basis, I'll make a call, ore receive a call, and there
will be NO sound. The strange part is that both endpoints are public IP
addresses so NAT isn't in play and a sniffer trace reveals that the packets
On Tue, 26 Oct 2010, Chris Ramirez wrote:
What I am needing to do is to trim the 1 from beginning of the RDNIS and
I have tried using the CUT function but cannot seem to make it work for
me. What we have is a phone number like this, 18881232342 and want to
make it like this 8881232342. I
Hi,
Bump to see if anyone can help us too.
Really this is a problem. I don't want to show the caller id number and
name to the Agent in certain conditions. Changing the CID will mess the
CDR/Queue log and this is not the acceptable behavior.
In the Dial app, everything is OK.
Alexandre
Em
Hi guys, a little OT but I figured this is the place that would know.
Is there a free or paid webapp where I can get inbound sms messages? I
only need to receive a few inbound sms messages a month but it cant be
my current cell number :-(
Any thoughts?
Cheers,
Dean
--
2010/10/26 Mike Diehl mdi...@diehlnet.com
Hi all,
I seem to be having a strange problem with a sip trunk.
On a fairly frequent basis, I'll make a call, ore receive a call, and there
will be NO sound. The strange part is that both endpoints are public IP
addresses so NAT isn't in play and
Google voice...
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
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On Tue, Oct 26, 2010 at 4:41 PM, Dean Collins
dear
please send these configurations.
thanks
On Tue, Oct 26, 2010 at 3:04 PM, Flavio Miranda
flaviormira...@hotmail.comwrote:
hi,
So, I think it depend of what environment are you setting up your link . In
my case, E1 R2 Digital Brazil standard (Variant=br), I needed to change
Hi,
/etc/dahdi/system.conf
Att,# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 (MASTER) HDB3/
span=1,1,0,cas,hdb3cas=1-15:1101dchan=16cas=17-31:1101#echocanceller=mg2,1-15,17-31
/etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
On Sun, Oct 24, 2010 at 2:20 PM, Nic Colledge n...@njcolledge.net wrote:
I made a debug log of the register and unregister process for a single Zoiper
client using IAX and have emailed it direct to you.
The error shows in the file as:
[Oct 24 19:07:32] ERROR[1403] netsock2.c: getnameinfo():
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