[Asterisk-Users] Can't pickup call when dialing *8 extension (resent)

[hgaillac-sip at yahoo.fr]

*8 is coded in res_features.so .
What are the right extension to dial for pickup  calls
between  sip<=>sip or zap<=>sip ... 

Harry
--- Rich Adamson <radamson at routers.com> a écrit :

> You might have to use *8#. At least I do with my
> 7960.
> 
> ------------------------
> 
> > I added callgroup=1 and pickupgroup=1 for sip
> channels
> > however I can't pickup a call (see below ) between
> sip
> > phones when i dial *8 .
> > 
> > May I have to add app_pickup to solve this
> problem.
> > I use asterisk-1.2
> > 
> > Regards
> > Harry
> > 
> > 
> > serveur1*CLI>
> > <-- SIP read from 80.119.8.167:5060:
> > ACK sip:*8 at nxs.yi.org:5050 SIP/2.0
> > Via: SIP/2.0/UDP
> > 80.119.8.167;branch=z9hG4bKe1bb.87855e92.0
> > From: "alice"
> <sip:85 at nxs.yi.org>;tag=AF3B88E-55239161
> > Call-ID: b16b7b62-c85b30e0-5fdbcb3b at
192.168.0.20
> > To: <sip:*8 at nxs.yi.org>;tag=as543ba455
> > CSeq: 2 ACK
> > User-Agent: Sip EXpress router(0.9.4 (i386/linux))
> > Content-Length: 0
> > 
> > --- (8 headers 0 lines)---
> > Destroying call
> > 'b16b7b62-c85b30e0-5fdbcb3b at 192.168.0.20'
> >     -- Nobody picked up in 10000 ms
> > Reliably Transmitting (NAT) to 80.119.8.167:5060:
> > CANCEL sip:86 at 192.168.0.21 SIP/2.0
> > Via: SIP/2.0/UDP
> > 80.119.8.167:5050;branch=z9hG4bK60e70916;rport
> > From: "alice"
> > <sip:84 at 80.119.8.167:5050>;tag=as7cefba23
> > To: <sip:86 at 192.168.0.21>
> > Contact: <sip:84 at 80.119.8.167:5050>
> > Call-ID:
> 50b2bf516e9f43a5415036b700b0e075 at 80.119.8.167
> > CSeq: 102 CANCEL
> > User-Agent: Asterisk PBX
> > Max-Forwards: 70
> > Content-Length: 0
> > 


	

	
		
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