Not Found Error

mshoaib · March 23, 2015, 5:30am

Hi.
I’m running Asterisk SVN-branch-1.8-r331578M. I’m getting not found error mostly i sending calls to asterisk. my configration is as below

extensions.usr

[code]context 691 {
_67X. => {
Dial(SIP/${EXTEN:2}@x.x.x…4:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…4:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…4:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…4:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…4:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…4:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…4:5060,120);
Hangup();
}
}

context N691-1 {
_68X. => {
Dial(SIP/${EXTEN:2}@x.x.x…2:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…2:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…2:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…2:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…2:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…2:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…2:5060,120);
Hangup();
}
}

context N691-2 {
_69X. => {
Dial(SIP/${EXTEN:2}@x.x.x…3:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…3:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…3:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…3:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…3:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…3:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…3:5060,120);
Dial(SIP/${EXTEN:2}@x.x.x…3:5060,120);
Hangup();
}
}[/code]

extensions.ael

[code]context box1 {
_67X. => {
Dial(IAX/691/${EXTEN},120);
Dial(IAX/691/${EXTEN},120);
Dial(IAX/691/${EXTEN},120);
Dial(IAX/691/${EXTEN},120);
Dial(IAX/691/${EXTEN},120);
Dial(IAX/691/${EXTEN},120);
Dial(IAX/691/${EXTEN},120);
Hangup();
}

_68X.	=> {
	Dial(IAX/N691-1/${EXTEN},120);
	Dial(IAX/N691-1/${EXTEN},120);
	Dial(IAX/N691-1/${EXTEN},120);
	Dial(IAX/N691-1/${EXTEN},120);
	Dial(IAX/N691-1/${EXTEN},120);
	Dial(IAX/N691-1/${EXTEN},120);
	Dial(IAX/N691-1/${EXTEN},120);
	Hangup();
}
				
_69X.	=> {
	Dial(IAX/N691-2/${EXTEN},120);
	Dial(IAX/N691-2/${EXTEN},120);
	Dial(IAX/N691-2/${EXTEN},120);
	Dial(IAX/N691-2/${EXTEN},120);
	Dial(IAX/N691-2/${EXTEN},120);
	Dial(IAX/N691-2/${EXTEN},120);
	Dial(IAX/N691-2/${EXTEN},120);
	Hangup();
}

}[/code]

sip.conf

[code][general]
context=default
useragent=Box1
bindport=5060
directmedia=no
disallow=all ; First disallow all codecs
allow=g729
allow=g723

[95.85.38.168]
type=peer
host=95.85.38.168
context=box1
call-limit=46

[/code]

david55 · March 23, 2015, 9:25am

All Asterisk 1.8 versions are past the end of their support life.

Few people use AEL.

You haven’t provided any logs. The minimum is verbose level 3, but this may well require sip debugging enabled.

mshoaib · March 23, 2015, 6:26pm

SIP Debug

[code]<------------>
– Executing [6803047101236@176960231:1] Dial(“SIP/95.85.38.168-000014a4”, “IAX/N691-1/6803047101236,120”) in new stack
– Called IAX/N691-1/6803047101236
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]
> Ringing channel IAX/N691-1-20
– IAX/N691-1-20 is ringing

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK445711722b066fc5;received=95.85.38.168;rport=5060
From: sip:966595214786@95.85.38.168;tag=6ec1cd8500391ad9
To: sip:6803047101236@188.166.42.31;tag=as05fc6cdb
Call-ID: 64af6ed116c517fc1f2aa6bc00021ae4@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803047101236@188.166.42.31:5060
Content-Length: 0

<------------>
– IAX/N691-1-20 is making progress passing it to SIP/95.85.38.168-000014a4
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK445711722b066fc5;received=95.85.38.168;rport=5060
From: sip:966595214786@95.85.38.168;tag=6ec1cd8500391ad9
To: sip:6803047101236@188.166.42.31;tag=as05fc6cdb
Call-ID: 64af6ed116c517fc1f2aa6bc00021ae4@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803047101236@188.166.42.31:5060
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 185385731 185385731 IN IP4 188.166.42.31
s=Asterisk PBX SVN-branch-1.8-r331578M
c=IN IP4 188.166.42.31
t=0 0
m=audio 10794 RTP/AVP 18 4
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:95.85.38.168:5060 —>
INVITE sip:6803037954401@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK5fb5b5e234524792
From: sip:neu183@95.85.38.168;tag=62824d5d40648c52
To: sip:6803037954401@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 INVITE
Call-ID: 76127fdd2fb708ff1f2aa6bc00021ae5@95.85.38.168
Contact: sip:neu183@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Supported: timer
Session-Expires: 600;refresher=uas
Content-Length: 211
Content-Type: application/sdp

v=0
o=- 1427150274 1427150274 IN IP4 108.60.207.68
s=VOS2009
c=IN IP4 108.60.207.68
t=0 0
m=audio 9500 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 10 lines) —
Sending to 95.85.38.168:5060 (NAT)
Using INVITE request as basis request - 76127fdd2fb708ff1f2aa6bc00021ae5@95.85.38.168
Found peer ‘95.85.38.168’ for ‘neu183’ from 95.85.38.168:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x101 (g723|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 108.60.207.68:9500
Looking for 6803037954401 in 176960231 (domain 188.166.42.31)
list_route: hop: sip:neu183@95.85.38.168:5060

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK5fb5b5e234524792;received=95.85.38.168;rport=5060
From: sip:neu183@95.85.38.168;tag=62824d5d40648c52
o: sip:6803037954401@188.166.42.31
Call-ID: 76127fdd2fb708ff1f2aa6bc00021ae5@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803037954401@188.166.42.31:5060
Content-Length: 0

<------------>
– Executing [6803037954401@176960231:1] Dial(“SIP/95.85.38.168-000014a5”, “IAX/N691-1/6803037954401,120”) in new stack
– Called IAX/N691-1/6803037954401
> Ringing channel IAX/N691-1-16
– IAX/N691-1-16 is ringing
– IAX/N691-1-16 is making progress passing it to SIP/95.85.38.168-0000149e
> Ringing channel IAX/N691-1-21
– IAX/N691-1-21 is ringing

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK5fb5b5e234524792;received=95.85.38.168;rport=5060
From: sip:neu183@95.85.38.168;tag=62824d5d40648c52
To: sip:6803037954401@188.166.42.31;tag=as5124e178
Call-ID: 76127fdd2fb708ff1f2aa6bc00021ae5@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803037954401@188.166.42.31:5060
Content-Length: 0

<------------>
– IAX/N691-1-21 is making progress passing it to SIP/95.85.38.168-000014a5
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK5fb5b5e234524792;received=95.85.38.168;rport=5060
From: sip:neu183@95.85.38.168;tag=62824d5d40648c52
To: sip:6803037954401@188.166.42.31;tag=as5124e178
Call-ID: 76127fdd2fb708ff1f2aa6bc00021ae5@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803037954401@188.166.42.31:5060
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 1087366358 1087366358 IN IP4 188.166.42.31
s=Asterisk PBX SVN-branch-1.8-r331578M
c=IN IP4 188.166.42.31
t=0 0
m=audio 15412 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
> Ringing channel IAX/N691-1-9
– IAX/N691-1-9 is ringing
– IAX/N691-1-9 is making progress passing it to SIP/95.85.38.168-00001496
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]
> Answering channel IAX/N691-1-12
– IAX/N691-1-12 answered SIP/95.85.38.168-0000149f
Audio is at 5060
– peer joint caps (0x100 (g729))
Adding codec 0x100 (g729) to SDP

<— Reliably Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK71a856875215adda;received=95.85.38.168;rport=5060
From: sip:91039921670@95.85.38.168;tag=1bc9547c61156d81
To: sip:6803039490926@188.166.42.31;tag=as0d530243
Call-ID: 299baecb1b81bc5f1f2aa6bc00021adf@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803039490926@188.166.42.31:5060
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 18215231 18215232 IN IP4 188.166.42.31
s=Asterisk PBX SVN-branch-1.8-r331578M
c=IN IP4 188.166.42.31
t=0 0
m=audio 16718 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:95.85.38.168:5060 —>
ACK sip:6803039490926@188.166.42.31:5060 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK71a856875215adda
From: sip:91039921670@95.85.38.168;tag=1bc9547c61156d81
To: sip:6803039490926@188.166.42.31;tag=as0d530243
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 ACK
Call-ID: 299baecb1b81bc5f1f2aa6bc00021adf@95.85.38.168
Contact: sip:91039921670@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Content-Length: 0

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

<— SIP read from UDP:95.85.38.168:5060 —>
INVITE sip:6803068203232@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK08d33bc84488996e
From: sip:906300@95.85.38.168;tag=0357866c306e46f0
To: sip:6803068203232@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 INVITE
Call-ID: 09557054739fedf91f2aa6bc00021ae6@95.85.38.168
Contact: sip:906300@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Supported: timer
Session-Expires: 600;refresher=uas
Content-Length: 212
Content-Type: application/sdp

v=0
o=- 1427150277 1427150277 IN IP4 85.195.86.170
s=VOS2009
c=IN IP4 85.195.86.170
t=0 0
m=audio 13184 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 10 lines) —
Sending to 95.85.38.168:5060 (NAT)
Using INVITE request as basis request - 09557054739fedf91f2aa6bc00021ae6@95.85.38.168
Found peer ‘95.85.38.168’ for ‘906300’ from 95.85.38.168:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x101 (g723|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 85.195.86.170:13184
Looking for 6803068203232 in 176960231 (domain 188.166.42.31)
list_route: hop: sip:906300@95.85.38.168:5060

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK08d33bc84488996e;received=95.85.38.168;rport=5060
From: sip:906300@95.85.38.168;tag=0357866c306e46f0
To: sip:6803068203232@188.166.42.31
Call-ID: 09557054739fedf91f2aa6bc00021ae6@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803068203232@188.166.42.31:5060
Content-Length: 0

<------------>
– Executing [6803068203232@176960231:1] Dial(“SIP/95.85.38.168-000014a6”, “IAX/N691-1/6803068203232,120”) in new stack
– Called IAX/N691-1/6803068203232
> Ringing channel IAX/N691-1-5
– IAX/N691-1-5 is ringing
– IAX/N691-1-5 is making progress passing it to SIP/95.85.38.168-000014a2
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]
> Ringing channel IAX/N691-1-7
– IAX/N691-1-7 is ringing

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK08d33bc84488996e;received=95.85.38.168;rport=5060
From: sip:906300@95.85.38.168;tag=0357866c306e46f0
To: sip:6803068203232@188.166.42.31;tag=as5b888f47
Call-ID: 09557054739fedf91f2aa6bc00021ae6@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803068203232@188.166.42.31:5060
Content-Length: 0

<------------>
– IAX/N691-1-7 is making progress passing it to SIP/95.85.38.168-000014a6
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK08d33bc84488996e;received=95.85.38.168;rport=5060
From: sip:906300@95.85.38.168;tag=0357866c306e46f0
To: sip:6803068203232@188.166.42.31;tag=as5b888f47
Call-ID: 09557054739fedf91f2aa6bc00021ae6@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803068203232@188.166.42.31:5060
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 1103481229 1103481229 IN IP4 188.166.42.31
s=Asterisk PBX SVN-branch-1.8-r331578M
c=IN IP4 188.166.42.31
t=0 0
m=audio 13292 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:95.85.38.168:5060 —>
CANCEL sip:6803075768615@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK4d4f99834bf54de3
From: sip:966595288406@95.85.38.168;tag=6be7cba2559543ed
To: sip:6803075768615@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 CANCEL
Call-ID: 38f167bb1b89ac081f2aa6bc00021ae1@95.85.38.168
Contact: sip:966595288406@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 95.85.38.168:5060 (NAT)

<— Reliably Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK4d4f99834bf54de3;received=95.85.38.168;rport=5060
From: sip:966595288406@95.85.38.168;tag=6be7cba2559543ed
To: sip:6803075768615@188.166.42.31;tag=as0be4bc48
Call-ID: 38f167bb1b89ac081f2aa6bc00021ae1@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK4d4f99834bf54de3;received=95.85.38.168;rport=5060
From: sip:966595288406@95.85.38.168;tag=6be7cba2559543ed
To: sip:6803075768615@188.166.42.31;tag=as0be4bc48
Call-ID: 38f167bb1b89ac081f2aa6bc00021ae1@95.85.38.168
CSeq: 1 CANCEL
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (176960231, 6803075768615, 1) exited non-zero on ‘SIP/95.85.38.168-000014a1’

<— SIP read from UDP:95.85.38.168:5060 —>
ACK sip:6803075768615@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK4d4f99834bf54de3
From: sip:966595288406@95.85.38.168;tag=6be7cba2559543ed
To: sip:6803075768615@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 ACK
Call-ID: 38f167bb1b89ac081f2aa6bc00021ae1@95.85.38.168
Contact: sip:966595288406@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Retransmitting #1 (NAT) to 95.85.38.168:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK4d4f99834bf54de3;received=95.85.38.168;rport=5060
From: sip:966595288406@95.85.38.168;tag=6be7cba2559543ed
To: sip:6803075768615@188.166.42.31;tag=as0be4bc48
Call-ID: 38f167bb1b89ac081f2aa6bc00021ae1@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:95.85.38.168:5060 —>
ACK sip:95.85.38.168 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK4d4f99834bf54de3;received=95.85.38.168;rport=5060
From: sip:966595288406@95.85.38.168;tag=6be7cba2559543ed
To: sip:6803075768615@188.166.42.31;tag=as0be4bc48
Call-ID: 38f167bb1b89ac081f2aa6bc00021ae1@95.85.38.168
CSeq: 1 ACK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
User-Agent: VOS2009 V2.1.2.0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘38f167bb1b89ac081f2aa6bc00021ae1@95.85.38.168’ Method: ACK
> Ringing channel IAX/N691-1-6
– IAX/N691-1-6 is ringing
– IAX/N691-1-6 is making progress passing it to SIP/95.85.38.168-0000149b
Really destroying SIP dialog ‘2a55b26e3b549d291f2aa6bc00021adc@95.85.38.168’ Method: BYE
> Ringing channel IAX/N691-1-19
– IAX/N691-1-19 is ringing
– IAX/N691-1-19 is making progress passing it to SIP/95.85.38.168-000014a3
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]
> Ringing channel IAX/N691-1-16
– IAX/N691-1-16 is ringing
– IAX/N691-1-16 is making progress passing it to SIP/95.85.38.168-0000149e
> Ringing channel IAX/N691-1-21
– IAX/N691-1-21 is ringing
– IAX/N691-1-21 is making progress passing it to SIP/95.85.38.168-000014a5
> Ringing channel IAX/N691-1-20
– IAX/N691-1-20 is ringing
– IAX/N691-1-20 is making progress passing it to SIP/95.85.38.168-000014a4
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]
> Answering channel IAX/N691-1-5
– IAX/N691-1-5 answered SIP/95.85.38.168-000014a2
Audio is at 5060
– peer joint caps (0x1 (g723))
Adding codec 0x1 (g723) to SDP

<— Reliably Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK5b69677218fcbd01;received=95.85.38.168;rport=5060
From: sip:966580167711@95.85.38.168;tag=3fe7ab137d336885
To: sip:6803056008337@188.166.42.31;tag=as65d3474a
Call-ID: 5235ee084a520c751f2aa6bc00021ae2@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803056008337@188.166.42.31:5060
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 1451114140 1451114141 IN IP4 188.166.42.31
s=Asterisk PBX SVN-branch-1.8-r331578M
c=IN IP4 188.166.42.31
t=0 0
m=audio 11164 RTP/AVP 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=ptime:30
a=sendrecv

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

<— SIP read from UDP:95.85.38.168:5060 —>
ACK sip:6803056008337@188.166.42.31:5060 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK5b69677218fcbd01
From: sip:966580167711@95.85.38.168;tag=3fe7ab137d336885
To: sip:6803056008337@188.166.42.31;tag=as65d3474a
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 ACK
Call-ID: 5235ee084a520c751f2aa6bc00021ae2@95.85.38.168
Contact: sip:966580167711@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Content-Length: 0

<------------->
— (11 headers 0 lines) —
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]

<— SIP read from UDP:95.85.38.168:5060 —>
CANCEL sip:6803017490388@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK0fe0962e5f074130
From: sip:966599824123@95.85.38.168;tag=59bc2dc510fb1249
To: sip:6803017490388@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 CANCEL
Call-ID: 0de3443637f8c7211f2aa6bc00021adb@95.85.38.168
Contact: sip:966599824123@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 95.85.38.168:5060 (NAT)

<— Reliably Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK0fe0962e5f074130;received=95.85.38.168;rport=5060
From: sip:966599824123@95.85.38.168;tag=59bc2dc510fb1249
To: sip:6803017490388@188.166.42.31;tag=as1427f0a4
Call-ID: 0de3443637f8c7211f2aa6bc00021adb@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
upported: replaces, timer
Content-Length: 0

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

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK0fe0962e5f074130;received=95.85.38.168;rport=5060
From: sip:966599824123@95.85.38.168;tag=59bc2dc510fb1249
To: sip:6803017490388@188.166.42.31;tag=as1427f0a4
Call-ID: 0de3443637f8c7211f2aa6bc00021adb@95.85.38.168
CSeq: 1 CANCEL
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (176960231, 6803017490388, 1) exited non-zero on ‘SIP/95.85.38.168-0000149b’

<— SIP read from UDP:95.85.38.168:5060 —>
ACK sip:6803017490388@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK0fe0962e5f074130
From: sip:966599824123@95.85.38.168;tag=59bc2dc510fb1249
To: sip:6803017490388@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 ACK
Call-ID: 0de3443637f8c7211f2aa6bc00021adb@95.85.38.168
Contact: sip:966599824123@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Retransmitting #1 (NAT) to 95.85.38.168:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK0fe0962e5f074130;received=95.85.38.168;rport=5060
From: sip:966599824123@95.85.38.168;tag=59bc2dc510fb1249
To: sip:6803017490388@188.166.42.31;tag=as1427f0a4
Call-ID: 0de3443637f8c7211f2aa6bc00021adb@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:95.85.38.168:5060 —>
ACK sip:95.85.38.168 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK0fe0962e5f074130;received=95.85.38.168;rport=5060
From: sip:966599824123@95.85.38.168;tag=59bc2dc510fb1249
To: sip:6803017490388@188.166.42.31;tag=as1427f0a4
Call-ID: 0de3443637f8c7211f2aa6bc00021adb@95.85.38.168
CSeq: 1 ACK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
User-Agent: VOS2009 V2.1.2.0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘0de3443637f8c7211f2aa6bc00021adb@95.85.38.168’ Method: ACK
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]

<— SIP read from UDP:95.85.38.168:5060 —>
INVITE sip:6803046310789@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7a05a9aa6f95590b
From: sip:143javednoal@95.85.38.168;tag=634091ae00d38b86
To: sip:6803046310789@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 INVITE
Call-ID: 4b0325bc34eba8561f2aa6bc00021ae7@95.85.38.168
Contact: sip:143javednoal@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Supported: timer
Session-Expires: 600;refresher=uas
Content-Length: 210
Content-Type: application/sdp

v=0
o=- 1427150288 1427150288 IN IP4 85.17.26.219
s=VOS2009
c=IN IP4 85.17.26.219
t=0 0
m=audio 18510 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 10 lines) —
Sending to 95.85.38.168:5060 (NAT)
Using INVITE request as basis request - 4b0325bc34eba8561f2aa6bc00021ae7@95.85.38.168
Found peer ‘95.85.38.168’ for ‘143javednoal’ from 95.85.38.168:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x101 (g723|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 85.17.26.219:18510
Looking for 6803046310789 in 176960231 (domain 188.166.42.31)
list_route: hop: sip:143javednoal@95.85.38.168:5060

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7a05a9aa6f95590b;received=95.85.38.168;rport=5060
From: sip:143javednoal@95.85.38.168;tag=634091ae00d38b86
To: sip:6803046310789@188.166.42.31
Call-ID: 4b0325bc34eba8561f2aa6bc00021ae7@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803046310789@188.166.42.31:5060
Content-Length: 0

<------------>
– Executing [6803046310789@176960231:1] Dial(“SIP/95.85.38.168-000014a7”, “IAX/N691-1/6803046310789,120”) in new stack
– Called IAX/N691-1/6803046310789

<— SIP read from UDP:95.85.38.168:5060 —>
CANCEL sip:6803046310789@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7a05a9aa6f95590b
From: sip:143javednoal@95.85.38.168;tag=634091ae00d38b86
To: sip:6803046310789@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 CANCEL
Call-ID: 4b0325bc34eba8561f2aa6bc00021ae7@95.85.38.168
Contact: sip:143javednoal@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 95.85.38.168:5060 (NAT)

<— Reliably Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7a05a9aa6f95590b;received=95.85.38.168;rport=5060
From: sip:143javednoal@95.85.38.168;tag=634091ae00d38b86
To: sip:6803046310789@188.166.42.31;tag=as66b16434
Call-ID: 4b0325bc34eba8561f2aa6bc00021ae7@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7a05a9aa6f95590b;received=95.85.38.168;rport=5060
From: sip:143javednoal@95.85.38.168;tag=634091ae00d38b86
To: sip:6803046310789@188.166.42.31;tag=as66b16434
Call-ID: 4b0325bc34eba8561f2aa6bc00021ae7@95.85.38.168
CSeq: 1 CANCEL
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (176960231, 6803046310789, 1) exited non-zero on ‘SIP/95.85.38.168-000014a7’

<— SIP read from UDP:95.85.38.168:5060 —>
ACK sip:6803046310789@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7a05a9aa6f95590b
From: sip:143javednoal@95.85.38.168;tag=634091ae00d38b86
To: sip:6803046310789@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 ACK
Call-ID: 4b0325bc34eba8561f2aa6bc00021ae7@95.85.38.168
Contact: sip:143javednoal@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Retransmitting #1 (NAT) to 95.85.38.168:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7a05a9aa6f95590b;received=95.85.38.168;rport=5060
From: sip:143javednoal@95.85.38.168;tag=634091ae00d38b86
To: sip:6803046310789@188.166.42.31;tag=as66b16434
Call-ID: 4b0325bc34eba8561f2aa6bc00021ae7@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:95.85.38.168:5060 —>
ACK sip:95.85.38.168 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7a05a9aa6f95590b;received=95.85.38.168;rport=5060
From: sip:143javednoal@95.85.38.168;tag=634091ae00d38b86
To: sip:6803046310789@188.166.42.31;tag=as66b16434
Call-ID: 4b0325bc34eba8561f2aa6bc00021ae7@95.85.38.168
CSeq: 1 ACK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
User-Agent: VOS2009 V2.1.2.0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘4b0325bc34eba8561f2aa6bc00021ae7@95.85.38.168’ Method: ACK
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]

<— SIP read from UDP:95.85.38.168:5060 —>
INVITE sip:6803058403085@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7c66136926bb1730
From: sip:966599543048@95.85.38.168;tag=4caf0a6a4438c9d2
To: sip:6803058403085@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 INVITE
Call-ID: 33ec6a7d44daa1c61f2aa6bc00021ae8@95.85.38.168
Contact: sip:966599543048@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Supported: timer
Session-Expires: 600;refresher=uas
Content-Length: 180
Content-Type: application/sdp

v=0
o=- 1427150289 1427150289 IN IP4 74.201.159.82
s=VOS2009
c=IN IP4 74.201.159.82
t=0 0
m=audio 40544 RTP/AVP 18 4
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=sendrecv
<------------->
— (14 headers 9 lines) —
Sending to 95.85.38.168:5060 (NAT)
Using INVITE request as basis request - 33ec6a7d44daa1c61f2aa6bc00021ae8@95.85.38.168
Found peer ‘95.85.38.168’ for ‘966599543048’ from 95.85.38.168:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 4
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Capabilities: us - 0x101 (g723|g729), peer - audio=0x101 (g723|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x101 (g723|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 74.201.159.82:40544
Looking for 6803058403085 in 176960231 (domain 188.166.42.31)
list_route: hop: sip:966599543048@95.85.38.168:5060

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7c66136926bb1730;received=95.85.38.168;rport=5060
From: sip:966599543048@95.85.38.168;tag=4caf0a6a4438c9d2
To: sip:6803058403085@188.166.42.31
Call-ID: 33ec6a7d44daa1c61f2aa6bc00021ae8@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803058403085@188.166.42.31:5060
Content-Length: 0

<------------>
– Executing [6803058403085@176960231:1] Dial(“SIP/95.85.38.168-000014a8”, “IAX/N691-1/6803058403085,120”) in new stack
– Called IAX/N691-1/6803058403085
> Ringing channel IAX/N691-1-24
– IAX/N691-1-24 is ringing

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7c66136926bb1730;received=95.85.38.168;rport=5060
From: sip:966599543048@95.85.38.168;tag=4caf0a6a4438c9d2
To: sip:6803058403085@188.166.42.31;tag=as5c97a1bd
Call-ID: 33ec6a7d44daa1c61f2aa6bc00021ae8@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803058403085@188.166.42.31:5060
Content-Length: 0

<------------>
– IAX/N691-1-24 is making progress passing it to SIP/95.85.38.168-000014a8
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK7c66136926bb1730;received=95.85.38.168;rport=5060
From: sip:966599543048@95.85.38.168;tag=4caf0a6a4438c9d2
To: sip:6803058403085@188.166.42.31;tag=as5c97a1bd
Call-ID: 33ec6a7d44daa1c61f2aa6bc00021ae8@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803058403085@188.166.42.31:5060
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 619531060 619531060 IN IP4 188.166.42.31
s=Asterisk PBX SVN-branch-1.8-r331578M
c=IN IP4 188.166.42.31
t=0 0
m=audio 11382 RTP/AVP 18 4
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=ptime:20
a=sendrecv

<------------>
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK70650c0a38cae80e;received=95.85.38.168;rport=5060
From: sip:966583381668@95.85.38.168;tag=2c62755d162a8a35
To: sip:6803046482433@188.166.42.31;tag=as4c4eec9f
Call-ID: 08c5a12a190428661f2aa6bc00021ad6@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803046482433@188.166.42.31:5060
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1515971479 1515971480 IN IP4 188.166.42.31
s=Asterisk PBX SVN-branch-1.8-r331578M
c=IN IP4 188.166.42.31
t=0 0
m=audio 15536 RTP/AVP 18 4
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=ptime:20
a=sendrecv

<------------>
> Ringing channel IAX/N691-1-7
– IAX/N691-1-7 is ringing
– IAX/N691-1-7 is making progress passing it to SIP/95.85.38.168-000014a6
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]

<— SIP read from UDP:95.85.38.168:5060 —>
CANCEL sip:6803046482433@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK70650c0a38cae80e
From: sip:966583381668@95.85.38.168;tag=2c62755d162a8a35
To: sip:6803046482433@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 CANCEL
Call-ID: 08c5a12a190428661f2aa6bc00021ad6@95.85.38.168
Contact: sip:966583381668@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 95.85.38.168:5060 (NAT)

<— Reliably Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK70650c0a38cae80e;received=95.85.38.168;rport=5060
From: sip:966583381668@95.85.38.168;tag=2c62755d162a8a35
To: sip:6803046482433@188.166.42.31;tag=as4c4eec9f
Call-ID: 08c5a12a190428661f2aa6bc00021ad6@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK70650c0a38cae80e;received=95.85.38.168;rport=5060
From: sip:966583381668@95.85.38.168;tag=2c62755d162a8a35
To: sip:6803046482433@188.166.42.31;tag=as4c4eec9f
Call-ID: 08c5a12a190428661f2aa6bc00021ad6@95.85.38.168
CSeq: 1 CANCEL
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (176960231, 6803046482433, 1) exited non-zero on ‘SIP/95.85.38.168-00001496’

<— SIP read from UDP:95.85.38.168:5060 —>
ACK sip:6803046482433@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK70650c0a38cae80e
From: sip:966583381668@95.85.38.168;tag=2c62755d162a8a35
To: sip:6803046482433@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 ACK
Call-ID: 08c5a12a190428661f2aa6bc00021ad6@95.85.38.168
Contact: sip:966583381668@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Retransmitting #1 (NAT) to 95.85.38.168:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK70650c0a38cae80e;received=95.85.38.168;rport=5060
From: sip:966583381668@95.85.38.168;tag=2c62755d162a8a35
To: sip:6803046482433@188.166.42.31;tag=as4c4eec9f
Call-ID: 08c5a12a190428661f2aa6bc00021ad6@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:95.85.38.168:5060 —>
ACK sip:95.85.38.168 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK70650c0a38cae80e;received=95.85.38.168;rport=5060
From: sip:966583381668@95.85.38.168;tag=2c62755d162a8a35
To: sip:6803046482433@188.166.42.31;tag=as4c4eec9f
Call-ID: 08c5a12a190428661f2aa6bc00021ad6@95.85.38.168
CSeq: 1 ACK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
User-Agent: VOS2009 V2.1.2.0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘08c5a12a190428661f2aa6bc00021ad6@95.85.38.168’ Method: ACK

<— SIP read from UDP:95.85.38.168:5060 —>
INVITE sip:6803083962524@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK2aacd0fb027ac662
From: sip:966594165140@95.85.38.168;tag=370ce3027f233ff6
To: sip:6803083962524@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 INVITE
Call-ID: 0670cd122fa3f5881f2aa6bc00021ae9@95.85.38.168
Contact: sip:966594165140@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Supported: timer
Session-Expires: 600;refresher=uas
Content-Length: 180
Content-Type: application/sdp

v=0
o=- 1427150293 1427150293 IN IP4 74.201.159.82
s=VOS2009
c=IN IP4 74.201.159.82
t=0 0
m=audio 40964 RTP/AVP 18 4
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=sendrecv
<------------->
— (14 headers 9 lines) —
Sending to 95.85.38.168:5060 (NAT)
Using INVITE request as basis request - 0670cd122fa3f5881f2aa6bc00021ae9@95.85.38.168
Found peer ‘95.85.38.168’ for ‘966594165140’ from 95.85.38.168:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 4
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Capabilities: us - 0x101 (g723|g729), peer - audio=0x101 (g723|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x101 (g723|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 74.201.159.82:40964
Looking for 6803083962524 in 176960231 (domain 188.166.42.31)
list_route: hop: sip:966594165140@95.85.38.168:5060

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK2aacd0fb027ac662;received=95.85.38.168;rport=5060
From: sip:966594165140@95.85.38.168;tag=370ce3027f233ff6
To: sip:6803083962524@188.166.42.31
Call-ID: 0670cd122fa3f5881f2aa6bc00021ae9@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803083962524@188.166.42.31:5060
Content-Length: 0

<------------>
– Executing [6803083962524@176960231:1] Dial(“SIP/95.85.38.168-000014a9”, “IAX/N691-1/6803083962524,120”) in new stack
– Called IAX/N691-1/6803083962524

<— SIP read from UDP:95.85.38.168:5060 —>
INVITE sip:6803064039029@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK71936fee7d82d26d
From: sip:966598965786@95.85.38.168;tag=3ca7c7981a4240be
To: sip:6803064039029@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 INVITE
Call-ID: 0b4e022a2df0463a1f2aa6bc00021aea@95.85.38.168
Contact: sip:966598965786@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Supported: timer
Session-Expires: 600;refresher=uas
Content-Length: 180
Content-Type: application/sdp

v=0
o=- 1427150293 1427150293 IN IP4 74.201.159.82
s=VOS2009
c=IN IP4 74.201.159.82
t=0 0
m=audio 41028 RTP/AVP 18 4
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=sendrecv
<------------->
— (14 headers 9 lines) —
Sending to 95.85.38.168:5060 (NAT)
Using INVITE request as basis request - 0b4e022a2df0463a1f2aa6bc00021aea@95.85.38.168
Found peer ‘95.85.38.168’ for ‘966598965786’ from 95.85.38.168:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 4
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Capabilities: us - 0x101 (g723|g729), peer - audio=0x101 (g723|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x101 (g723|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 74.201.159.82:41028
Looking for 6803064039029 in 176960231 (domain 188.166.42.31)
list_route: hop: sip:966598965786@95.85.38.168:5060

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK71936fee7d82d26d;received=95.85.38.168;rport=5060
From: sip:966598965786@95.85.38.168;tag=3ca7c7981a4240be
To: sip:6803064039029@188.166.42.31
Call-ID: 0b4e022a2df0463a1f2aa6bc00021aea@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803064039029@188.166.42.31:5060
Content-Length: 0

<------------>
– Executing [6803064039029@176960231:1] Dial(“SIP/95.85.38.168-000014aa”, “IAX/N691-1/6803064039029,120”) in new stack
– Called IAX/N691-1/6803064039029
> Ringing channel IAX/N691-1-26
– IAX/N691-1-26 is ringing

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK2aacd0fb027ac662;received=95.85.38.168;rport=5060
From: sip:966594165140@95.85.38.168;tag=370ce3027f233ff6
To: sip:6803083962524@188.166.42.31;tag=as75a3447d
Call-ID: 0670cd122fa3f5881f2aa6bc00021ae9@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803083962524@188.166.42.31:5060
Content-Length: 0

<------------>
– IAX/N691-1-26 is making progress passing it to SIP/95.85.38.168-000014a9
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK2aacd0fb027ac662;received=95.85.38.168;rport=5060
From: sip:966594165140@95.85.38.168;tag=370ce3027f233ff6
To: sip:6803083962524@188.166.42.31;tag=as75a3447d
Call-ID: 0670cd122fa3f5881f2aa6bc00021ae9@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803083962524@188.166.42.31:5060
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1382451074 1382451074 IN IP4 188.166.42.31
s=Asterisk PBX SVN-branch-1.8-r331578M
c=IN IP4 188.166.42.31
t=0 0
m=audio 12614 RTP/AVP 18 4
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=ptime:20
a=sendrecv

<------------>
+[TCP/0.0.0.0:0]
+[UDP/0.0.0.0:0]
> Ringing channel IAX/N691-1-27
– IAX/N691-1-27 is ringing

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK71936fee7d82d26d;received=95.85.38.168;rport=5060
From: sip:966598965786@95.85.38.168;tag=3ca7c7981a4240be
To: sip:6803064039029@188.166.42.31;tag=as436b25bc
Call-ID: 0b4e022a2df0463a1f2aa6bc00021aea@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803064039029@188.166.42.31:5060
Content-Length: 0

<------------>
– IAX/N691-1-27 is making progress passing it to SIP/95.85.38.168-000014aa
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK71936fee7d82d26d;received=95.85.38.168;rport=5060
From: sip:966598965786@95.85.38.168;tag=3ca7c7981a4240be
To: sip:6803064039029@188.166.42.31;tag=as436b25bc
Call-ID: 0b4e022a2df0463a1f2aa6bc00021aea@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803064039029@188.166.42.31:5060
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1054913174 1054913174 IN IP4 188.166.42.31
s=Asterisk PBX SVN-branch-1.8-r331578M
c=IN IP4 188.166.42.31
t=0 0
m=audio 17210 RTP/AVP 18 4
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=ptime:20
a=sendrecv

<------------>
> Ringing channel IAX/N691-1-24
– IAX/N691-1-24 is ringing
– IAX/N691-1-24 is making progress passing it to SIP/95.85.38.168-000014a8
> Ringing channel IAX/N691-1-16
– IAX/N691-1-16 is ringing
– IAX/N691-1-16 is making progress passing it to SIP/95.85.38.168-0000149e

<— SIP read from UDP:95.85.38.168:5060 —>
INVITE sip:6803030614964@188.166.42.31 SIP/2.0
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK69dd5ade40ea13ca
From: sip:966593558682@95.85.38.168;tag=2502b918743b0960
To: sip:6803030614964@188.166.42.31
User-Agent: VOS2009 V2.1.2.0
CSeq: 1 INVITE
Call-ID: 77887c17695c50871f2aa6bc00021aeb@95.85.38.168
Contact: sip:966593558682@95.85.38.168:5060
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Supported: timer
Session-Expires: 600;refresher=uas
Content-Length: 180
Content-Type: application/sdp

v=0
o=- 1427150295 1427150295 IN IP4 74.201.159.82
s=VOS2009
c=IN IP4 74.201.159.82
t=0 0
m=audio 41420 RTP/AVP 18 4
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=sendrecv
<------------->
— (14 headers 9 lines) —
Sending to 95.85.38.168:5060 (NAT)
Using INVITE request as basis request - 77887c17695c50871f2aa6bc00021aeb@95.85.38.168
Found peer ‘95.85.38.168’ for ‘966593558682’ from 95.85.38.168:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 4
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Capabilities: us - 0x101 (g723|g729), peer - audio=0x101 (g723|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x101 (g723|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 74.201.159.82:41420
Looking for 6803030614964 in 176960231 (domain 188.166.42.31)
list_route: hop: sip:966593558682@95.85.38.168:5060

<— Transmitting (NAT) to 95.85.38.168:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 95.85.38.168:5060;branch=z9hG4bK69dd5ade40ea13ca;received=95.85.38.168;rport=5060
From: sip:966593558682@95.85.38.168;tag=2502b918743b0960
To: sip:6803030614964@188.166.42.31
Call-ID: 77887c17695c50871f2aa6bc00021aeb@95.85.38.168
CSeq: 1 INVITE
Server: Box1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 600;refresher=uas
Contact: sip:6803030614964@188.166.42.31:5060
Content-Length: 0

<------------>
– Executing [6803030614964@176960231:1] Dial(“SIP/95.85.38.168-000014ab”, “IAX/N691-1/6803030614964,120”) in new stack
– Called IAX/N691-1/6803030614964[/code]

mshoaib · March 23, 2015, 6:46pm