[Asterisk-Users] extension to extension dialing

[Nora Lavelle]

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
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