[Asterisk-Users] connection between asterisk and cisco

2005-12-09 Thread muhammad usman
HI!

how are you people. i am a newbie in asterisk and
voip.
i need your help.

the scenerio is like this.

1.all local SIP users will be connected to asterisk
via IP.

2.PSTN will be connected to AS5300.pstn will give us a
local prefix like 333. so any one calling at
333 will go to my as5300.

3.now i want if someone calls via PSTN to a number
333 this should go to my some sip user e.g john
(connect to asterisk via ip). but only to john.

4.now when john dials to any number outside 333 range
, it should be dialed to the destination via
AS5300(which is connected to PSTN). and destination
should see that it is called by a number 333.

5.now if all this scenerio is possible, how the
asterisk server and As5300 will talk to each other.
what protocol can be used between them.
and what physical connection i.e like ethernet or E-1
connection between as5300 and asterisk server.


6.which billing radius server you recommend, and what
kind of cards will be required in a5300.


thanks a lot for reading this.
and thanks for reply in advance.


any other suggestions are also welcome.

regards







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[asterisk-users] digim tdm2400p fxo fake answer supervision problem.

2011-01-03 Thread Muhammad Usman
Hi. I am using Digium TDM2400PFXO with Asterisk1.6. When call sets in the
box , it answers the call even the phone is not picked. ideally it should
answer the call when the phone is picked up. Its charging the clients.
Please let me know how can I cover this ? Thanks in advance.
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[asterisk-users] Fix Fake Answer Supervision In asterisk1.6

2011-01-10 Thread Muhammad Usman
Hi,
I have installed asterisk1.6+DAHDI for TDM2400P Digium card. When call hits
the box, the gets answered even the other end phone in not picked. How can I
fix this as ideally it should answer the call when other end phone is
picked.
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Re: [asterisk-users] (no subject)

2011-04-29 Thread Muhammad Usman
you running GSM FWTs with asterisk ?

On Mon, Apr 25, 2011 at 6:51 AM, Abid Saleem abid_aster...@hotmail.comwrote:

  HI,

 I am trying to setup a Class 4 termination setup using a kind of channel
 hunting scenerio. I have some SIP DID numbers assigned from the local
 telecom provider for termination. MY call comes from my wholesale client and
 lands on a switch, then it is routed to asterisk. I want asterisk to route
 this call to my local DID provider on the next available channel with DID
 number as the new Caller ID. This is just like GSM gateway that recieves the
 call and then re-originates the call using the next available SIM card
 number.

 Can someone help me how can I configure Asterisk to perform this?

 Thanks

 Abid.

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[asterisk-users] Multiple IAX2 Trunks Load balancing

2013-12-12 Thread Muhammad Usman
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to
load balance incoming calls over IAX2 trunks. If any trunk goes down the
calls traffic will be shared with other available trunks. When it gets Up
the script is supposed to perform as desired i.e in load balance mode.


Thanks in advance.
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Re: [asterisk-users] Multiple IAX2 Trunks Load balancing

2013-12-13 Thread Muhammad Usman
yeah -- searching how to perform this magic ...


On Fri, Dec 13, 2013 at 2:29 PM, Steven Howes steve-li...@geekinter.netwrote:

 On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote:
  Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to
 load balance incoming calls over IAX2 trunks. If any trunk goes down the
 calls traffic will be shared with other available trunks. When it gets Up
 the script is supposed to perform as desired i.e in load balance mode.

 Sounds wonderful.

 S
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Re: [asterisk-users] Multiple IAX2 Trunks Load balancing

2013-12-13 Thread Muhammad Usman
Friends let me define the scenario please;
Scenario:
2 asterisk servers (A  B) are connected using 05 IAX2 trunks between them.
The machine A is running asterisk  Openvpn server in TUN mode (5 instances
with difference IP addresses for clients). The machine B is running
asterisk with 05 OpenVPN clients using 05 bandwidths. The IAX trunks are
established between each pair of P-2-P ip address of machine A (The OPENVPN
Server)  machine B (The Openvpn client).
Requirement:
Required dial plan configuration at machine A for incoming calls from VoIP
Switch/VOS which can forward the calls to IAX2 trunks in round robin
fashion like Load Balancing. If any trunk goes down it starts forwarding
the traffic to other available trunks  when it gets UP the dialplan should
perform as desired. Like L.B  Fail-over scenarios.


On Fri, Dec 13, 2013 at 8:52 PM, Hans Witvliet aster...@a-domani.nl wrote:

 On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote:
  On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
   Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want
   to load balance incoming calls over IAX2 trunks. If any trunk goes
   down the calls traffic will be shared with other available trunks.
   When it gets Up the script is supposed to perform as desired i.e in
   load balance mode.
 
   Thanks in advance.
  
 
  Hans said:

 
  Perhaps it is possible to do the L.B. at the O.S. or network level, and
 let
  all trunks appear to asterisk to one single trunk.
 
  Don asks:
 
  What's the value of load balancing multiple IAX trunks between the same
  system pair? What resources are being balanced?
 
 ++

 Perhaps the O.P. can explain about his intentions...

 In some situations it makes sense though:
 If you have to connect two servers, and use different kind of
 infrastructure / multiple providers...

 hw


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