[Asterisk-Users] extension to extension dialing

Nora Lavelle nora at silverspringnet.com
Fri Jan 27 11:28:33 MST 2006 

Hmm.. I definitely have type=friend in the sip.conf and I added
qualify=yes but, I think that's default anyways.. When I call from the
outside and enter his extension it goes through to him fine but, when I
go extension to extension it automatically goes to voicemail.. Here are
the messages from the console:

    -- Executing Macro("SIP/130-58df", "stdexten|SIP/124") in new stack
    -- Executing Dial("SIP/130-58df", "SIP/124|20") in new stack
    -- Called 124
Jan 27 10:27:10 WARNING[28243]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call 7998a5e87b708f4374ca0ec212863b6d at 10.200.1.234
for seqno 102 (Critical Request)
  == No one is available to answer at this time
    -- Executing Goto("SIP/130-58df", "s-NOANSWER|1") in new stack
    -- Goto (macro-stdexten,s-NOANSWER,1)
    -- Executing VoiceMail("SIP/130-58df", "u124") in new stack
    -- Playing 'voicemail/default/124/greet' (language 'en')
Jan 27 10:27:10 NOTICE[28243]: sched.c:290 ast_sched_del: Attempted to
delete non-existant schedule entry 22838!
    -- Playing 'vm-isunavail' (language 'en')
    -- Playing 'vm-intro' (language 'en')

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary
Richardson
Sent: Thursday, January 26, 2006 6:48 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
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> >
> >
> >
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