[asterisk-users] Trying to dial out on teliax
BSumrall
Brads at ftnco.com
Tue May 29 05:02:04 MST 2007
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4060e
(gsm|ulaw|alaw|speex|ilbc|h261)/video=0x40000 (h261), combined - 0x40e
(gsm|ulaw|alaw|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 66.176.193.46:5016
Looking for 9158332042 in home (domain 66.109.17.92)
<--- Reliably Transmitting (no NAT) to 66.176.193.46:5081 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
66.176.193.46:5081;branch=z9hG4bKc91e2033-ccf8-1810-9367-0013d3ee21fe;receiv
ed=66.176.193.46;rport=5081
From: "Brad S"
<sip:204 at 66.109.17.92>;tag=44002033-ccf8-1810-9363-0013d3ee21fe
To: <sip:9158332042 at 66.109.17.92>;tag=as06d3acaa
Call-ID: 2afa1f33-ccf8-1810-9363-0013d3ee21fe at usmc
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'2afa1f33-ccf8-1810-9363-0013d3ee21fe at usmc' in 32000 ms (Method: INVITE)
<--- SIP read from 66.176.193.46:5081 --->
ACK sip:9158332042 at 66.109.17.92 SIP/2.0
CSeq: 2 ACK
Via: SIP/2.0/UDP
66.176.193.46:5081;branch=z9hG4bKc91e2033-ccf8-1810-9367-0013d3ee21fe;rport
From: "Brad S"
<sip:204 at 66.109.17.92>;tag=44002033-ccf8-1810-9363-0013d3ee21fe
Call-ID: 2afa1f33-ccf8-1810-9363-0013d3ee21fe at usmc
To: <sip:9158332042 at 66.109.17.92>;tag=as06d3acaa
Proxy-Authorization: Digest username="204", realm="asterisk",
nonce="1a3db830", uri="sip:9158332042 at 66.109.17.92", algorithm=md5,
response="cb7cedabcc0b4c20d2b948d05f67218c"
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
Content-Length: 0
Max-Forwards: 70
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of BSumrall
Sent: Tuesday, May 29, 2007 3:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Trying to dial out on teliax
Trying to launch my first dial out to Teliax and getting this error
[May 29 03:08:06] WARNING[1955]: pbx.c:4644 add_pri: Unable to register
extension '204', priority 2 in 'brad', already in use
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[default]
exten => 800xxxxxxx,1,Answer()
exten => 800xxxxxxx,2,Wait(1)
exten => 800xxxxxxx,3,Queue(service|t||8005181896|45)
exten => 800xxxxxxx,1,Answer()
exten => 800xxxxxxx,2,Wait(1)
exten => 800xxxxxxx,3,AgentLogin()
exten => h,1,DeadAGI(postqueue.agi)
[brad]
exten => 204,1,Wait()
exten => 204,2,Answer
exten => 204,3,Playback(demo-congrats)
exten => 204,4,Hangup
exten => 204,2,Dial(Zap/g2,20)
;exten => 204,Voivemail(u100)
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