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Angus Comber
2005-Oct-28 14:46 UTC
[Asterisk-Users] Why can't I dial - just using SIP internally

Hello

I have setup a couple of sip accounts - here is my sip.conf:
context=default
disallow=all
allow=ulaw
allow=alaw
allow=gsm

[200]
username=200
type=friend
secret=1234
port=5060
nat=never
mailbox=200@default
dtmfmode=rfc2833
context=default
callerid="Angus" <200>
host=dynamic
insecure=very
group=1
callgroup=1
pickupgroup=1

[201]
username=201
type=friend
secret=1234
port=5060
nat=never
dtmfmode=rfc2833
context=default
callerid="Lisa" <201>
host=dynamic
insecure=very
group=1
callgroup=1
pickupgroup=1





my extensions.conf:

[frompstnanalog]
exten => 787367,1,Dial(SIP/200,1)
exten => 787367,2,Voicemail(su200)
exten => 787367,3,Hangup


[default]
;exten => _X.,1,Dial(ZAP/g1/${EXTEN},20,Ttm)
;exten => _X.,2,Hangup

exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten => _2XX,2,Voicemail(su${EXTEN})
exten => _2XX,3,Hangup

exten => *97,1,Answer
exten => *97,2,VoicemailMain(${CALLERIDNUM}@default)
exten => *97,3,Hangup


I have setup two IP phones, they register OK but cannot dial each other.  I
had to switch on sip debug to get anything on the asterisk console:

pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
INVITE sip:201@192.168.0.20;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-17vz3vxf4uhz;rport
From: "Angus" <sip:200@192.168.0.20>;tag=oa5ljlnorj
To: <sip:201@192.168.0.20;user=phone>
Call-ID: 3c2670cc1fbd-civgrs69z207@83-104-202-25
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:200@192.168.0.21:5060;line=exb2unjb>
P-Key-Flags: keys="3"
User-Agent: snom190-3.56m
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 342

v=0
o=root 2065976712 2065976712 IN IP4 192.168.0.21
s=call
c=IN IP4 192.168.0.21
t=0 0
m=audio 10000 RTP/AVP 0 8 3 18 4 9 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:9 g722/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

--- (18 headers 16 lines)---
Using INVITE request as basis request -
3c2670cc1fbd-civgrs69z207@83-104-202-25
Sending to 192.168.0.21 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.0.21:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.0.21:5060;branch=z9hG4bK-17vz3vxf4uhz;rport;received=192.168.0.21
From: "Angus" <sip:200@192.168.0.20>;tag=oa5ljlnorj
To: <sip:201@192.168.0.20;user=phone>;tag=as7203b20e
Call-ID: 3c2670cc1fbd-civgrs69z207@83-104-202-25
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:201@192.168.0.20>
Proxy-Authenticate: Digest realm="asterisk",
nonce="24b5d1a5"
Content-Length: 0


---
Scheduling destruction of call '3c2670cc1fbd-civgrs69z207@83-104-202-25'
in
15000 ms
Found user '200'
pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
ACK sip:201@192.168.0.20;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-17vz3vxf4uhz;rport
From: "Angus" <sip:200@192.168.0.20>;tag=oa5ljlnorj
To: <sip:201@192.168.0.20;user=phone>;tag=as7203b20e
Call-ID: 3c2670cc1fbd-civgrs69z207@83-104-202-25
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:200@192.168.0.21:5060;line=exb2unjb>
Content-Length: 0


--- (9 headers 0 lines)---
pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
INVITE sip:201@192.168.0.20;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-ass0mb36ivsw;rport
From: "Angus" <sip:200@192.168.0.20>;tag=oa5ljlnorj
To: <sip:201@192.168.0.20;user=phone>
Call-ID: 3c2670cc1fbd-civgrs69z207@83-104-202-25
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:200@192.168.0.21:5060;line=exb2unjb>
P-Key-Flags: keys="3"
User-Agent: snom190-3.56m
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Proxy-Authorization: Digest
username="200",realm="asterisk",nonce="24b5d1a5",uri="sip:201@192.168.0.20;user=phone",response="a5598b627eb4c3bad2084bd553daad3f",algorithm=md5
Content-Type: application/sdp
Content-Length: 342

v=0
o=root 2065976712 2065976712 IN IP4 192.168.0.21
s=call
c=IN IP4 192.168.0.21
t=0 0
m=audio 10000 RTP/AVP 0 8 3 18 4 9 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:9 g722/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

--- (19 headers 16 lines)---
Using INVITE request as basis request -
3c2670cc1fbd-civgrs69z207@83-104-202-25
Sending to 192.168.0.21 : 5060 (non-NAT)
Found user '200'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 9
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.21:10000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format g729
Found description format g723
Found description format g722
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10f
(g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 201 in default (domain 192.168.0.20)
Reliably Transmitting (no NAT) to 192.168.0.21:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.0.21:5060;branch=z9hG4bK-ass0mb36ivsw;rport;received=192.168.0.21
From: "Angus" <sip:200@192.168.0.20>;tag=oa5ljlnorj
To: <sip:201@192.168.0.20;user=phone>;tag=as7203b20e
Call-ID: 3c2670cc1fbd-civgrs69z207@83-104-202-25
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:201@192.168.0.20>
Content-Length: 0


---
pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
ACK sip:201@192.168.0.20;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-ass0mb36ivsw;rport
From: "Angus" <sip:200@192.168.0.20>;tag=oa5ljlnorj
To: <sip:201@192.168.0.20;user=phone>;tag=as7203b20e
Call-ID: 3c2670cc1fbd-civgrs69z207@83-104-202-25
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:200@192.168.0.21:5060;line=exb2unjb>
Content-Length: 0


--- (9 headers 0 lines)---
Destroying call '3c2670cc1fbd-civgrs69z207@83-104-202-25'
pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
SUBSCRIBE sip:200@192.168.0.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-rx3fpn1gi0jd;rport
From: <sip:200@192.168.0.20>;tag=8n7jwspbjv
To: <sip:200@192.168.0.20>
Call-ID: 3c26700bb98c-j7ouu086ymgy@83-104-202-25
CSeq: 3 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:200@192.168.0.21:5060;line=exb2unjb>
Event: message-summary
Accept: application/simple-message-summary
Expires: 3600
Content-Length: 0


--- (12 headers 0 lines)---
Using latest SUBSCRIBE request as basis request
Sending to 192.168.0.21 : 5060 (non-NAT)
Found peer '200'
Looking for 200 in default (domain 192.168.0.20)
Transmitting (no NAT) to 192.168.0.21:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.0.21:5060;branch=z9hG4bK-rx3fpn1gi0jd;rport;received=192.168.0.21
From: <sip:200@192.168.0.20>;tag=8n7jwspbjv
To: <sip:200@192.168.0.20>;tag=as7a1c09da
Call-ID: 3c26700bb98c-j7ouu086ymgy@83-104-202-25
CSeq: 3 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:200@192.168.0.20>
Content-Length: 0


---
Destroying call '3c26700bb98c-j7ouu086ymgy@83-104-202-25'
pbx*CLI>


I can't seem to find anything in this output pointing to the problem!  Can
anyone help?

Angus
asterisk users - Oct 2005 - Why can't I dial - just using SIP internally

[Asterisk-Users] Why can't I dial - just using SIP internally
