Multiple Invites Sent

jkent · April 11, 2014, 10:56pm

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