[Asterisk-Users] Help with sip setup because can't receive calls!!!!!!

Tue, 31 Jan 2006 04:36:28 -0800

Hi all, I am resending this message, so far no one has helped me with this incoming call issue. there is no problem with outbound call but there is no inbound call to my sip phone. the only message I get when I call from pstn is "unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' (case =0)". my configuration files are attached below. any help would be greatly appreciated. many thanks in advance.
ABC

abc def <[EMAIL PROTECTED]> wrote:
there is no error message coming up on the pbx for in-bound calls (there is only debugging messages for outbound calls).
 
thanks in advance for any hint or suggestion.
Ama
 
I just post my configuration file here for sip phone:
extensions.conf
-------------------------------------------------------------------------
[globals]
[default]
include => incoming
include => outgoing
include => iax
inculde => sip
include => sccp
[sip]
exten => 2171,1,Dial(SIP/stargate1,20)
;exten => 2171,1,Dial(SIP/2171,20)
exten => 2171,2,Hangup
exten => 2172,1,Dial(SIP/stargate2,20)
;exten => 2172,1,Dial(SIP/2172,20)
exten => 2172,2,Hangup
exten => 2173,1,Dial(SIP/stargate3,20)
;exten => 2173,1,Dial(SIP/2173,20)
exten => 2173,2,Hangup
[sccp]
[skinny]
[incoming]
exten => ; _214943[5-9]6,1,Dial(SIP/stargate3)
exten => _214943[5-9]6,2,Hangup
[outgoing]
exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN})
exten => _XXXXXXXX,2,Hangup
-------------------------------------------------------------------------
sip.conf
-------------------------------------------------------------------------
[general]
context=default                 ; Default context for incoming calls
                                ; Set this to your host name or domain name
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                               
register => stargate1:[EMAIL PROTECTED]/2171
register => stargate2:[EMAIL PROTECTED]/2172
register => stargate3:[EMAIL PROTECTED]/2173
;---------------------------------------------- NAT SUPPORT ------------------------
nat=no                         ; Global NAT settings  (Affects all peers and users)
                              

[local_sip]
type=friend
host=10.47.200.136
context=default
[stargate1] ;cisco 9760
;[2171]
type=friend
host=dynamic ;10.47.200.140 ;dynamic
defaultip=10.47.200.140
username=stargate1
secret=xxx
callerid="21495071" <2171>
allow=all
qualify=200
nat=no
defaultip=10.47.200.140

[stargate2] ;Polycom 601
;[2172]
type=friend
host=dynamic ;10.47.200.141  ;dynamic
defaultip=10.47.200.141
username=xxx
secret=2stargate
callerid="21495072" <2172>
allow=all
qualify=200
nat=no
defaultip=10.47.200.141
[stargate3] ;Aastra 480i
;[2173]
type=friend
host=dynamic ;10.47.200.137 ;dynamic
defaultip=10.47.200.137
username=stargate3
callerid="starg ate3" <2173>
secret=xxx
allow=all
qualify=200
nat=no
defaultip=10.47.200.137
----------------------------------------------------------------------------

[EMAIL PROTECTED] wrote:
What error do you get when trying to call the SIP phones?
 
PaulH
 
----- Original Message -----
From: abc def
Sent: Wednesday, January 25, 2006 11:58 PM
Subject: [Asterisk-Users] Help with sip setup because can't receive calls

Hi all,
I read many posts on asterisk mail site and been trying many different things but still I can't get my sip phones to work with asterisk.
  I have a full blown-up voip netwok with two asterisk servers connected
to pstn network with iax phones and cisco sccp phones which all work fine.
however, I have been struggeling to configure my sip phones (polycom 601, Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip phones to anywhere else but not receive phone calls. I can see the phones on "sip show registry" and "sip show peers" but no track phone calls for sip.
  
  can you please shed some light on me how to go about solving this
problem?
  
  thank you and best regards,
  Ama
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