Multiple Invites Sent

Hey guys I need a guru on this one I have scoured the web and forums and have yet to find an answer and were two weeks into the project and my boss is getting very anxious on getting the server up as quickly as possible. We have an Asterisk server that is sending multiple invites and we also have no audio. The best I could find for the multiple invites is a DNS issue and I have performed the fix for enabling dnsmgr but it didnt resolve the problem. If there are any gurus out there that have a minute to look over and give some suggestions I would be very grateful.

The PBX internal address is the same as internal on a NAT

Dash and Bandwidth.com are the provider using IP based trunks.

Phone numbers have been edited for security purposes
2142222222 (Phone calling in)
12141111111@y.y.y.y (Number and PBX we are (attempting) to use reinvite to forward the call)
15555555555 (Our DID)

sip.conf

[general]
bindport = 5060
bindaddr = x.x.x.x External NAT Address
externip= x.x.x.x External NAT Address
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite
fromuser=+15555555555 (DID Number)

[Bandwidth]
type=friend
host=208.94.159.10
port=5060
;nat=force_rport,comedia
qualify=yes
context=incoming
canreinvite=no
notifyringing=yes

[BandwidthOut]
type=friend
host=67.231.8.195
port=5060
;nat=force_rport,comedia
fromuser=+15555555555 (DID Number)
qualify=no
context=outgoing
;directmedia=no
;directrtpsetup=no
canreinvite=no
notifyringing=yes
directmedia=outgoing

extensions.conf

[incoming]

;greetings and start recording
exten => _15555555555,1,Answer()
exten => _15555555555,n(start),agi(googletts.agi,"Please stand by",fr)
exten => _15555555555,n,Set(CALLERID(num)=+15555555555)
exten => _15555555555,n,Dial(SIP/BandwidthOut/12145555555)
exten => _15555555555,n,goto(end)

SIP Debug

[code]Reliably Transmitting (no NAT) to 208.94.159.10:5060:
OPTIONS sip:208.94.159.10 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2eb58820
Max-Forwards: 70
From: “asterisk” sip:asterisk@x.x.x.x;tag=as7b114ccf
To: sip:208.94.159.10
Contact: sip:asterisk@x.x.x.x:5060
Call-ID: 44310cc6164c9c954f6f631669858e32@x.x.x.x:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.0.0
Date: Fri, 11 Apr 2014 22:37:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:208.94.159.10:5060 —>
SIP/2.0 200 OK
From: “asterisk” sip:asterisk@x.x.x.x;tag=as7b114ccf
To: sip:208.94.159.10;tag=a9f5ed0-a1f960c4-0-52f722e8-0
Call-ID: 44310cc6164c9c954f6f631669858e32@x.x.x.x:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2eb58820
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘44310cc6164c9c954f6f631669858e32@x.x.x.x:5060’ Method: OPTIONS

<— SIP read from UDP:208.94.159.10:5060 —>
INVITE sip:15555555555@x.x.x.x:5060;transport=udp SIP/2.0
From: sip:+12142222222@208.94.159.10:5060;isup-oli=61;tag=c00080a-13c4-5348706e-4a5d4c85-721159e6
To: sip:15555555555@x.x.x.x:5060
Call-ID: CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10
CSeq: 1 INVITE
Via: SIP/2.0/UDP 208.94.159.10:5060;branch=z9hG4bK-17f880-5348706e-4a5d4c85-48ab8b27
Max-Forwards: 70
P-Asserted-Identity: sip:+12142222222@cxc.dashcs.com:5060
Content-Disposition: session;handling=required
Contact: sip:+12142222222@208.94.159.10:5060;isup-oli=61;maddr=208.94.159.10;transport=udp
Content-Type: application/sdp
Content-Length: 274

v=0
o=Acme_UAS 0 1 IN IP4 192.168.47.72
s=SIP Media Capabilities
c=IN IP4 67.231.4.102
t=0 0
m=audio 7866 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=maxptime:30
a=sendrecv
<------------->
— (12 headers 13 lines) —
Sending to 208.94.159.10:5060 (no NAT)
Sending to 208.94.159.10:5060 (no NAT)
Using INVITE request as basis request - CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10
Found peer ‘Bandwidth’ for ‘+12142222222’ from 208.94.159.10:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 67.231.4.102:7866
Looking for 15555555555 in incoming (domain x.x.x.x)
list_route: route/path hop: sip:+12142222222@208.94.159.10:5060;isup-oli=61;maddr=208.94.159.10;transport=udp

<— Transmitting (no NAT) to 208.94.159.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.94.159.10:5060;branch=z9hG4bK-17f880-5348706e-4a5d4c85-48ab8b27;received=208.94.159.10
From: sip:+12142222222@208.94.159.10:5060;isup-oli=61;tag=c00080a-13c4-5348706e-4a5d4c85-721159e6
To: sip:15555555555@x.x.x.x:5060
Call-ID: CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10
CSeq: 1 INVITE
Server: Asterisk PBX 12.0.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:15555555555@x.x.x.x:5060
Content-Length: 0

<------------>
– Executing [15555555555@incoming:1] Answer(“SIP/Bandwidth-00000000”, “”) in new stack
Audio is at 11132
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 208.94.159.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.94.159.10:5060;branch=z9hG4bK-17f880-5348706e-4a5d4c85-48ab8b27;received=208.94.159.10
From: sip:+12142222222@208.94.159.10:5060;isup-oli=61;tag=c00080a-13c4-5348706e-4a5d4c85-721159e6
To: sip:15555555555@x.x.x.x:5060;tag=as08b7a092
Call-ID: CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10
CSeq: 1 INVITE
Server: Asterisk PBX 12.0.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:15555555555@x.x.x.x:5060
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 1009248094 1009248094 IN IP4 x.x.x.x
s=Asterisk PBX 12.0.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 11132 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

<------------>
Retransmitting #1 (no NAT) to 208.94.159.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.94.159.10:5060;branch=z9hG4bK-17f880-5348706e-4a5d4c85-48ab8b27;received=208.94.159.10
From: sip:+12142222222@208.94.159.10:5060;isup-oli=61;tag=c00080a-13c4-5348706e-4a5d4c85-721159e6
To: sip:15555555555@x.x.x.x:5060;tag=as08b7a092
Call-ID: CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10
CSeq: 1 INVITE
Server: Asterisk PBX 12.0.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:15555555555@x.x.x.x:5060
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 1009248094 1009248094 IN IP4 x.x.x.x
s=Asterisk PBX 12.0.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 11132 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


<— SIP read from UDP:208.94.159.10:5060 —>
ACK sip:15555555555@x.x.x.x:5060 SIP/2.0
From: sip:+12142222222@208.94.159.10:5060;isup-oli=61;tag=c00080a-13c4-5348706e-4a5d4c85-721159e6
To: sip:15555555555@x.x.x.x:5060;tag=as08b7a092
Call-ID: CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10
CSeq: 1 ACK
Via: SIP/2.0/UDP 208.94.159.10:5060;branch=z9hG4bK-17f886-5348706e-4a5d4d19-3366897a
Max-Forwards: 67
Contact: sip:+12142222222@208.94.159.10:5060;isup-oli=61;maddr=208.94.159.10;transport=udp
Content-Length: 0

<------------->
— (9 headers 0 lines) —
– Executing [15555555555@incoming:2] AGI(“SIP/Bandwidth-00000000”, “googletts.agi,“Please stand by”,fr”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/googletts.agi
Use of uninitialized value $AGI{“arg_3”} in length at /var/lib/asterisk/agi-bin/googletts.agi line 140, line 23.
Use of uninitialized value $AGI{“arg_4”} in length at /var/lib/asterisk/agi-bin/googletts.agi line 145, line 23.
– Playing ‘/tmp/07179782a7e10a85bab49db2b72e5614’ (escape_digits=) (sample_offset 0)
> 0xe838c0 – Probation passed - setting RTP source address to 67.231.4.102:7866
– <SIP/Bandwidth-00000000>AGI Script googletts.agi completed, returning 0
– Executing [15555555555@incoming:3] Set(“SIP/Bandwidth-00000000”, “CALLERID(num)=+15555555555”) in new stack
– Executing [15555555555@incoming:4] Dial(“SIP/Bandwidth-00000000”, “SIP/BandwidthOut/12141111111”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 14832
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 67.231.8.195:5060:
INVITE sip:12141111111@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK46f272b6
Max-Forwards: 70
From: sip:+15555555555@x.x.x.x;tag=as028ed2dd
To: sip:12141111111@y.y.y.y:5060
Contact: sip:+15555555555@x.x.x.x:5060
Call-ID: 0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.0.0
Date: Fri, 11 Apr 2014 22:37:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 4123066 4123066 IN IP4 x.x.x.x
s=Asterisk PBX 12.0.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14832 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


-- Called SIP/BandwidthOut/12141111111

Retransmitting #1 (no NAT) to 67.231.8.195:5060:
INVITE sip:12141111111@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK46f272b6
Max-Forwards: 70
From: sip:+15555555555@x.x.x.x;tag=as028ed2dd
To: sip:12141111111@y.y.y.y:5060
Contact: sip:+15555555555@x.x.x.x:5060
Call-ID: 0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.0.0
Date: Fri, 11 Apr 2014 22:37:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 4123066 4123066 IN IP4 x.x.x.x
s=Asterisk PBX 12.0.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14832 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


Retransmitting #2 (no NAT) to 67.231.8.195:5060:
INVITE sip:12141111111@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK46f272b6
Max-Forwards: 70
From: sip:+15555555555@x.x.x.x;tag=as028ed2dd
To: sip:12141111111@y.y.y.y:5060
Contact: sip:+15555555555@x.x.x.x:5060
Call-ID: 0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.0.0
Date: Fri, 11 Apr 2014 22:37:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 4123066 4123066 IN IP4 x.x.x.x
s=Asterisk PBX 12.0.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14832 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


Retransmitting #3 (no NAT) to 67.231.8.195:5060:
INVITE sip:12141111111@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK46f272b6
Max-Forwards: 70
From: sip:+15555555555@x.x.x.x;tag=as028ed2dd
To: sip:12141111111@y.y.y.y:5060
Contact: sip:+15555555555@x.x.x.x:5060
Call-ID: 0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.0.0
Date: Fri, 11 Apr 2014 22:37:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 4123066 4123066 IN IP4 x.x.x.x
s=Asterisk PBX 12.0.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14832 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


Retransmitting #4 (no NAT) to 67.231.8.195:5060:
INVITE sip:12141111111@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK46f272b6
Max-Forwards: 70
From: sip:+15555555555@x.x.x.x;tag=as028ed2dd
To: sip:12141111111@y.y.y.y:5060
Contact: sip:+15555555555@x.x.x.x:5060
Call-ID: 0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.0.0
Date: Fri, 11 Apr 2014 22:37:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 4123066 4123066 IN IP4 x.x.x.x
s=Asterisk PBX 12.0.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14832 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


Retransmitting #5 (no NAT) to 67.231.8.195:5060:
INVITE sip:12141111111@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK46f272b6
Max-Forwards: 70
From: sip:+15555555555@x.x.x.x;tag=as028ed2dd
To: sip:12141111111@y.y.y.y:5060
Contact: sip:+15555555555@x.x.x.x:5060
Call-ID: 0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.0.0
Date: Fri, 11 Apr 2014 22:37:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 4123066 4123066 IN IP4 x.x.x.x
s=Asterisk PBX 12.0.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14832 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


Retransmitting #6 (no NAT) to 67.231.8.195:5060:
INVITE sip:12141111111@y.y.y.y:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK46f272b6
Max-Forwards: 70
From: sip:+15555555555@x.x.x.x;tag=as028ed2dd
To: sip:12141111111@y.y.y.y:5060
Contact: sip:+15555555555@x.x.x.x:5060
Call-ID: 0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.0.0
Date: Fri, 11 Apr 2014 22:37:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 4123066 4123066 IN IP4 x.x.x.x
s=Asterisk PBX 12.0.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14832 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


Scheduling destruction of SIP dialog ‘0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060’ in 32000 ms (Method: INVITE)
[Apr 11 16:38:04] WARNING[19359]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission 0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Apr 11 16:38:04] WARNING[19359]: chan_sip.c:4288 retrans_pkt: Hanging up call 0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
– SIP/BandwidthOut-00000001 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [15555555555@incoming:5] Goto(“SIP/Bandwidth-00000000”, “end”) in new stack
– Goto (incoming,15555555555,48)
– Executing [15555555555@incoming:48] Hangup(“SIP/Bandwidth-00000000”, “”) in new stack
== Spawn extension (incoming, 15555555555, 48) exited non-zero on 'SIP/Bandwidth-00000000’
Scheduling destruction of SIP dialog ‘CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:+12142222222@208.94.159.10:5060;isup-oli=61;maddr=208.94.159.10;transport=udp for address/port to send to
set_destination: set destination to 208.94.159.10:5060
Reliably Transmitting (no NAT) to 208.94.159.10:5060:
BYE sip:+12142222222@208.94.159.10:5060;isup-oli=61;maddr=208.94.159.10;transport=udp SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK3a951ffe
Max-Forwards: 70
From: sip:15555555555@x.x.x.x:5060;tag=as08b7a092
To: sip:+12142222222@208.94.159.10:5060;isup-oli=61;tag=c00080a-13c4-5348706e-4a5d4c85-721159e6
Call-ID: CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10
CSeq: 102 BYE
User-Agent: Asterisk PBX 12.0.0
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


Really destroying SIP dialog ‘0203f0fa14cd73d4120b44421d1ec81f@x.x.x.x:5060’ Method: INVITE
Retransmitting #1 (no NAT) to 208.94.159.10:5060:
BYE sip:+12142222222@208.94.159.10:5060;isup-oli=61;maddr=208.94.159.10;transport=udp SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK3a951ffe
Max-Forwards: 70
From: sip:15555555555@x.x.x.x:5060;tag=as08b7a092
To: sip:+12142222222@208.94.159.10:5060;isup-oli=61;tag=c00080a-13c4-5348706e-4a5d4c85-721159e6
Call-ID: CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10
CSeq: 102 BYE
User-Agent: Asterisk PBX 12.0.0
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


<— SIP read from UDP:208.94.159.10:5060 —>
SIP/2.0 200 OK
From: sip:15555555555@x.x.x.x:5060;tag=as08b7a092
To: sip:+12142222222@208.94.159.10;isup-oli=61;tag=c00080a-13c4-5348706e-4a5d4c85-721159e6
Call-ID: CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10
CSeq: 102 BYE
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK3a951ffe
Content-Length: 0

<------------->
— (7 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘CXC-74-7cda6318-c00080a-13c4-5348706e-4a5d4c85-6b41ab28@208.94.159.10’ Method: ACK
Reliably Transmitting (no NAT) to 208.94.159.10:5060:
OPTIONS sip:208.94.159.10 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK05309921
Max-Forwards: 70
From: “asterisk” sip:asterisk@x.x.x.x;tag=as5f1bf855
To: sip:208.94.159.10
Contact: sip:asterisk@x.x.x.x:5060
Call-ID: 7c3bce437085c0a472bc6412655a4271@x.x.x.x:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 12.0.0
Date: Fri, 11 Apr 2014 22:38:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:208.94.159.10:5060 —>
SIP/2.0 200 OK
From: “asterisk” sip:asterisk@x.x.x.x;tag=as5f1bf855
To: sip:208.94.159.10;tag=a9f5ed0-a58d4bc9-0-52f80d60-0
Call-ID: 7c3bce437085c0a472bc6412655a4271@x.x.x.x:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK05309921
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘7c3bce437085c0a472bc6412655a4271@x.x.x.x:5060’ Method: OPTIONS
[/code]

There is nothing wrong on the Asterisk side.

You might not have a route to y.y.y.y.

They may not have a route to x.x.x.x

A firewall may be blocking port 5060 traffic.

You might not have a port forwarding rule from x.x.x to you internal machine.

DNS problems seem unlikely, and they will be affecting the machine at y.y.y.y, not your Asterisk system.

You do seem to have a flakey local network, given the number of retransmissions of the 200 OK.

Thank you for the response I have verified that x and y can communicate and can resolve.

I’m told the firewall is setup with a pass-through configuration so its sitting on the web so we don’t have to worry about ports being blocked.

I have continued with the debugging and also worked to get a live test with the vendor and they do see the multiple invites and say they are sending a 183 trying message that my server ignores and keeps sending invites most likely because its looking for a 200 ack.

The vendor does seem to be using a proxy for the outbound dialing.

With this new knowledge how do i configure asterisk to accept the 183 trying and are there any specific proxy settings ill need to configure?

183 is progress, not trying.

To get Asterisk to accept it you have to configure your network so that it reaches Asterisk.

Hi David,

Thanks for your help it seems the bossman has blocked the IP that he gave me in documentation to use and I didnt have access to the firewall to check and couldn’t get a valid answer on even if it was NAT’d or what was in the ACL to verify. You were right on in determining it was a network issue.

Heres how we found the exact issue, after debugging we found out by using tcpdump and capturing the packets and verified against the voip vendor that they were actually sending the ringing response and i was receiving no traffic back from the trunk so asterisk would send multiple invites waiting for the ringing 183 response.

Thanks,
John Kent