Hello:
I’m newbie in asterisk, please help me.
My context is as follows:
192.168.4.2 --> Asterisk 11.13.1 complied from source
192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway
When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension 7777 (configured as a hotline on TG100) to asterisk server, but asterisk server sends me “SIP/2.0 401 Unauthorized” response, I think it’s a matter of contexts but I didn’t find the problem.
Below are sip.conf, extensions.conf and debug from 192.168.4.4 (TG100 GSM gateway).
Thanks in advance.
SIP.CONF
[general]
context = incoming-call
allowguest = yes
srvlookup = no
udpbindaddr = 0.0.0.0
tcpenable = no
qualify = yes
office
type = friend
host = dynamic
context = from-office-call
dtmfmode = auto
disallow = all
allow = g722
allow = alaw
allow = ulaw
201
description = grandstream-gxp2160
secret = 201
202
description = grandstream-dp715
secret = 202
203
description = grandstream-dp710
secret = 203
301
description = grandstream-gxp2130
secret = 301
401
description = grandstream-gxp2160
secret = 401
[11111111]
description = audiocodes-fxo-port5
type = friend
host = 192.168.4.3
secret = 11111111
context = incoming-call
canreinvite = no
dtmfmode = auto
disallow = all
allow = g722
allow = alaw
allow = ulaw
[555555555]
description = yeastar-neogate-tg100
type = friend
host = 192.168.4.4
secret = 555555555
context = incoming-call
canreinvite = no
dtmfmode = auto
disallow = all
allow = g722
allow = alaw
allow = ulaw
EXTENSIONS.CONF
[globals]
[incoming-call]
exten => _11111111,1,Goto(main-menu,start,1)
exten => _555555555,1,Goto(main-menu,start,1)
exten => _7777,1,Goto(main-menu,start,1)
[outgoing-call]
exten => _[24]XXXXXXX,1,Dial(SIP/${EXTEN}@11111111)
exten =>_09XXXXXXX,1,Dial(SIP/${EXTEN}@555555555)
[from-office-call]
exten => 0,1,Goto(main-menu,start,1)
exten => 201,1,Dial(SIP/201)
exten => 202,1,Dial(SIP/202)
exten => 203,1,Dial(SIP/203)
exten => 301,1,Dial(SIP/301)
exten => 401,1,Dial(SIP/401)
include => outgoing-call
[main-menu]
exten => start,1,Answer()
same => Wait(5)
same => n,Background(enter-ext-of-person)
same => n,WaitExten(20)
exten => 1,1,Dial(SIP/201,20)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => 2,1,Dial(SIP/202,20)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => 3,1,Dial(SIP/203,20)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => 4,1,Dial(SIP/301,20)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => 5,1,Dial(SIP/401,20)
same => n,Playback(vm-nobodyavail)
same => n,Hangup()
exten => i,1,Playback(pbx-invalid)
same => n,Goto(main-menu,start,1)
exten => t,1,Playback(vm-goodbye)
same => n,Hangup()
DEBUG
<— SIP read from UDP:192.168.4.4:5060 —>
INVITE sip:7777@192.168.4.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport
Max-Forwards: 70
From: “999999999” sip:999999999@192.168.4.4;tag=as67354416
To: sip:7777@192.168.4.2:5060
Contact: sip:999999999@192.168.4.4
Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
CSeq: 102 INVITE
User-Agent: TG100
Date: Wed, 12 Nov 2014 10:13:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 1426707418 1426707418 IN IP4 192.168.4.4
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.4.4
t=0 0
m=audio 10048 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (14 headers 13 lines) —
Sending to 192.168.4.4:5060 (no NAT)
Sending to 192.168.4.4:5060 (no NAT)
Using INVITE request as basis request - 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
Found peer ‘555555555’ for ‘999999999’ from 192.168.4.4:5060
<— Reliably Transmitting (no NAT) to 192.168.4.4:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;received=192.168.4.4;rport=5060
From: “999999999” sip:999999999@192.168.4.4;tag=as67354416
To: sip:7777@192.168.4.2:5060;tag=as16de6e5c
Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
CSeq: 102 INVITE
Server: Asterisk PBX 11.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="72011a6b"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘12663beb04ae514f10c4b3a145368d5c@192.168.4.4’ in 6400 ms (Method: INVITE)
<— SIP read from UDP:192.168.4.4:5060 —>
ACK sip:7777@192.168.4.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.4:5060;branch=z9hG4bK0b7c4652;rport
Max-Forwards: 70
From: “999999999” sip:999999999@192.168.4.4;tag=as67354416
To: sip:7777@192.168.4.2:5060;tag=as16de6e5c
Contact: sip:999999999@192.168.4.4
Call-ID: 12663beb04ae514f10c4b3a145368d5c@192.168.4.4
CSeq: 102 ACK
User-Agent: TG100
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘6e9eab843b74d0860b108ed13a5d22c9@192.168.4.4’ Method: OPTIONS
Really destroying SIP dialog ‘12663beb04ae514f10c4b3a145368d5c@192.168.4.4’ Method: ACK
uc*CLI>