[asterisk-users] sip configuration masking the peers
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[asterisk-users] sip configuration masking the peers
Hi all,
I need to specify two groups of peers who are on two sub networks, the
case is as follows:
two groups of users (that are supposed to use the X-lite) group1 and group2,
each group is on a sub network net1, and net2, respectively, each group has
its own dial plan defined in the extension.conf,
we have defined the peers in the sip.conf for both groups, and successfully
made a call between two peers from the groups, but the idea is we need to
prevent users from network1 to register as peers of group1,
I suppose this would be a configuration solution, but I am afraid that do
know what are the right needed configurations:
here is definition of two peers each from different group:
[1010]
type=friend
host=dynamic
context=group1
secret=pass
host=dynamic
callerid=TestAccount1010
vm Extension=test 1010
mailbox=1...@default
nat=yes
[2003]
type=friend
context=group2
secret=pass
host=dynamic
callerid=Account2003
vm Extension=test 2003
mailbox=2...@default
nat=yes
each of group1 and group2 context are defined in the extension configuration
as follows :
exten = _2XXX,1,Dial(SIP/${EXTEN})
exten = _2XXX,n,Playback(unavailable)
exten = _2XXX,n,Hangup()
exten = _1XXX,1,Dial(SIP/${EXTEN})
exten = _1XXX,n,Playback(unavailable)
exten = _1XXX,n,Hangup()
in order the both groups can talk to each other,
currentlly users in network1 can register as peer 2003 which is supposed to
be allowed just for users from network2 , although this registration is
supposed to be failed, any suggestions plz!!
hope I made the scenario clear , any help would appreciated.
Thanks in advance.
___
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[asterisk-users] about monitored calls storing
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[asterisk-users] An outside Caller ID not shown,
Hi there, I am using the Asterisk as the PBX, and need to know the caller ID for the incoming call, but when I show the caller Id, it gives the Zaptel channel that recieves the inbound calls in the asterisk, Am I missing some configuration ! what should I do to be able to exteract the real callerId (that from the outside) from that channel, Thanks in advance, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] An outside Caller ID not shown,
thanks for replying, I'll give it a try On Sun, May 31, 2009 at 1:24 PM, Rob Hillis r...@hillis.dyndns.org wrote: Sounds like you're looking at the wrong variable. You should be looking at CALLERID(num). peace keeper wrote: Hi there, I am using the Asterisk as the PBX, and need to know the caller ID for the incoming call, but when I show the caller Id, it gives the Zaptel channel that recieves the inbound calls in the asterisk, Am I missing some configuration ! what should I do to be able to exteract the real callerId (that from the outside) from that channel, Thanks in advance, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] integrating CTI
Hi there, I am integrating CTI functionality to my java application and using Asterisk as PBX. I need some advices as to whether I am in the right track. • Asterisk server is configured and working fine. • I have a generic sip hard phone • I have X-lite soft phone installed. To provide for CTI functionality, I am utilizing the asterisk manager API, I designed a java application that can do the following: • Agent login • Originate call • Hang up • Transfer call • Record on server Is this the right approach, or I should do it differently, please advice? thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
