[asterisk-users] Paging in Asterisk

[Paul Hales]

Does 'sip show peers' actually show the phone as registered?

PaulH


On Mon, 2007-10-15 at 02:05 +1000, bu at westnet.com.au wrote:
> Actually, forget everything else.
> 
> Even when I simply pick up the handset and dial 6600, I get those errors 
> in console, so it's not related to paging or call files or anything 
> special, I guess..
> 
> Any ideas?
> 
> bu
> 
> bu at westnet.com.au wrote:
> > Hi All,
> >
> > I've been trying to send a message to the list for the past 3 days, but 
> > I neither get bounces nor the message appearing in the list, so someone 
> > on IRC sugested I reply to an existing message.
> >
> > My subject is related to this message, although slightly different.
> >
> > Apologies if my actual messages appear in the list. Here's a paste of 
> > one of my past messages:
> > --------------------------------------
> > I'm playing with a PA-type setup, where people can dial a number, and 
> > Asterisk would place a call file to get another phone to dial in (auto 
> > answering) and play to it a sound.
> >
> > It's woking, but I'm getting some errors, as I'll paste below.
> >
> > So, my setup:
> >     Asterisk 1.4.13
> >     Debian GNU/Linux 4.0
> >     Linux Kernel 2.6.18-5-686
> >
> > SIP client:
> >     snom360 5.3 soft-phone
> >     SIP/pa at from-sip
> >
> > My call file:
> >     Channel: Local/pa at localtest/n
> >     Extension: 6600
> >
> > extensions.conf:
> >
> >     [from-sip]
> >     exten => 6600,1,Answer
> >     exten => 6600,n,Wait(1)
> >     exten => 6600,n,Playback(demo-thanks)
> >     exten => 6600,n,Hangup
> >    
> >     [localtest]
> >     exten => pa,1,SIPAddHeader(Call-Info:<sip:asterisk>\;answer-after=0)
> >     exten => pa,n,Dial(SIP/pa)
> >
> >
> > The Console (-rvvv):
> >
> >   -- Attempting call on Local/pa at localtest/n for 6600 at default:1 (Retry 1)
> >   -- Executing [pa at localtest:1] 
> > SIPAddHeader("Local/pa at localtest-f5d9,2", 
> > "Call-Info:<sip:asterisk>;answer-after=0") in new stack
> >   -- Executing [pa at localtest:2] Dial("Local/pa at localtest-f5d9,2", 
> > "SIP/pa") in new stack
> >   -- Called pa
> >   -- SIP/pa-081ddd30 is ringing
> >   -- SIP/pa-081ddd30 answered Local/pa at localtest-f5d9,2
> > == Starting Local/pa at localtest-f5d9,1 at default,6600,1 failed so 
> > falling back to exten 's'
> >   -- Executing [s at default:1] Wait("Local/pa at localtest-f5d9,1", "1") in 
> > new stack
> > [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for 
> > gsmtolin
> > [Oct 11 19:10:52] WARNING[3912]: translate.c:163 framein: no samples for 
> > gsmtolin
> > .
> > .
> > .
> > And it keeps going, really fast, indefinately, until hung up.
> >
> > Any ideas? How about the 's' error above?
> >
> > Thanks,
> > bu
> >
> >
> > Nick Couchman wrote:
> >   
> >> Our office does not have a PA system, and in our current phone system 
> >> we have a certain extension that we dial that pages over the speaker 
> >> of all the phones in the office.  Does Asterisk support this feature? 
> >>  If so, could someone tell me the best way to set this up in AsteriskNOW?
> >>
> >>
> >> Thanks,
> >>
> >> Nick
> >>
> >>     
> >
> >
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> 
> 
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