rendered paste body*CLI> sip show peer 5601
* Name : 5601
Secret : <Set>
MD5Secret : <Not set>
Context : numberplan-custom-1
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup : 1
Pickupgroup : 1
Mailbox : 5601
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 0
Dynamic : Yes
Callerid : "Alexandre Fiori" <>
MaxCallBR : 384 kbps
Expire : 1638
Insecure : no
Nat : RFC3581
ACL : No
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 10.0.5.16 Port 5061
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 5601
SIP Options : (none)
Codecs : 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264)
Codec Order : (none)
Auto-Framing: No
Status : Unmonitored
Useragent : Ekiga/2.0.11
Reg. Contact : sip:5601@10.0.5.16:5061;transport=udp
here's the call debug from sip:
<--- SIP read from 10.0.5.16:5072 --->
INVITE sip:6878@10.0.0.7 SIP/2.0
Date: Thu, 31 Jan 2008 13:23:29 GMT
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.0.5.16:5072;branch=z9hG4bK68192661-6dce-dc11-90e6-0016d393c2e6;rport
User-Agent: Ekiga/2.0.11
From: "Alexandre Fiori" <sip:5601@10.0.0.7>;tag=16762561-6dce-dc11-90e6-0016d393c2e6
Call-ID: 5c712561-6dce-dc11-90e6-0016d393c2e6@kong
To: <sip:6878@10.0.0.7>
Contact: <sip:5601@10.0.5.16:5061;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Content-Length: 246
Max-Forwards: 70
v=0
o=- 1201785809 1201785809 IN IP4 10.0.5.16
s=Opal SIP Session
c=IN IP4 10.0.5.16
t=0 0
m=audio 5036 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 5038 RTP/AVP 31
a=rtpmap:31 H261/90000
<------------->
--- (13 headers 11 lines) ---
Sending to 10.0.5.16 : 5072 (NAT)
Using INVITE request as basis request - 5c712561-6dce-dc11-90e6-0016d393c2e6@kong
Found no matching peer or user for '10.0.5.16:5072'
Found RTP audio format 3
Found RTP audio format 101
Found RTP video format 31
Peer audio RTP is at port 10.0.5.16:5036
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Found unknown media description format H261 for ID 31
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.5.16:5036
Looking for 6878 in default (domain 10.0.0.7)
<--- Reliably Transmitting (NAT) to 10.0.5.16:5072 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.5.16:5072;branch=z9hG4bK68192661-6dce-dc11-90e6-0016d393c2e6;received=10.0.5.16;rport=5072
From: "Alexandre Fiori" <sip:5601@10.0.0.7>;tag=16762561-6dce-dc11-90e6-0016d393c2e6
To: <sip:6878@10.0.0.7>;tag=as24ab869c
Call-ID: 5c712561-6dce-dc11-90e6-0016d393c2e6@kong
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Jan 31 11:21:34] NOTICE[20792]: chan_sip.c:13859 handle_request_invite: Call from '' to extension '6878' rejected because extension not found.
Scheduling destruction of SIP dialog '5c712561-6dce-dc11-90e6-0016d393c2e6@kong' in 32000 ms (Method: INVITE)
<--- SIP read from 10.0.5.16:5072 --->
ACK sip:6878@10.0.0.7 SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.0.5.16:5072;branch=z9hG4bK68192661-6dce-dc11-90e6-0016d393c2e6;rport
From: "Alexandre Fiori" <sip:5601@10.0.0.7>;tag=16762561-6dce-dc11-90e6-0016d393c2e6
Call-ID: 5c712561-6dce-dc11-90e6-0016d393c2e6@kong
To: <sip:6878@10.0.0.7>;tag=as24ab869c
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE
Content-Length: 0
Max-Forwards: 70
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '5c712561-6dce-dc11-90e6-0016d393c2e6@kong' Method: ACK
here's users.conf:
[5601]
callwaiting = yes
context = numberplan-custom-1
fullname = Alexandre Fiori
hasagent = no
hasdirectory = no
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
deletevoicemail = no
host = dynamic
mailbox = 5601
secret = 5601
threewaycalling = yes
vmsecret = 5601
registeriax = no
registersip = yes
canreinvite = no
nat = no
dtmfmode = rfc2833
disallow = all
allow = all
here's extensions.conf:
;if I uncomment this and call 123 it works.
;[default]
;exten => 123,1,Playback(busy)
[internal]
exten => _68XX,1,Dial(h323/${EXTEN}@CCM-CRU,25,wWtT)
[numberplan-custom-1]
include = internal
plancomment = interal
parked = no