Hi everybody,
I have some problems making calls from a sip user (HT286) to the pstn trough
Digium Wildcard TE110P, i allways have an error : SIP 403
INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd
From: "test"
<sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79
To: <sip:0170708959@192.168.1.4;user=phone>
Contact: <sip:4000@192.168.50.1;user=phone>
Call-ID: 9663c234cb61f1a2@192.168.50.1
CSeq: 8618 INVITE
User-Agent: Grandstream HT286 1.0.5.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 223
v=0
o=4000 8000 8000 IN IP4 192.168.50.1
s=SIP Call
c=IN IP4 192.168.50.1
t=0 0
m=audio 5004 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
12 headers, 11 lines
Using latest request as basis request
Sending to 192.168.50.1 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd
From: "test"
<sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79
To: <sip:0170708959@192.168.1.4;user=phone>;tag=as6d84bb7a
Call-ID: 9663c234cb61f1a2@192.168.50.1
CSeq: 8618 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0170708959@192.168.1.4>
Proxy-Authenticate: Digest realm="asterisk",
nonce="2bc58039"
Content-Length: 0
to 192.168.50.1:5060
Scheduling destruction of call '9663c234cb61f1a2@192.168.50.1' in 15000
ms
Found user '4000'
sipserver*CLI>
Sip read:
ACK sip:0170708959@192.168.1.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd
From: "test"
<sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79
To: <sip:0170708959@192.168.1.4;user=phone>;tag=as6d84bb7a
Contact: <sip:4000@192.168.50.1;user=phone>
Call-ID: 9663c234cb61f1a2@192.168.50.1
CSeq: 8618 ACK
User-Agent: Grandstream HT286 1.0.5.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
11 headers, 0 lines
sipserver*CLI>
Sip read:
INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bKa87a89a0afb2ac73
From: "test"
<sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79
To: <sip:0170708959@192.168.1.4;user=phone>
Contact: <sip:4000@192.168.50.1;user=phone>
Proxy-Authorization: DIGEST username="4000",
realm="asterisk",
algorithm=MD5, uri="sip:0170708959@192.168.1.4;user=phone",
nonce="2bc58039",
response="c1a4fc068a553db1091ce7a4d94d3ffe"
Call-ID: 9663c234cb61f1a2@192.168.50.1
CSeq: 8619 INVITE
User-Agent: Grandstream HT286 1.0.5.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 223
v=0
o=4000 8000 8000 IN IP4 192.168.50.1
s=SIP Call
c=IN IP4 192.168.50.1
t=0 0
m=audio 5004 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
13 headers, 11 lines
Using latest request as basis request
Sending to 192.168.50.1 : 5060 (non-NAT)
Found user '4000'I>
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.50.1:5004
Found description format PCMA
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
Looking for 0170708959 in from-internal
list_route: hop: <sip:4000@192.168.50.1;user=phone>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bKa87a89a0afb2ac73
From: "test"
<sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79
To: <sip:0170708959@192.168.1.4;user=phone>
Call-ID: 9663c234cb61f1a2@192.168.50.1
CSeq: 8619 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0170708959@192.168.1.4>
Content-Length: 0
to 192.168.50.1:5060
-- Executing Dial("SIP/4000-784c", "Zap/g1/170708959")
in new stack
-- Making new call for cr 32781>Protocol Discriminator: Q.931 (8) len=32
>Call Ref: len= 2 (reference 13/0xD) (Originator)
>Message type: SETUP (5)
>[04 03 80 90 a3]
>Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
>capability: Speech (0)
> Ext: 1 Trans mode/rate: 64kbps, circuit-mode
>(16)
> Ext: 1 User information layer 1: A-Law (35)
>[18 03 a1 83 81]
>Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred
>Dchan: 0
> ChanSel: Reserved
> Ext: 1 Coding: 0 Number Specified Channel Type:
>3
> Ext: 1 Channel: 1 ]
>[6c 02 00 c3]
>Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
>Unknown Number Plan (0)
> Presentation: Number not available (67)
'' ]
>[70 0a a1 31 37 30 37 30 38 39 35 39]
>Called Number (len=12) [ Ext: 1 TON: National Number (2) NPI:
>ISDN/Telephony Numbering Plan (E.164/E.163) (1) '170708959' ]
>[a1]
>Sending Complete (len= 1)
-- Called g1/170708959
< Protocol Discriminator: Q.931 (8) len=9
< Call Ref: len= 2 (reference 32781/0x800D) (Terminator)
< Message type: RELEASE COMPLETE (90)
< [08 02 80 b2]
< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
User (0)
< Ext: 1 Cause: Facility not subscribed (50), class =
Service or Option not Available (3) ]
-- Processing IE 8 (cs0, Cause)
-- Channel 0/1, span 1 got hangup
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
-- Executing Hangup("SIP/4000-784c", "") in new stack
== Spawn extension (from-internal, 0170708959, 2) exited non-zero on
'SIP/4000-784c'
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bKa87a89a0afb2ac73
From: "test"
<sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79
To: <sip:0170708959@192.168.1.4;user=phone>;tag=as45291b29
Call-ID: 9663c234cb61f1a2@192.168.50.1
CSeq: 8619 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:0170708959@192.168.1.4>
Content-Length: 0
to 192.168.50.1:5060
sipserver*CLI>
Sip read:
ACK sip:0170708959@192.168.1.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bKa87a89a0afb2ac73
From: "test"
<sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79
To: <sip:0170708959@192.168.1.4;user=phone>;tag=as45291b29
Contact: <sip:4000@192.168.50.1;user=phone>
Proxy-Authorization: DIGEST username="4000",
realm="asterisk",
algorithm=MD5, uri="sip:0170708959@192.168.1.4;user=phone",
nonce="2bc58039",
response="ef5381582f5b9280ac14a94032c98b74"
Call-ID: 9663c234cb61f1a2@192.168.50.1
CSeq: 8619 ACK
User-Agent: Grandstream HT286 1.0.5.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
12 headers, 0 lines
Destroying call '9663c234cb61f1a2@192.168.50.1'
I don't know what to do, i tried all possible configurations with no
success, some help or guidance will be very great, thanks.
Everything is ok on the ZAp channel:
sipserver*CLI> pri show span 1
Primary D-channel: 16
Status: Provisioned, Up, Active
Switchtype: EuroISDN
Type: Network
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
Extensions.conf:
[general]
static=yes
writeprotect=no
[globals]
EMERGENCY=0
EMERGENCY_TRUNK=Zap/17
EMERGENCY_NUM=some_test_phone_number
MAX_RING_TIME = 20
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g1 ; Local Trunk interface
TRUNKMSD=0 ; MSD digits to strip
(usually 1 or 0)
[from-internal]
exten => _0X.,1,Dial(Zap/g1/${EXTEN:1})
exten => _0X.,2,Hangup
Sip.conf:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
[4000]
username=4000
type=friend
secret=test
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=never
mailbox=4000@default
host=dynamic
dtmfmode=info
context=from-internal
callgroup=1
canreinvite=no
callerid=test
disallow=all
allow=ulaw
allow=alaw
Thanks you for your help:
Ihsane.
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