[Asterisk-Users] extension to extension dialing

Colin Anderson ColinA at landmarkmasterbuilder.com
Thu Jan 26 20:30:05 MST 2006


Snom's don't care, port 2057 is fine. Can you ping each phone from the Linux
console?



-----Original Message-----
From: Gary Richardson [mailto:gary.richardson at gmail.com]
Sent: Thursday, January 26, 2006 7:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extension to extension dialing


In your sip.conf, make sure these phones have a Type=Friend entry and
a qualify=yes. I don't think the qualify=yes is required, but it helps
in debuging.

About the port, I'm not too sure about sipura and snom phones (I only
have Cisco phones :(). That could have something to do with it..

On 1/26/06, Nora Lavelle <nora at silverspringnet.com> wrote:
>
> Hi there gary. thanks so much for your help. we're using sipura-841 and
snom 320s.
>
> Here's the sip show peers.. that's weird that extension 130 has port
2057.. could that be the problem ?
>
> -nora
>
> Name/username    Host            Dyn Nat ACL Mask             Port
Status
>
> 201/201          10.200.0.56      D          255.255.255.255  5060
Unmonitor
> ed
> 130/130          10.200.0.10      D          255.255.255.255  2057
Unmonitor
> ed
> 129/129          10.200.0.5       D          255.255.255.255  5060
Unmonitor
> ed
> 127/127          10.201.0.30      D          255.255.255.255  5060
Unmonitor
> ed
> 126/126          10.201.0.29      D          255.255.255.255  5060
Unmonitor
> ed
> 125/125          10.201.0.35      D          255.255.255.255  5060
Unmonitor
> ed
> 124/124          10.201.0.31      D          255.255.255.255  5060
Unmonitor
> ed
> 102/102          10.200.0.48      D          255.255.255.255  5060
Unmonitor
> ed
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com on behalf of Gary Richardson
> Sent: Thu 1/26/2006 5:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] extension to extension dialing
>
> Check your error messages in you asterisk console. Perhaps your sip
> secret or caller id is broken?
>
> What type of phones are you using?
>
> On 1/26/06, Nora Lavelle <nora at silverspringnet.com> wrote:
> >
> >
> >
> > Sorry for all the newbie questions. I really appreciate everyone's help
> > today.
> >
> >
> >
> > Okay I've got outgoing and incoming calls working with no echo. yay! Now
I'm
> > having an issue with SIP extension to extension calling. Any time I dial
> > another extension it goes right into voice mail.  My extensions.conf is
> > pretty small and rough but, here's what I have right now. Most of it was
> > taken from the voip-info website. Any help as always VERY appreciated.
> >
> >
> >
> > Thanks again!
> >
> > Nora Lavelle
> >
> >
> >
> > # cat extensions.conf
> >
> > [incoming]
> >
> > exten => s,1,Answer();
> >
> > exten => s,2,Background(ssn-greeting);
> >
> > exten => *,1,Directory(default)
> >
> > exten => 205,1,Wait(2)
> >
> > exten => 205,2,Record(/tmp/asterisk-recording:gsm)
> >
> > exten => 205,3,Wait(2)
> >
> > exten => 205,4,Playback(/tmp/asterisk-recording)
> >
> > exten => 205,5,Wait(2)
> >
> > exten => 205,6,Hangup
> >
> >
> >
> > [internal]
> >
> > exten => 101,1,Macro(stdexten,SIP/101)
> >
> > exten => 102,1,Macro(stdexten,SIP/102)
> >
> > exten => 103,1,Macro(stdexten,SIP/103)
> >
> > exten => 123,1,Macro(stdexten,SIP/123)
> >
> > exten => 124,1,Macro(stdexten,SIP/124)
> >
> > exten => 125,1,Macro(stdexten,SIP/125)
> >
> > exten => 126,1,Macro(stdexten,SIP/126)
> >
> > exten => 127,1,Macro(stdexten,SIP/127)
> >
> > exten => 128,1,Macro(stdexten,SIP/128)
> >
> > exten => 129,1,Macro(stdexten,SIP/129)
> >
> > exten => 130,1,Macro(stdexten,SIP/130)
> >
> > exten => 135,1,Macro(stdexten,SIP/135)
> >
> > exten => 117,1,Macro(stdexten,SIP/117)
> >
> > exten => 201,1,Macro(stdexten,SIP/201)
> >
> >
> >
> > [voicemail]
> >
> > exten => 300,1,Answer
> >
> > exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
> >
> > exten => 300,3,Hangup
> >
> >
> >
> > [local]
> >
> > exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _9NXXXXXX,2,Congestion
> >
> >
> >
> > [longdistance]
> >
> > exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _91NXXNXXXXXX,2,Congestion
> >
> >
> >
> > [macro-stdexten]
> >
> > exten => s,1,Dial(${ARG1},20)
> >
> > exten => s,2,Goto(s-${DIALSTATUS},1)
> >
> > exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
> >
> > exten => s-NOANSWER,2,Goto(default,s,1)
> >
> > exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
> >
> > exten => s-BUSY,2,Goto(default,s,1)
> >
> > exten => s-.,1,Goto(s-NOANSWER,1)
> >
> > exten => a,1,VoicemailMain(${MACRO_EXTEN})
> >
> >
> >
> > [default]
> >
> > include => incoming
> >
> > include => internal
> >
> > include => voicemail
> >
> > include => local
> >
> > include => longdistance
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



More information about the asterisk-users mailing list