Softphone (Ekiga) to Asterisk

Asterisk Asterisk Support
1

Hi, im new to asterisk and i am trying to get a simple system working. Any ideas why i get “busy here” when i try to make a call to asterisk from Ekiga?

[code]<— SIP read from UDP:192.168.1.66:5060 —>
INVITE sip:101@192.168.1.68 SIP/2.0
CSeq: 1 INVITE
v: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK770852a8-b00a-1910-9a6b-90e6badafd43;rport
User-Agent: Ekiga/4.0.1
f: “peter” sip:101@192.168.1.68;tag=16ff51a8-b00a-1910-9a6a-90e6badafd43
i: 7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC
k: 100rel,replaces
t: sip:101@192.168.1.68
m: “peter” sip:101@192.168.1.66:5060
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
l: 1362
c: application/sdp
Max-Forwards: 70

v=0
o=- 1377169122 1 IN IP4 192.168.1.66
s=Ekiga/4.0.1
c=IN IP4 192.168.1.66
t=0 0
m=audio 5094 RTP/AVP 3 93 100 0 8 99 114 9 92 102 103 94 107 106 105 104 101
a=sendrecv
a=rtpmap:3 gsm/8000/1
a=rtpmap:93 Speex/16000/1
a=rtpmap:100 CELT/48000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:99 CELT/32000/1
a=rtpmap:114 iLBC/8000/1
a=fmtp:114 mode=20
a=rtpmap:9 G722/8000/1
a=rtpmap:92 G7221/16000/1
a=fmtp:92 bitrate=24000
a=rtpmap:102 G7221/16000/1
a=fmtp:102 bitrate=32000
a=rtpmap:103 AMR-WB/16000/1
a=fmtp:103 octet-align=1
a=rtpmap:94 Speex/8000/1
a=rtpmap:107 G726-16/8000/1
a=rtpmap:106 G726-24/8000/1
a=rtpmap:105 G726-32/8000/1
a=rtpmap:104 G726-40/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=maxptime:30
m=video 5096 RTP/AVP 91 31 34 110 113 112 118
b=AS:4096
b=TIAS:4096000
a=sendrecv
a=rtpmap:91 theora/90000
a=fmtp:91 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:110 H263-1998/90000
a=fmtp:110 D=1;F=1;I=1;J=1;CIF=1;CIF4=1;QCIF=1
a=rtpmap:113 H264/90000
a=fmtp:113 max-fs=6336;max-mbps=190080;profile-level-id=42801e
a=rtpmap:112 H264/90000
a=fmtp:112 packetization-mode=1;max-fs=6336;max-mbps=190080;profile-level-id=42801e
a=rtpmap:118 MP4V-ES/90000
a=fmtp:118 profile-level-id=5
<------------->
— (13 headers 48 lines) —
Sending to 192.168.1.66:5060 (NAT)
Using INVITE request as basis request - 7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC
Found peer ‘101’ for ‘101’ from 192.168.1.66:5060

<— Reliably Transmitting (no NAT) to 192.168.1.66:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK770852a8-b00a-1910-9a6b-90e6badafd43;received=192.168.1.66;rport=5060
From: “peter” sip:101@192.168.1.68;tag=16ff51a8-b00a-1910-9a6a-90e6badafd43
To: sip:101@192.168.1.68;tag=as133a4159
Call-ID: 7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC
CSeq: 1 INVITE
Server: Asterisk PBX SVN-branch-1.8-r396427
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="315ecac5"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.66:5060 —>
ACK sip:101@192.168.1.68 SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK770852a8-b00a-1910-9a6b-90e6badafd43;rport
From: “peter” sip:101@192.168.1.68;tag=16ff51a8-b00a-1910-9a6a-90e6badafd43
Call-ID: 7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC
To: sip:101@192.168.1.68;tag=as133a4159
Content-Length: 0
Max-Forwards: 70

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.1.66:5060 —>
INVITE sip:101@192.168.1.68 SIP/2.0
CSeq: 2 INVITE
v: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK871652a8-b00a-1910-9a6c-90e6badafd43;rport
User-Agent: Ekiga/4.0.1
Authorization: Digest username=“101”, realm=“asterisk”, nonce=“315ecac5”, uri="sip:101@192.168.1.68", algorithm=MD5, response="86204d1e9a70d9bf29966616e0ba8383"
f: “peter” sip:101@192.168.1.68;tag=16ff51a8-b00a-1910-9a6a-90e6badafd43
i: 7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC
k: 100rel,replaces
t: sip:101@192.168.1.68
m: “peter” sip:101@192.168.1.66:5060
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
l: 1362
c: application/sdp
Max-Forwards: 70

v=0
o=- 1377169122 1 IN IP4 192.168.1.66
s=Ekiga/4.0.1
c=IN IP4 192.168.1.66
t=0 0
m=audio 5094 RTP/AVP 3 93 100 0 8 99 114 9 92 102 103 94 107 106 105 104 101
a=sendrecv
a=rtpmap:3 gsm/8000/1
a=rtpmap:93 Speex/16000/1
a=rtpmap:100 CELT/48000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:99 CELT/32000/1
a=rtpmap:114 iLBC/8000/1
a=fmtp:114 mode=20
a=rtpmap:9 G722/8000/1
a=rtpmap:92 G7221/16000/1
a=fmtp:92 bitrate=24000
a=rtpmap:102 G7221/16000/1
a=fmtp:102 bitrate=32000
a=rtpmap:103 AMR-WB/16000/1
a=fmtp:103 octet-align=1
a=rtpmap:94 Speex/8000/1
a=rtpmap:107 G726-16/8000/1
a=rtpmap:106 G726-24/8000/1
a=rtpmap:105 G726-32/8000/1
a=rtpmap:104 G726-40/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=maxptime:30
m=video 5096 RTP/AVP 91 31 34 110 113 112 118
b=AS:4096
b=TIAS:4096000
a=sendrecv
a=rtpmap:91 theora/90000
a=fmtp:91 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:110 H263-1998/90000
a=fmtp:110 D=1;F=1;I=1;J=1;CIF=1;CIF4=1;QCIF=1
a=rtpmap:113 H264/90000
a=fmtp:113 max-fs=6336;max-mbps=190080;profile-level-id=42801e
a=rtpmap:112 H264/90000
a=fmtp:112 packetization-mode=1;max-fs=6336;max-mbps=190080;profile-level-id=42801e
a=rtpmap:118 MP4V-ES/90000
a=fmtp:118 profile-level-id=5
<------------->
— (14 headers 48 lines) —
Sending to 192.168.1.66:5060 (no NAT)
Using INVITE request as basis request - 7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC
Found peer ‘101’ for ‘101’ from 192.168.1.66:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 93
Found RTP audio format 100
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 99
Found RTP audio format 114
Found RTP audio format 9
Found RTP audio format 92
Found RTP audio format 102
Found RTP audio format 103
Found RTP audio format 94
Found RTP audio format 107
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 104
Found RTP audio format 101
Found audio description format gsm for ID 3
Found audio description format Speex for ID 93
Found unknown media description format CELT for ID 100
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CELT for ID 99
Found audio description format iLBC for ID 114
Found audio description format G722 for ID 9
Found audio description format G7221 for ID 92
Found audio description format G7221 for ID 102
Found unknown media description format AMR-WB for ID 103
Found audio description format Speex for ID 94
Found unknown media description format G726-16 for ID 107
Found unknown media description format G726-24 for ID 106
Found audio description format G726-32 for ID 105
Found unknown media description format G726-40 for ID 104
Found audio description format telephone-event for ID 101
Found RTP video format 91
Found RTP video format 31
Found RTP video format 34
Found RTP video format 110
Found RTP video format 113
Found RTP video format 112
Found RTP video format 118
Found video description format h261 for ID 31
Found video description format H263 for ID 34
Found video description format H263-1998 for ID 110
Found video description format H264 for ID 113
Found video description format H264 for ID 112
Found video description format MP4V-ES for ID 118
Capabilities: us - 0x4040e (gsm|ulaw|alaw|ilbc|h261), peer - audio=0x200003e0e (gsm|ulaw|alaw|g726|speex|speex16|ilbc|g722|siren7)/video=0x7c0000 (h261|h263|h263p|h264|mpeg4)/text=0x0 (nothing), combined - 0x4040e (gsm|ulaw|alaw|ilbc|h261)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.66:5094
Looking for 101 in home (domain 192.168.1.68)
list_route: hop: sip:101@192.168.1.66:5060

<— Transmitting (no NAT) to 192.168.1.66:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK871652a8-b00a-1910-9a6c-90e6badafd43;received=192.168.1.66;rport=5060
From: “peter” sip:101@192.168.1.68;tag=16ff51a8-b00a-1910-9a6a-90e6badafd43
To: sip:101@192.168.1.68
Call-ID: 7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.8-r396427
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:101@192.168.1.68:5060
Content-Length: 0

<------------>
– Executing [101@home:1] Dial(“SIP/101-00000004”, “SIP/101”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 24426
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.66:5060:
INVITE sip:101@192.168.1.66:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.68:5060;branch=z9hG4bK46a1f231
Max-Forwards: 70
From: “peter” sip:101@192.168.1.68;tag=as45b109f5
To: sip:101@192.168.1.66:5060
Contact: sip:101@192.168.1.68:5060
Call-ID: 3611aed27aa52e7e5d1d49d770735d42@192.168.1.68:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r396427
Date: Thu, 22 Aug 2013 10:58:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370

v=0
o=root 1945701816 1945701816 IN IP4 192.168.1.68
s=Asterisk PBX SVN-branch-1.8-r396427
c=IN IP4 192.168.1.68
t=0 0
m=audio 24426 RTP/AVP 0 8 97 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called SIP/101

<— SIP read from UDP:192.168.1.66:5060 —>
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.68:5060;branch=z9hG4bK46a1f231
From: “peter” sip:101@192.168.1.68;tag=as45b109f5
Call-ID: 3611aed27aa52e7e5d1d49d770735d42@192.168.1.68:5060
To: sip:101@192.168.1.66:5060
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:192.168.1.66:5060 —>
SIP/2.0 486 Busy Here
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.68:5060;branch=z9hG4bK46a1f231
User-Agent: Ekiga/4.0.1
From: “peter” sip:101@192.168.1.68;tag=as45b109f5
Call-ID: 3611aed27aa52e7e5d1d49d770735d42@192.168.1.68:5060
To: “peter” sip:101@192.168.1.66;tag=9f2b52a8-b00a-1910-9a6c-90e6badafd43
Contact: “peter” sip:101@192.168.1.66:5060
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– Got SIP response 486 “Busy Here” back from 192.168.1.66:5060
Transmitting (no NAT) to 192.168.1.66:5060:
ACK sip:101@192.168.1.66:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.68:5060;branch=z9hG4bK46a1f231
Max-Forwards: 70
From: “peter” sip:101@192.168.1.68;tag=as45b109f5
To: sip:101@192.168.1.66:5060;tag=9f2b52a8-b00a-1910-9a6c-90e6badafd43
Contact: sip:101@192.168.1.68:5060
Call-ID: 3611aed27aa52e7e5d1d49d770735d42@192.168.1.68:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.8-r396427
Content-Length: 0

-- SIP/101-00000005 is busy

== Everyone is busy/congested at this time (1:1/0/0)
– Auto fallthrough, channel ‘SIP/101-00000004’ status is ‘BUSY’

<— Reliably Transmitting (no NAT) to 192.168.1.66:5060 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK871652a8-b00a-1910-9a6c-90e6badafd43;received=192.168.1.66;rport=5060
From: “peter” sip:101@192.168.1.68;tag=16ff51a8-b00a-1910-9a6a-90e6badafd43
To: sip:101@192.168.1.68;tag=as7f9cd976
Call-ID: 7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.8-r396427
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0

<------------>

<— SIP read from UDP:192.168.1.66:5060 —>
ACK sip:101@192.168.1.68 SIP/2.0
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK871652a8-b00a-1910-9a6c-90e6badafd43;rport
Authorization: Digest username=“101”, realm=“asterisk”, nonce=“315ecac5”, uri="sip:101@192.168.1.68", algorithm=MD5, response="b437435642e807e4787cd057d685eee2"
From: “peter” sip:101@192.168.1.68;tag=16ff51a8-b00a-1910-9a6a-90e6badafd43
Call-ID: 7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC
To: sip:101@192.168.1.68;tag=as7f9cd976
Content-Length: 0
Max-Forwards: 70

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘7aff51a8-b00a-1910-9a6a-90e6badafd43@peter-PC’ Method: ACK
Really destroying SIP dialog ‘3611aed27aa52e7e5d1d49d770735d42@192.168.1.68:5060’ Method: INVITE
[/code]

sip.conf

[code][general]
context=default
allowguest=no
srvlookup=yes
udpbindaddr=0.0.0.0
tcpenable=no

[101]
type=friend
secret=xxxxxxx
qualify=no
nat=no
host=dynamic
canreinvite=no
context=home
user=101

disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
allow=h261[/code]

extensions.conf

[home] exten => 101,1,Dial(SIP/101) exten => 600,1,Answer() exten => 600,2,Playback(custom/this-call-may-be-monitored-or-recorded) exten => 600,3,Echo() exten => 600,4,Playback(custom/this-call-may-be-monitored-or-recorded) exten => 600,5,Hangup()

2

So you are calling to yourself. Your ekiga phone is telling asterisk that the phone is busy(and its true since you are calling). Enable the callwaiting feature if exist in ekiga or use another client with multiline support.

3

awesome, thank you, i got it working.