[asterisk-users] Mediatrix 1204
Robbie Hughes
spam at dynsysgroup.com
Wed Apr 11 10:42:57 MST 2007
Hi -
I've recently bought a mediatrix 1204 and have had a complete nightmare
getting it up and running with an asterisk at home setup. I know this isn't a
mediatrix list but I'm at my wits end and the support with this product is
atrocious. (mine was even shipped with firmware that was incompatible with
the win32 software it came with so I wasted a day trying to work out why the
SNMP software wouldn't work )
I've finally managed to get incoming calls to work properly by getting it to
forward all calls to 4000 which is then passed on to the asterisk proxy and
treated as an inbound route that gets answered correctly.
The problem is then that when I place an outbound call through the gateway
it also forwards that back as well. It then uses each channel in order until
it fails as they're all busy.
The xml configuration file is at http://www.ascensus.co.uk/config.xml
The asterisk debug log is as below with my mobile replaced with mymobileno:
I've also attached sip.conf below. If anyone has any idea how to get this
thing to accept outgoing calls I would be very grateful of any input. All
the docs and howto's I've found state that it should 'just work' once the
inbound settings are working but I've not found that to be the case. The
settings are all defaults except the following:
Static IP address
Proxy server address
VAD on 711 disabled
Comfort noise disabled
AutomaticCallEnable yes
AutomaticCallTargetAddress 4000 (which is obviously the problem...)
Any help appreciated
Thanks
robbie
Sip.conf
<snip>
[inbound]
type=friend
host=192.168.0.253
context=from-pstn
canreinvite=no
allow=ulaw
allow=alaw
<snip>
asterisk1*CLI>
-- Executing Macro("SIP/4005-9d61", "dialout-trunk|7|mymobilenumber|")
in new stack
-- Executing GotoIf("SIP/4005-9d61", "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/4005-9d61", "record-enable|4005|OUT") in new
stack
-- Executing GotoIf("SIP/4005-9d61", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing GotoIf("SIP/4005-9d61", "1?5:8") in new stack
-- Goto (macro-record-enable,s,5)
-- Executing DBget("SIP/4005-9d61", "RecEnable=RECORD-OUT/4005") in new
stack
-- DBget: varname=RecEnable, family=RECORD-OUT, key=4005
-- DBget: Value not found in database.
-- Executing SetVar("SIP/4005-9d61",
"CALLFILENAME=OUT4005-20070411-181258-1176311578.13302") in new stack
-- Executing Goto("SIP/4005-9d61", "s|14") in new stack
-- Goto (macro-record-enable,s,14)
-- Executing GotoIf("SIP/4005-9d61", "0?15:99") in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("SIP/4005-9d61", "NO RECORDING NEEDED") in new stack
-- Executing GotoIf("SIP/4005-9d61", "fooBgate:?7") in new stack
-- Executing SetCallerID("SIP/4005-9d61", "Bgate: Treatment (Large)
<4005>") in new stack
-- Executing Goto("SIP/4005-9d61", "9") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing SetGroup("SIP/4005-9d61", "OUT_7") in new stack
-- Executing CheckGroup("SIP/4005-9d61", "") in new stack
-- Executing SetVar("SIP/4005-9d61", "DIAL_NUMBER=mymobilenumber") in
new stack
-- Executing SetVar("SIP/4005-9d61", "DIAL_TRUNK=7") in new stack
-- Executing AGI("SIP/4005-9d61", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("SIP/4005-9d61", "OUTNUM=mymobilenumber") in new
stack
-- Executing Cut("SIP/4005-9d61", "custom=OUT_7|:|1") in new stack
-- Executing GotoIf("SIP/4005-9d61", "0?19") in new stack
-- Executing Dial("SIP/4005-9d61", "SIP/inbound/mymobilenumber") in new
stack
We're at 192.168.0.254 port 12542
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:mymobilenumber at 192.168.0.253 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435
From: "Bgate: Treatment (Large)" <sip:4005 at 192.168.0.254>;tag=as5b17ec6a
To: <sip:mymobilenumber at 192.168.0.253>
Contact: <sip:4005 at 192.168.0.254>
Call-ID: 27db49cf7a2f459c28dd631e4ae428be at 192.168.0.254
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 11 Apr 2007 17:12:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 1321 1321 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 12542 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 192.168.0.253:5060
-- Called inbound/mymobilenumber
asterisk1*CLI>
Sip read:
SIP/2.0 100 Trying
Call-ID: 27db49cf7a2f459c28dd631e4ae428be at 192.168.0.254
CSeq: 102 INVITE
From: "Bgate: Treatment (Large)" <sip:4005 at 192.168.0.254>;tag=as5b17ec6a
To: <sip:mymobilenumber at 192.168.0.253>;tag=2120bdca0a07567
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435
Content-Length: 0
User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1
8 headers, 0 lines
asterisk1*CLI>
Sip read:
INVITE sip:4000 at 192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK2a550f4da
Content-Length: 296
To: sip:4000 at 192.168.0.254
From: Incoming <sip:3330001 at 192.168.0.254>;tag=68b2ce27259fa46
Call-ID: 9dd4c369dbcf118b22b91aa08728f726 at 192.168.0.254
CSeq: 527193825 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: Incoming <sip:3330001 at 192.168.0.253>
Supported: replaces
User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1
v=0
o=MxSIP 1616130847127821823 1523581411134402127 IN IP4 192.168.0.253
s=-
c=IN IP4 192.168.0.253
t=0 0
a=sendrecv
m=audio 5004 RTP/AVP 0 18 4 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
15 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.253 : 5060 (non-NAT)
Found peer 'inbound'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Peer audio RTP is at port 192.168.0.253:5004
Found description format PCMU
Found description format G729
Found description format G723
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
Looking for 4000 in from-pstn
list_route: hop: <sip:3330001 at 192.168.0.253>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK2a550f4da
From: Incoming <sip:3330001 at 192.168.0.254>;tag=68b2ce27259fa46
To: sip:4000 at 192.168.0.254
Call-ID: 9dd4c369dbcf118b22b91aa08728f726 at 192.168.0.254
CSeq: 527193825 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4000 at 192.168.0.254>
Content-Length: 0
to 192.168.0.253:5060
-- Executing SetVar("SIP/192.168.0.254-b2159358", "FROM_DID=4000") in
new stack
-- Executing Goto("SIP/192.168.0.254-b2159358", "aa_2|s|1") in new stack
-- Goto (aa_2,s,1)
-- Executing GotoIf("SIP/192.168.0.254-b2159358", "0?4") in new stack
-- Executing Answer("SIP/192.168.0.254-b2159358", "") in new stack
We're at 192.168.0.254 port 18268
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK2a550f4da
From: Incoming <sip:3330001 at 192.168.0.254>;tag=68b2ce27259fa46
To: sip:4000 at 192.168.0.254;tag=as3f53dd1e
Call-ID: 9dd4c369dbcf118b22b91aa08728f726 at 192.168.0.254
CSeq: 527193825 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4000 at 192.168.0.254>
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 1321 1321 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 18268 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
to 192.168.0.253:5060
-- Executing Wait("SIP/192.168.0.254-b2159358", "1") in new stack
asterisk1*CLI>
Sip read:
ACK sip:4000 at 192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK865a4fbe1
Content-Length: 0
To: sip:4000 at 192.168.0.254;tag=as3f53dd1e
From: Incoming <sip:3330001 at 192.168.0.254>;tag=68b2ce27259fa46
Call-ID: 9dd4c369dbcf118b22b91aa08728f726 at 192.168.0.254
CSeq: 527193825 ACK
Contact: Incoming <sip:3330001 at 192.168.0.253>
User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1
9 headers, 0 lines
-- Executing SetVar("SIP/192.168.0.254-b2159358", "DIR-CONTEXT=general")
in new stack
-- Executing DigitTimeout("SIP/192.168.0.254-b2159358", "3") in new
stack
-- Set Digit Timeout to 3
-- Executing ResponseTimeout("SIP/192.168.0.254-b2159358", "7") in new
stack
-- Set Response Timeout to 7
-- Executing BackGround("SIP/192.168.0.254-b2159358", "custom/aa_2") in
new stack
-- Playing 'custom/aa_2' (language 'en')
asterisk1*CLI>
Sip read:
INVITE sip:4000 at 192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK672206098
Content-Length: 294
To: sip:4000 at 192.168.0.254
From: Incoming <sip:3330002 at 192.168.0.254>;tag=c4f38b97a1b7d91
Call-ID: 9c3107ca1b254febdba49a505d8d8bba at 192.168.0.254
CSeq: 1922136933 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: Incoming <sip:3330002 at 192.168.0.253>
Supported: replaces
User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1
v=0
o=MxSIP 666448690115909283 852756300509302897 IN IP4 192.168.0.253
s=-
c=IN IP4 192.168.0.253
t=0 0
a=sendrecv
m=audio 5006 RTP/AVP 0 18 4 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
15 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.253 : 5060 (non-NAT)
Found peer 'inbound'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Peer audio RTP is at port 192.168.0.253:5006
Found description format PCMU
Found description format G729
Found description format G723
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
Looking for 4000 in from-pstn
list_route: hop: <sip:3330002 at 192.168.0.253>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK672206098
From: Incoming <sip:3330002 at 192.168.0.254>;tag=c4f38b97a1b7d91
To: sip:4000 at 192.168.0.254
Call-ID: 9c3107ca1b254febdba49a505d8d8bba at 192.168.0.254
CSeq: 1922136933 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4000 at 192.168.0.254>
Content-Length: 0
to 192.168.0.253:5060
-- Executing SetVar("SIP/192.168.0.254-b21601d0", "FROM_DID=4000") in
new stack
-- Executing Goto("SIP/192.168.0.254-b21601d0", "aa_2|s|1") in new stack
-- Goto (aa_2,s,1)
-- Executing GotoIf("SIP/192.168.0.254-b21601d0", "0?4") in new stack
-- Executing Answer("SIP/192.168.0.254-b21601d0", "") in new stack
We're at 192.168.0.254 port 13034
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK672206098
From: Incoming <sip:3330002 at 192.168.0.254>;tag=c4f38b97a1b7d91
To: sip:4000 at 192.168.0.254;tag=as662b9545
Call-ID: 9c3107ca1b254febdba49a505d8d8bba at 192.168.0.254
CSeq: 1922136933 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4000 at 192.168.0.254>
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 1321 1321 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 13034 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
to 192.168.0.253:5060
-- Executing Wait("SIP/192.168.0.254-b21601d0", "1") in new stack
asterisk1*CLI>
Sip read:
ACK sip:4000 at 192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bKcc0a3292d
Content-Length: 0
To: sip:4000 at 192.168.0.254;tag=as662b9545
From: Incoming <sip:3330002 at 192.168.0.254>;tag=c4f38b97a1b7d91
Call-ID: 9c3107ca1b254febdba49a505d8d8bba at 192.168.0.254
CSeq: 1922136933 ACK
Contact: Incoming <sip:3330002 at 192.168.0.253>
User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1
9 headers, 0 lines
-- Executing SetVar("SIP/192.168.0.254-b21601d0", "DIR-CONTEXT=general")
in new stack
-- Executing DigitTimeout("SIP/192.168.0.254-b21601d0", "3") in new
stack
-- Set Digit Timeout to 3
-- Executing ResponseTimeout("SIP/192.168.0.254-b21601d0", "7") in new
stack
-- Set Response Timeout to 7
-- Executing BackGround("SIP/192.168.0.254-b21601d0", "custom/aa_2") in
new stack
-- Playing 'custom/aa_2' (language 'en')
asterisk1*CLI>
Sip read:
SIP/2.0 480 Temporarily Unavailable
Call-ID: 27db49cf7a2f459c28dd631e4ae428be at 192.168.0.254
CSeq: 102 INVITE
From: "Bgate: Treatment (Large)" <sip:4005 at 192.168.0.254>;tag=as5b17ec6a
To: <sip:mymobilenumber at 192.168.0.253>;tag=2120bdca0a07567
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435
Content-Length: 0
User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1
8 headers, 0 lines
-- Got SIP response 480 "Temporarily Unavailable" back from
192.168.0.253
Transmitting:
ACK sip:mymobilenumber at 192.168.0.253 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435
From: "Bgate: Treatment (Large)" <sip:4005 at 192.168.0.254>;tag=as5b17ec6a
To: <sip:mymobilenumber at 192.168.0.253>;tag=2120bdca0a07567
Contact: <sip:4005 at 192.168.0.254>
Call-ID: 27db49cf7a2f459c28dd631e4ae428be at 192.168.0.254
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.0.253:5060
-- SIP/inbound-5c55 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Goto("SIP/4005-9d61", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/4005-9d61", "Dial failed due to CONGESTION") in
new stack
-- Executing Macro("SIP/4005-9d61", "outisbusy") in new stack
-- Executing Answer("SIP/4005-9d61", "") in new stack
-- Executing Playtones("SIP/4005-9d61", "congestion") in new stack
-- Executing Congestion("SIP/4005-9d61", "") in new stack
Destroying call '27db49cf7a2f459c28dd631e4ae428be at 192.168.0.254'
asterisk1*CLI>
Sip read:
INVITE sip:4000 at 192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK44078e2aa
Content-Length: 295
To: sip:4000 at 192.168.0.254
From: Incoming <sip:3330003 at 192.168.0.254>;tag=13c1d178078adcb
Call-ID: a977c0f985da61435380e0d413d33c96 at 192.168.0.254
CSeq: 375158022 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: Incoming <sip:3330003 at 192.168.0.253>
Supported: replaces
User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1
v=0
o=MxSIP 1766393978165944161 643256180583431736 IN IP4 192.168.0.253
s=-
c=IN IP4 192.168.0.253
t=0 0
a=sendrecv
m=audio 5008 RTP/AVP 0 18 4 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
15 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.253 : 5060 (non-NAT)
Found peer 'inbound'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Peer audio RTP is at port 192.168.0.253:5008
Found description format PCMU
Found description format G729
Found description format G723
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
Looking for 4000 in from-pstn
list_route: hop: <sip:3330003 at 192.168.0.253>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK44078e2aa
From: Incoming <sip:3330003 at 192.168.0.254>;tag=13c1d178078adcb
To: sip:4000 at 192.168.0.254
Call-ID: a977c0f985da61435380e0d413d33c96 at 192.168.0.254
CSeq: 375158022 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4000 at 192.168.0.254>
Content-Length: 0
to 192.168.0.253:5060
-- Executing SetVar("SIP/192.168.0.254-b2167540", "FROM_DID=4000") in
new stack
-- Executing Goto("SIP/192.168.0.254-b2167540", "aa_2|s|1") in new stack
-- Goto (aa_2,s,1)
-- Executing GotoIf("SIP/192.168.0.254-b2167540", "0?4") in new stack
-- Executing Answer("SIP/192.168.0.254-b2167540", "") in new stack
We're at 192.168.0.254 port 11990
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK44078e2aa
From: Incoming <sip:3330003 at 192.168.0.254>;tag=13c1d178078adcb
To: sip:4000 at 192.168.0.254;tag=as4f5cc934
Call-ID: a977c0f985da61435380e0d413d33c96 at 192.168.0.254
CSeq: 375158022 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4000 at 192.168.0.254>
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 1321 1321 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 11990 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
to 192.168.0.253:5060
-- Executing Wait("SIP/192.168.0.254-b2167540", "1") in new stack
asterisk1*CLI>
Sip read:
ACK sip:4000 at 192.168.0.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bKd6fd96da5
Content-Length: 0
To: sip:4000 at 192.168.0.254;tag=as4f5cc934
From: Incoming <sip:3330003 at 192.168.0.254>;tag=13c1d178078adcb
Call-ID: a977c0f985da61435380e0d413d33c96 at 192.168.0.254
CSeq: 375158022 ACK
Contact: Incoming <sip:3330003 at 192.168.0.253>
User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1
9 headers, 0 lines
-- Executing SetVar("SIP/192.168.0.254-b2167540", "DIR-CONTEXT=general")
in new stack
-- Executing DigitTimeout("SIP/192.168.0.254-b2167540", "3") in new
stack
-- Set Digit Timeout to 3
-- Executing ResponseTimeout("SIP/192.168.0.254-b2167540", "7") in new
stack
-- Set Response Timeout to 7
-- Executing BackGround("SIP/192.168.0.254-b2167540", "custom/aa_2") in
new stack
-- Playing 'custom/aa_2' (language 'en')
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on
'SIP/4005-9d61' in macro 'outisbusy'
== Spawn extension (from-internal, 88mymobilenumber, 2) exited non-zero on
'SIP/4005-9d61'
-- Executing Macro("SIP/4005-9d61", "hangupcall") in new stack
-- Executing ResetCDR("SIP/4005-9d61", "w") in new stack
-- Executing NoCDR("SIP/4005-9d61", "") in new stack
-- Executing Wait("SIP/4005-9d61", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/4005-9d61' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/4005-9d61'
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