rendered paste bodymordor:/etc/asterisk # cat extensions.conf
[general]
static = no
writeprotect = yes
autofallthrough = no
[globals]
TRUNK=SIP/bv-trunk
[bvcontext]
exten => _X.,1,NoOp(Incoming Call from BroadVoice ${CALLERID} for ${EXTEN}@${SIPDOMAIN})
exten => _X.,2,Answer
exten => _X.,3,Background(digits/2)
exten => _X.,4,Dial(SIP/100)
core set verbose 5:
mordor*CLI>
<--- SIP read from 147.135.8.128:5060 --->
INVITE sip:NXXNXXXXXX@72.1.130.34:5060 SIP/2.0
Call-ID: a01fc-a@147.135.8.128
CSeq: 1 INVITE
From: "Seton Fam Hosp"<sip:NXXNXXXXXX@147.135.8.128;user=phone>;tag=ikmn
To: "Dominic Eidson"<sip:s@72.1.130.34>
Via: SIP/2.0/UDP 147.135.8.128:5060
Contact: <sip:NXXNXXXXXX@147.135.8.128:5060>
Supported: 100rel
Content-Length: 296
Content-Type: application/sdp
v=0
o=2475100407 10 10 IN IP4 147.135.8.247
s=-
c=IN IP4 147.135.8.250
t=0 0
m=audio 30718 RTP/AVP 0 18 8 2 96 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:97 t38/8000
a=rtpmap:101 telephone-event/8000
<------------->
^@--- (10 headers 13 lines) ---
^@Sending to 147.135.8.128 : 5060 (no NAT)
^@Using INVITE request as basis request - a01fc-a@147.135.8.128
^@Found peer 'bv-trunk'
^@Found RTP audio format 0
^@Found RTP audio format 18
^@Found RTP audio format 8
^@Found RTP audio format 2
^@Found RTP audio format 96
^@Found RTP audio format 97
^@Found RTP audio format 101
^@Peer audio RTP is at port 147.135.8.250:30718
^@Found description format PCMU for ID 0
^@Found description format G729 for ID 18
^@Found description format PCMA for ID 8
^@Found description format G726-32 for ID 2
^@Found description format iLBC for ID 96
^@Found description format t38 for ID 97
^@Found description format telephone-event for ID 101
^@Capabilities: us - 0x80f (g723|gsm|ulaw|alaw|g726), peer - audio=0xd0c (ulaw|a
law|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x80c (ulaw|alaw|g726)
^@Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telepho
ne-event), combined - 0x1 (telephone-event)
^@Peer audio RTP is at port 147.135.8.250:30718
^@Looking for NXXNXXXXXX in bvcontext (domain 72.1.130.34)
^@list_route: hop: <sip:NXXNXXXXXX@147.135.8.128:5060>
^@
<--- Transmitting (NAT) to 147.135.8.128:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.135.8.128:5060;received=147.135.8.128
From: "Seton Fam Hosp"<sip:NXXNXXXXXX@147.135.8.128;user=phone>;tag=ikmn
To: "Dominic Eidson"<sip:s@72.1.130.34>
Call-ID: a01fc-a@147.135.8.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:NXXNXXXXXX@72.1.130.34>
Content-Length: 0
<------------>
^@^MESC[Kmordor*CLI> ^M -- Executing [NXXNXXXXXX@bvcontext:1] NoOp("SIP/NXXNX
XXXXX-081d9918", "Incoming Call from BroadVoice for NXXNXXXXXX@72.1.130.34") in
new stack
^@ -- Executing [NXXNXXXXXX@bvcontext:2] Answer("SIP/NXXNXXXXXX-081d9918", ""
) in new stack
^@Audio is at 72.1.130.34 port 17250
^@Adding codec 0x800 (g726) to SDP
^@Adding codec 0x4 (ulaw) to SDP
^@Adding codec 0x8 (alaw) to SDP
^@Adding non-codec 0x1 (telephone-event) to SDP
^@
<--- Reliably Transmitting (NAT) to 147.135.8.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.8.128:5060;received=147.135.8.128
From: "Seton Fam Hosp"<sip:NXXNXXXXXX@147.135.8.128;user=phone>;tag=ikmn
To: "Dominic Eidson"<sip:s@72.1.130.34>;tag=as2907bf84
Call-ID: a01fc-a@147.135.8.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:NXXNXXXXXX@72.1.130.34>
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 976 976 IN IP4 72.1.130.34
s=session
c=IN IP4 72.1.130.34
t=0 0
m=audio 17250 RTP/AVP 2 0 8 101
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
^@ -- Executing [NXXNXXXXXX@bvcontext:3] BackGround("SIP/NXXNXXXXXX-081d9918"
, "digits/2") in new stack
^@ -- Playing 'digits/2' (language 'en')
^@^MESC[Kmordor*CLI>
<--- SIP read from 147.135.8.128:5060 --->
ACK sip:NXXNXXXXXX@72.1.130.34:5060 SIP/2.0
Call-ID: a01fc-a@147.135.8.128
CSeq: 1 ACK
From: "Seton Fam Hosp"<sip:NXXNXXXXXX@147.135.8.128;user=phone>;tag=ikmn
To: "Dominic Eidson"<sip:s@72.1.130.34>;tag=as2907bf84
Via: SIP/2.0/UDP 147.135.8.128:5060
Content-Length: 0
<------------->
^@--- (7 headers 0 lines) ---
^@^MESC[Kmordor*CLI>
<--- SIP read from 147.135.8.128:5060 --->
BYE sip:s@72.1.130.34:5060 SIP/2.0
Call-ID: a01fc-a@147.135.8.128
CSeq: 2 BYE
From: "Seton Fam Hosp"<sip:NXXNXXXXXX@147.135.8.128;user=phone>;tag=ikmn
To: "Dominic Eidson"<sip:s@72.1.130.34>;tag=as2907bf84
Via: SIP/2.0/UDP 147.135.8.128:5060
Content-Length: 0
<------------->
^@--- (7 headers 0 lines) ---
^@Sending to 147.135.8.128 : 5060 (NAT)
^@
<--- Transmitting (NAT) to 147.135.8.128:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.8.128:5060;received=147.135.8.128
From: "Seton Fam Hosp"<sip:NXXNXXXXXX@147.135.8.128;user=phone>;tag=ikmn
To: "Dominic Eidson"<sip:s@72.1.130.34>;tag=as2907bf84
Call-ID: a01fc-a@147.135.8.128
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:NXXNXXXXXX@72.1.130.34>
Content-Length: 0
<------------>
^@^MESC[Kmordor*CLI> ^M[Dec 27 11:29:03] WARNING[3228]: file.c:616 ast_readaudio
_callback: ^@Failed to write frame
^@ == Spawn extension (bvcontext, NXXNXXXXXX, 3) exited non-zero on 'SIP/NXXNXX
XXXX-081d9918'
^@^MESC[Kmordor*CLI> ^MReally destroying SIP dialog 'a01fc-a@147.135.8.128' Meth
od: BYE
^@^MESC[Kmordor*CLI> ^M[Dec 27 11:29:06] NOTICE[1047]: chan_sip.c:7055 sip_rereg
ister: ^@ -- Re-registration for NXXNXXXXXX@sip.broadvoice.com@sip.broadvoice
.com
^@REGISTER 13 headers, 0 lines
^@Reliably Transmitting (no NAT) to 147.135.8.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 72.1.130.34:5060;branch=z9hG4bK1abf1422;rport
From: <sip:NXXNXXXXXX@sip.broadvoice.com>;tag=as5ae8478b
To: <sip:NXXNXXXXXX@sip.broadvoice.com>
Call-ID: 61a6c8ef343a0b735cbf7dbf1331918b@192.168.1.254
CSeq: 1689 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="NXXNXXXXXX", realm="BroadWorks", algorithm=MD5,
uri="sip:sip.broadvoice.com", nonce="BroadWorksXew7fvkxhTyimm86BW", response="59
9e3893d15ef33951db5f0d7b472971", opaque="", qop=auth, cnonce="187f28c7", nc=0000
05bb
Expires: 120
Contact: <sip:s@72.1.130.34>
Event: registration
Content-Length: 0
---
^@^MESC[Kmordor*CLI>
<--- SIP read from 147.135.8.128:5060 --->
SIP/2.0 200 OK
Call-ID: 61a6c8ef343a0b735cbf7dbf1331918b@192.168.1.254
CSeq: 1689 REGISTER
From: <sip:NXXNXXXXXX@sip.broadvoice.com>;tag=as5ae8478b
To: <sip:NXXNXXXXXX@sip.broadvoice.com>
Via: SIP/2.0/UDP 72.1.130.34:5060;branch=z9hG4bK1abf1422
Contact: <sip:s@72.1.130.34>
Expires: 30
Event: registration
Content-Length: 0
<------------->
^@--- (10 headers 0 lines) ---
^@Scheduling destruction of SIP dialog '61a6c8ef343a0b735cbf7dbf1331918b@192.168
.1.254' in 32000 ms (Method: REGISTER)
^@[Dec 27 11:29:06] NOTICE[1047]: chan_sip.c:11961 handle_response_register: ^@O
utbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregi
stration in 23 s)
^@^MESC[Kmordor*CLI> ^MESC[Kmordor*CLI>