[Asterisk-Users] New to asterisk {cisco's won't ring}
Jeremy Kenney
jeremy at ds-tech.com
Fri Jun 18 15:27:01 MST 2004
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Found user 'dstech3'
Looking for 104 in dstech3
Reliably Transmitting (NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.50:5060;received=68.61.52.152
From: "Anonymous" <sip:dstech3@:0>;tag=2110378357
To: <sip:104@:0;user=phone>;tag=as2f1fb147
Call-ID: 2450357176 at 192.168.0.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:104 at 69.39.68.99>
Content-Length: 0
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Nik Martin
Sent: Friday, June 18, 2004 5:57 PM
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] New to asterisk {cisco's won't ring}
Jeremy Kenney wrote:
> I am new to asterisk I just downloaded it I setup some extensions I
> can't seem to get them to ring I can get my ata 186 to register but
> having problems with getting the phones to ring when I dial an
> extention
>
> Extentions.conf
>
> [dstech4]
> exten=>104,1,Answer
> exten=>104,3,Goto(dstech4,104)
>
> [dstech3]
> exten=>103,1,Answer
> exten=>103,2,Goto(dstech3,104)
>
This looks like you are putting asterisk in a loop in your dial plan. If
you are using SIP, I'd expect to see something like:
[outgoing]
{Whatever outbound dialing rules you might need here}
Include=>default
[default]
Exten => 104,1,Dial(SIP/dstech4),20)
Exten => 104,2,Congestion
Exten => 103,1,Dial(SIP/dstech3),20)
Exten => 104,2,Congestion
sip.conf
[dstech3]
type=friend
username=dstech3
secret=mypassword
qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
host=dynamic
canreinvite=no
Context=outgoing <--- you need to put the phone in a context
These are simplified examples you'd want to enhance them with voicemail,
etc, but start here to at least get th ephones to ring and communicate.
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