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julien at jsansonnens.ch
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 Posted: Mon Nov 30, 2015 10:50 am Post subject: [asterisk-users] force sip URI call through PBX |
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Hello,
When I do a SIP URI call from my softphone, the call is made directly
to the destination host (p2p), bypassing the PBX. So I lose the
possibility of recording, making statistics, etc ...
Is there a way to force URI calls through the PBX? I have found no
configuration at the client or at the server level. Do you know any
softphone that will allows me to do this ?
Thank you and have a nice day, Julien
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Julien Sansonnens
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darcy at Vex.Net
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| Posted: Mon Nov 30, 2015 11:15 am Post subject: [asterisk-users] force sip URI call through PBX |
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On Mon, 30 Nov 2015 16:42:02 +0100
Julien Sansonnens <julien@jsansonnens.ch> wrote:
| Quote: | When I do a SIP URI call from my softphone, the call is made directly
to the destination host (p2p), bypassing the PBX. So I lose the
possibility of recording, making statistics, etc ...
Is there a way to force URI calls through the PBX? I have found no
configuration at the client or at the server level. Do you know any
softphone that will allows me to do this ?
|
If two phones are calling each other directly then there is no server
setting that will reach across the Internet and grab the call. You
need to insure that the call is proxied through your PBX. That's just
a setup in your softphone. You will need to ask about that on a
mailing list for that software.
One thing that I do so that I can call SIP phones from my regular phone
through an ATA is set up extensions for them in Asterisk like this.
exten => 6135553638,1,Dial(SIP/my.friend@example.com)
Looks like a regular call to my users but bypasses the PSTN. This
might work for you as well.
Whatever solution you use, you may want to look at directmedia
settings. If you can talk directly to another SIP client Asterisk may
step out of the picture anyway not allowing you to record the call.
Turning that off forces all the calls to be proxied through you even if
they could talk directly.
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@Vex.Net
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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:
http://www.asterisk.org/hello
asterisk-users mailing list
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