Ekiga sip client and Asterisk wont work

Hi,
Again me :slight_smile: I have issue with Ekiga 4.0.1. Client are properly register with Asterisk but I cant make any call.

INVITE sip:555 SIP/2.0
CSeq: 2 INVITE
v: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bKacb32ac6-af10-e811-99ea-466c2fb0ad09;rport
User-Agent: Ekiga/4.0.1
Authorization: Digest username=“1-100”, realm=“asterisk”, nonce=“7a6b2ad7”, uri=“sip:555”, algorithm=MD5, response="cd15eda9178787bb114a5c8b30d913d9"
f: sip:tomasz@192.168.100.10;tag=b6a925c6-af10-e811-99ea-466c2fb0ad09
i: 4ead25c6-af10-e811-99ea-466c2fb0ad09@desktop
k: 100rel,replaces
t: sip:555
m: sip:tomasz@192.168.100.10
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
l: 871
c: application/sdp
Max-Forwards: 70

v=0
o=- 1518693576 1 IN IP4 192.168.100.10
s=Ekiga/4.0.1
c=IN IP4 192.168.100.10
t=0 0
m=audio 5074 RTP/AVP 116 0 8 101
a=sendrecv
a=rtpmap:116 Speex/16000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=maxptime:20
m=video 5076 RTP/AVP 90 31 34 99 103 102 104
b=AS:4096
b=TIAS:4096000
a=sendrecv
a=rtpmap:90 theora/90000
a=fmtp:90 height=576;width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
a=rtpmap:34 H263/90000
a=fmtp:34 F=1;CIF=1;CIF4=1;QCIF=1
a=rtpmap:99 H263-1998/90000
a=fmtp:99 D=1;F=1;I=1;J=1;CIF=1;CIF4=1;QCIF=1
a=rtpmap:103 H264/90000
a=fmtp:103 max-fs=6336;max-mbps=190080;profile-level-id=42801e
a=rtpmap:102 H264/90000
a=fmtp:102 packetization-mode=1;max-fs=6336;max-mbps=190080;profile-level-id=42801e
a=rtpmap:104 MP4V-ES/90000
a=fmtp:104 profile-level-id=5
<------------->
— (14 headers 31 lines) —
Sending to 192.168.100.10:5060 (no NAT)
Using INVITE request as basis request - 4ead25c6-af10-e811-99ea-466c2fb0ad09@desktop
Found peer ‘1-100’ for ‘tomasz’ from 192.168.100.10:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 116
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format Speex for ID 116
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 90
Found RTP video format 31
Found RTP video format 34
Found RTP video format 99
Found RTP video format 103
Found RTP video format 102
Found RTP video format 104
Found video description format h261 for ID 31
Found video description format H263 for ID 34
Found video description format H263-1998 for ID 99
Found video description format H264 for ID 103
Found video description format H264 for ID 102
Found video description format MP4V-ES for ID 104
Capabilities: us - (g729|ilbc|gsm|ulaw|alaw), peer - audio=(ulaw|alaw|speex16)/video=(h261|h263|h263p|h264|mpeg4)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.10:5074
Looking for s in vhz8xkn (domain 555)

<— Reliably Transmitting (no NAT) to 192.168.100.10:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.100.10:5060;branch=z9hG4bKacb32ac6-af10-e811-99ea-466c2fb0ad09;received=192.168.100.10;rport=5060
From: sip:tomasz@192.168.100.10;tag=b6a925c6-af10-e811-99ea-466c2fb0ad09
To: sip:555;tag=as62638a2f
Call-ID: 4ead25c6-af10-e811-99ea-466c2fb0ad09@desktop
CSeq: 2 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[Feb 15 01:19:36] NOTICE[14948][C-00000016]: chan_sip.c:26273 handle_request_invite: Call from ‘1-100’ (192.168.100.10:5060) to extension ‘555’ rejected because extension not found in context ‘vhz8xkn’.

voip01*CLI> dialplan show vhz8xkn

[ Context ‘vhz8xkn’ created by ‘pbx_config’ ]
‘555’ => 1. noop(start) [pbx_config]
2. Noop($EXTEN}) [pbx_config]
‘start’ => 1. Noop(Startuje z programem ${CONTEXT}) [pbx_config]
2. playback(28) [pbx_config]

-= 2 extensions (4 priorities) in 1 context. =-

I see what I did wrong. I have to call to number@domain instead just number

The reason is that a URI of “sip:555” parses as URI with scheme of ‘sip’ and hostname of ‘555’ so the extension then defaults to ‘s’.