If you have 10 billing plans from different providers, you have for sure at
least almost all the data. Use the prefix from the plans to build your own
database of prefixes and destinations.
Il giorno 23/mar/2012 05:06, Mikhail Lischuk mlisc...@itx.com.ua ha
scritto:
**
Is it a problem to parse
2012/3/22, John Knight j...@classiccitytelco.com:
I've tried this in the past and while FreePBX and its base modules work
fine in an http proxy environment, some applications like fop2 fail to
connect properly as they obviously rely on direct connections via ajax
using the browser as a
Hello,
I have a DID number 5672531308 , I want to add 92 prefix in it as been told
by my provider , so I can I do this in extensions.conf?
--
Regards,
Muhammad Ali
DIDx SUPPORT
http://www.didx.net
Skype: didxnet
Phone: +1-212-655-5763 / +1-850-433-8555
Direct : +1-567-2531308
Hello,
First let me apologize for posting about a GUI topic on here. There's a reason
why I did that, and it's because the underlying concept of this is connected to
Asterisk.Here's my situation:
Twenty wifi clients connecting to our wireless router (Cisco Linksys E4200
loaded with Tomato).
Hi,
How about having 2 NIC cards on the PBX(configure the machine as a gw
of sorts).
On 3/23/12, Sean McMaster sean.mcmas...@msn.com wrote:
Hello,
First let me apologize for posting about a GUI topic on here. There's a
reason why I did that, and it's because the underlying concept of this
On Mar 22, 2012, at 11:25 PM, Shaun Ruffell wrote:
On Fri, Mar 16, 2012 at 11:13:14PM +0400, Vahan Yerkanian wrote:
I've tried upgrading one of my servers with yum update to the
latest dahdi/asterisk, and found out that my 4th gen TE410P is
failing the dahdi init with
Running dahdi_cfg:
file /usr/lib64/libstdc++.so.6.0.10
/usr/lib64/libstdc++.so.6.0.10: ELF 64-bit LSB shared object, AMD x86-64,
version 1 (SYSV), stripped
file astdb2sqlite3.o astdb2sqlite3.o: ELF 64-bit LSB relocatable,
AMD x86-64, version 1 (SYSV), not stripped
file db1-ast/*.a
db1-ast/libdb1.a: current
Assuming this is outbound, no problem. For inbound, I don't think so
either. Can you be a little more specific?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Muhammad Ali
Sent: Friday, March 23, 2012 4:30 AM
To:
Hello,
ITSPA UK has unveiled the winners of its 4th annual Awards, an event
designed to celebrate innovation and best practice in the VoIP industry:
* http://www.itspaawards.org.uk/
Open Source VoIP Projects won a special category this year, Members'
Pick, for providing a real value to
The short answer is yes you can. Now the longer answer is give us more detail
if you want to know how. Are they asking you to add the 92 when you dial
5672531308, or is this question really about the callerid number?
***
Sam Lutgring
Director of
Hello,
backtrace was created. Can anyone help me with understanding it and
telling me what went wrong with my Asterisk-server ? Thanks in advance !
This is Asterisk 1.6.2.22.
[root@sip1 ~]# gdb -se /usr/sbin/asterisk -ex bt full -ex thread
apply all bt --batch -c core.sip1
Hi All;
If we need the admin to have the ability to hear what the agent is talking
without the agent or the customer feel, how this can be done? Is it using
MeetMe or something else? How? When the admin start hearing, there will be a
peep (because I do not need this, it should be silent
Check the chanspy application available in the dial plan.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_ChanSpy.
--- Jayesh
On Saturday, March 24, 2012, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
If we need the admin to have the ability to hear what the agent is
Hi,
I have seen similar post several times on the list but without a proper
solution. Is it possible to add custom SIP Headers in 1xx or 2xx response.
The SIPAddHeader only works for initial INVITE. Is there any workaround for
this. There are certain polycom phones which can open up an URL when a
You are welcome to an incomplete dataset I have. Data was gathered from
publically available sources, including the ITU and Wikipedia. Data does NOT
include information for country code 1.
http://rock.nyigc.net/e164.csv.gz
Enjoy.
-Original Message-
From:
And then how will I send calls over to the vpn trunk?
-Original Message-
From: James Mutuku listmut...@gmail.com
Date: Fri, 23 Mar 2012 12:28:26
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN
Hi,
How about having 2 NIC cards
On 24/03/2012 04:49, Sean McMaster wrote:
And then how will I send calls over to the vpn trunk?
via route with high metric...
Regards,
Eliezer
-Original Message-
From: James Mutukulistmut...@gmail.com
Date: Fri, 23 Mar 2012 12:28:26
To:asterisk-users@lists.digium.com
Subject: Re:
Either give it a 2nd address on the nic that can access the VPN modem
You can have lots of addresses on a nic to access different sinners on the LAN
Or just make sure the gateway knows to route the ipvpn traffic via the vpn modem
Cheers Duncan
On 24/03/2012, at 3:55 PM, Eliezer Croitoru
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