Sip log:
[code]<------------->
— (10 headers 0 lines) —
asterisk-server*CLI>
<— SIP read from UDP:172.#.#.#
INVITE sip:0YYYY-YYYY@172.#.#.#
Via: SIP/2.0/UDP 172.#.#.#
From: “3904” <sip:3904@172.#.#.#
To: <sip:0YYYY-YYYY@172.#.#.#
Remote-Party-ID: “3904” <sip:3904@172.#.#.#
Call-ID: 251aef42-c64d3a09@172.#.#.#
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username=“3904”,realm=“asterisk”,nonce=“731785ae”,uri="sip:0YYYY-YYYY@172.#.#.#
Contact: “3904” <sip:3904@172.#.#.#
Expires: 240
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 329
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 64996 64996 IN IP4 172.#.#.#
s=-
c=IN IP4 172.#.#.#
t=0 0
m=audio 16404 RTP/AVP 0 2 8 18 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
— (16 headers 16 lines) —
Sending to 172.#.#.#
Using INVITE request as basis request - 251aef42-c64d3a09@172.#.#.#
Found peer ‘3904’ for ‘3904’ from 172.#.#.#
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format PCMA for ID 8
Found audio description format G729a for ID 18
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.#.#.#
Looking for 0YYYY-YYYY in default (domain 172.#.#.#
list_route: hop: <sip:3904@172.#.#.#
asterisk-server*CLI>
<— Transmitting (no NAT) to 172.#.#.#
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.#.#.#
From: “3904” <sip:3904@172.#.#.#
To: <sip:0YYYY-YYYY@172.#.#.#
Call-ID: 251aef42-c64d3a09@172.#.#.#
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:0YYYY-YYYY@172.#.#.#
Content-Length: 0
<------------>
– Executing [0YYYY-YYYY@default:1] Dial(“SIP/3904-00000007”, “Khomp/b0L0/YYYY-YYYY,90”) in new stack
– Called b0L0/YYYY-YYYY
– Khomp/B0C0-0.0 is ringing
asterisk-server*CLI>
<— Transmitting (no NAT) to 172.#.#.#
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.#.#.#
From: “3904” <sip:3904@172.#.#.#
To: <sip:0YYYY-YYYY@172.#.#.#
Call-ID: 251aef42-c64d3a09@172.#.#.#
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:0YYYY-YYYY@172.#.#.#
Content-Length: 0
<------------>
– Khomp/B0C0-0.0 is making progress passing it to SIP/3904-00000007
Audio is at 172.#.#.#
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (no NAT) to 172.#.#.#
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.#.#.#
From: “3904” <sip:3904@172.#.#.#
To: <sip:0YYYY-YYYY@172.#.#.#
Call-ID: 251aef42-c64d3a09@172.#.#.#
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:0YYYY-YYYY@172.#.#.#
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 1608850785 1608850785 IN IP4 172.#.#.#
s=Asterisk PBX 1.6.2.6
c=IN IP4 172.#.#.#
t=0 0
m=audio 14152 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
– Khomp/B0C0-0.0 answered SIP/3904-00000007
Audio is at 172.#.#.#
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (no NAT) to 172.#.#.#
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.#.#.#
From: “3904” <sip:3904@172.#.#.#
To: <sip:0YYYY-YYYY@172.#.#.#
Call-ID: 251aef42-c64d3a09@172.#.#.#
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:0YYYY-YYYY@172.#.#.#
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 1608850785 1608850786 IN IP4 172.#.#.#
s=Asterisk PBX 1.6.2.6
c=IN IP4 172.#.#.#
t=0 0
m=audio 14152 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
asterisk-server*CLI>
<— SIP read from UDP:172.#.#.#
ACK sip:0YYYY-YYYY@172.#.#.#
Via: SIP/2.0/UDP 172.#.#.#
From: “3904” <sip:3904@172.#.#.#
To: <sip:0YYYY-YYYY@172.#.#.#
Call-ID: 251aef42-c64d3a09@172.#.#.#
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“3904”,realm=“asterisk”,nonce=“731785ae”,uri="sip:0YYYY-YYYY@172.#.#.#
Contact: “3904” <sip:3904@172.#.#.#
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 0
<------------->
— (11 headers 0 lines) —
[13-03-11 14:08:11] chan_khomp: <EV_LINK_STATUS> (d=00,l=000).
[13-03-11 14:08:31] chan_khomp: <EV_LINK_STATUS> (d=00,l=000).
Reliably Transmitting (no NAT) to 172.#.#.#
OPTIONS sip:3904@172.#.#.#
Via: SIP/2.0/UDP 172.#.#.#
Max-Forwards: 70
From: “asterisk” <sip:asterisk@172.#.#.#
To: <sip:3904@172.#.#.#
Contact: <sip:asterisk@172.#.#.#
Call-ID: 1e110083006d94152ac72453188b312e@172.#.#.#
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Mon, 11 Mar 2013 17:08:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
asterisk-server*CLI>
<— SIP read from UDP:172.#.#.#
SIP/2.0 486 Busy Here
To: <sip:3904@172.#.#.#
From: “asterisk” <sip:asterisk@172.#.#.#
Call-ID: 1e110083006d94152ac72453188b312e@172.#.#.#
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 172.#.#.#
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog '1e110083006d94152ac72453188b312e@172.#.#.#
[13-03-11 14:08:46] chan_khomp: <EV_LINK_STATUS> (d=00,l=000).
asterisk-server*CLI>
<— SIP read from UDP:172.#.#.#
BYE sip:0YYYY-YYYY@172.#.#.#
Via: SIP/2.0/UDP 172.#.#.#
From: “3904” <sip:3904@172.#.#.#
To: <sip:0YYYY-YYYY@172.#.#.#
Call-ID: 251aef42-c64d3a09@172.#.#.#
CSeq: 103 BYE
Max-Forwards: 70
Authorization: Digest username=“3904”,realm=“asterisk”,nonce=“731785ae”,uri="sip:0YYYY-YYYY@172.#.#.#
User-Agent: Cisco/SPA112-1.0.2(006)
P-RTP-Stat: PS=2219,OS=355040,PR=2221,OR=355360,PL=0,JI=0,LA=0,DU=44,EN=G711u,DE=G711u
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Sending to 172.#.#.#
asterisk-server*CLI>
<— Transmitting (no NAT) to 172.#.#.#
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.#.#.#
From: “3904” <sip:3904@172.#.#.#
To: <sip:0YYYY-YYYY@172.#.#.#
Call-ID: 251aef42-c64d3a09@172.#.#.#
CSeq: 103 BYE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (default, 0YYYY-YYYY, 1) exited non-zero on 'SIP/3904-00000007’
Really destroying SIP dialog '251aef42-c64d3a09@172.#.#.#
[13-03-11 14:09:06] chan_khomp: <EV_LINK_STATUS> (d=00,l=000).
Reliably Transmitting (no NAT) to 172.#.#.#
OPTIONS sip:3904@172.#.#.#
Via: SIP/2.0/UDP 172.#.#.#
Max-Forwards: 70
From: “asterisk” <sip:asterisk@172.#.#.#
To: <sip:3904@172.#.#.#
Contact: <sip:asterisk@172.#.#.#
Call-ID: 38e8d42936827cd10f0c62b72ee9c8d6@172.#.#.#
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Mon, 11 Mar 2013 17:09:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
asterisk-server*CLI>
<— SIP read from UDP:172.#.#.#
SIP/2.0 200 OK
To: <sip:3904@172.#.#.#
From: “asterisk” <sip:asterisk@172.#.#.#
Call-ID: 38e8d42936827cd10f0c62b72ee9c8d6@172.#.#.#
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 172.#.#.#
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog '38e8d42936827cd10f0c62b72ee9c8d6@172.#.#.#
[13-03-11 14:09:46] chan_khomp: <EV_LINK_STATUS> (d=00,l=000).
[/code]
*0YYYYYYYY is the number I’m calling
**172.#.#.# are IPs from our network