SIP Incoming Calls Fail 3

VoIPP · Aug 6, 2012

R8.1

I have all ports open to the IPO. I can make outgoing calls but incoming calls fail. Monitor shows me the call arriving but the IPO never responds to the call. I never see a Tx packet just Rx.

ICR to AA and then Huntgroup.

So what am I missing?

intrigrant · Aug 6, 2012

VoIPP said:
So what am I missing?

Incoming calls?

Do you have a URI with all *'s for contact info etc?

If it ain't dutch it ain't much

VoIPP · Aug 6, 2012

URI is set to "Use Credentials User Name".

intrigrant · Aug 6, 2012

Add a URI with all contact fields with a * and incoming line group set to zero.
Then try again.

If it ain't dutch it ain't much

kholladay · Aug 6, 2012

If you can post a trace of the inbound SIP call it will help. Follow this guide and post the results.

http://www.ipofficehelp.com/HOWTO_ConfigureMonitorComprehensive.aspx#SIPTroubleshoot

Kyle Holladay / IPOfficeHelp.com
ACSS/ACIS/APSS Avaya SME Communications
APDS Avaya Data
MCP/MCTS Exchange 2007/2010
Adtran ATSA, Aruba ACMA

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford

VoIPP · Aug 7, 2012

This is the Monitor trace after I setup a URI with all '*****'.

263441mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
INVITE sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK2edb006e
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as5e45febb
To: <sip:9205602433@172.16.8.6:5060>
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 5829e53859dba2663a45d3f870741b97@66.241.96.109
CSeq: 102 INVITE
User-Agent: packetrino
Max-Forwards: 70
Date: Tue, 07 Aug 2012 12:28:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 27878 27878 IN IP4 66.241.96.109
s=session
c=IN IP4 66.241.96.109
t=0 0
m=audio 18426 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv
264439mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
INVITE sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK2edb006e
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as5e45febb
To: <sip:9205602433@172.16.8.6:5060>
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 5829e53859dba2663a45d3f870741b97@66.241.96.109
CSeq: 102 INVITE
User-Agent: packetrino
Max-Forwards: 70
Date: Tue, 07 Aug 2012 12:28:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 27878 27878 IN IP4 66.241.96.109
s=session
c=IN IP4 66.241.96.109
t=0 0
m=audio 18426 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv
265438mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
INVITE sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK2edb006e
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as5e45febb
To: <sip:9205602433@172.16.8.6:5060>
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 5829e53859dba2663a45d3f870741b97@66.241.96.109
CSeq: 102 INVITE
User-Agent: packetrino
Max-Forwards: 70
Date: Tue, 07 Aug 2012 12:28:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 27878 27878 IN IP4 66.241.96.109
s=session
c=IN IP4 66.241.96.109
t=0 0
m=audio 18426 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv

********** SysMonitor v10.1 (43) [connected to 172.16.8.6 (00E00700000D)] **********
265832mS PRN: Monitor Status IP 500 V2 8.1(43)
265832mS PRN: LAW=U PRI=0, BRI=0, ALOG=4, VCOMP=10, MDM=0, WAN=0, MODU=0 LANM=0 CkSRC=0 VMAIL=1(VER=2 TYP=3) 1-X=0 CALLS=0(TOT=1)
267439mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
INVITE sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK2edb006e
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as5e45febb
To: <sip:9205602433@172.16.8.6:5060>
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 5829e53859dba2663a45d3f870741b97@66.241.96.109
CSeq: 102 INVITE
User-Agent: packetrino
Max-Forwards: 70
Date: Tue, 07 Aug 2012 12:28:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 27878 27878 IN IP4 66.241.96.109
s=session
c=IN IP4 66.241.96.109
t=0 0
m=audio 18426 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv
271625mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
INVITE sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK5ad305b7
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as514d9f2b
To: <sip:9205602433@172.16.8.6:5060>
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 565e617c062c80e01f8eb9e94ac18b72@66.241.96.109
CSeq: 102 INVITE
User-Agent: packetrino
Max-Forwards: 70
Date: Tue, 07 Aug 2012 12:28:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 27878 27878 IN IP4 66.241.96.109
s=session
c=IN IP4 66.241.96.109
t=0 0
m=audio 18382 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv
272622mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
INVITE sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK5ad305b7
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as514d9f2b
To: <sip:9205602433@172.16.8.6:5060>
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 565e617c062c80e01f8eb9e94ac18b72@66.241.96.109
CSeq: 102 INVITE
User-Agent: packetrino
Max-Forwards: 70
Date: Tue, 07 Aug 2012 12:28:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 27878 27878 IN IP4 66.241.96.109
s=session
c=IN IP4 66.241.96.109
t=0 0
m=audio 18382 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv
273622mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
INVITE sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK5ad305b7
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as514d9f2b
To: <sip:9205602433@172.16.8.6:5060>
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 565e617c062c80e01f8eb9e94ac18b72@66.241.96.109
CSeq: 102 INVITE
User-Agent: packetrino
Max-Forwards: 70
Date: Tue, 07 Aug 2012 12:28:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 27878 27878 IN IP4 66.241.96.109
s=session
c=IN IP4 66.241.96.109
t=0 0
m=audio 18382 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv
275621mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
INVITE sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK5ad305b7
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as514d9f2b
To: <sip:9205602433@172.16.8.6:5060>
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 565e617c062c80e01f8eb9e94ac18b72@66.241.96.109
CSeq: 102 INVITE
User-Agent: packetrino
Max-Forwards: 70
Date: Tue, 07 Aug 2012 12:28:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 27878 27878 IN IP4 66.241.96.109
s=session
c=IN IP4 66.241.96.109
t=0 0
m=audio 18382 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv
317154mS PRN: LoadCountControl: Marked As Successful Boot2
320569mS SIP Tx: UDP 172.16.8.6:5060 -> 64.2.142.93:5060
OPTIONS sip:Unknown@vitelity.net SIP/2.0
Via: SIP/2.0/UDP 172.16.8.6:5060;rport;branch=z9hG4bK5c6309eab3b610d043a1f4d9b393dd8b
From: <sip:Unknown@vitelity.net>;tag=d2bb1fbf66878138
To: <sip:Unknown@vitelity.net>
Call-ID: 3aeeeed455e4dadd82a7243129facce7
CSeq: 262645518 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
User-Agent: IP Office 8.1 (43)
Content-Length: 0

320623mS SIP Rx: UDP 64.2.142.93:5060 -> 172.16.8.6:5060
SIP/2.0 200 OPTIONS is almost as pointless as T38
Via: SIP/2.0/UDP 172.16.8.6:5060;rport=5060;branch=z9hG4bK5c6309eab3b610d043a1f4d9b393dd8b
From: <sip:Unknown@vitelity.net>;tag=d2bb1fbf66878138
To: <sip:Unknown@vitelity.net>;tag=37c906215f6623e2b0c0b8aa47fb6fb6.9ddc
Call-ID: 3aeeeed455e4dadd82a7243129facce7
CSeq: 262645518 OPTIONS
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0

********** SysMonitor v10.1 (43) [connected to 172.16.8.6 (00E00700000D)] **********
326673mS PRN: Monitor Status IP 500 V2 8.1(43)
326673mS PRN: LAW=U PRI=0, BRI=0, ALOG=4, VCOMP=10, MDM=0, WAN=0, MODU=0 LANM=0 CkSRC=0 VMAIL=1(VER=2 TYP=3) 1-X=0 CALLS=0(TOT=1)

hairlessupportmonkey · Aug 7, 2012

Do you have the STUN client enabled?

ACSS - SME
General Geek

VoIPP · Aug 7, 2012

No I didn't have STUN turned on but I am not sure that I need it. I've never setup a SIP trunk like this before. This is a static IP SIP so I don't think the ipo even registers with the ITSP.

tlpeter · Aug 7, 2012

This line is interesting and i think th source of the problem (i have never seen this)

SIP/2.0 200 OPTIONS is almost as pointless as T38

BAZINGA!

I'm not insane, my mother had me tested!

VoIPP · Aug 7, 2012

It is

http://www.vitelity.net.

I've used them for years and haven't really had too many issues and at $1.49/month for DID it's hard to beat.

Not sure why this line is giving so much toruble though.

amriddle01 · Aug 7, 2012

Really, that sounds quite expensive for a DID/DDI, well not exactly stunningly cheap

VoIPP · Aug 7, 2012

If someone could recommend a SIP provider in North America that plays well with the IPO it would be apreciated.

redphone · Aug 7, 2012

According to your trace the OPTIONS message you are being sent to IP address 64.2.142.93.

Inbound calls are coming from IP address 66.241.96.109.

IP Office will not answer an INVITE if it does not recognize the source IP address. This is the default "Association Method" setting.

You could try changing the Association Method to "To header hostpart against ITSP domain" and change your ITSP Domain Name to 172.16.8.6. That should match the TO header and still hopefully allow for outbound calls to work.

If not, you may need to build a separate trunk for inbound calls and set the ITSP Proxy Address to 66.241.96.109. It appears from what you have provided so far that the ITSP IP address for inbound calls is different from outbound calls.

trinetintl · Aug 7, 2012

For a second there I thought $1.49 per month? Is it too good to be true but it is...they charge you by the minute...! However I see they have Mexico numbers...I am going to take a closer look later. [dazed]

RE

http://www.trinet-intl.com

APSS - SME
ACIS - SME

cruzades · Aug 7, 2012

Did you separated this account into 2 different SIP Line?

VoIPP · Aug 7, 2012

redphone, I think I made some progress with your second suggestion by setting the Proxy to 66.241.96.109. The IPO is now at least trying to respond to the incoming call but it is sending " SIP/2.0 404 Not Found".

333948mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
INVITE sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK05c34dc2
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as1cdaf159
To: <sip:9205602433@172.16.8.6:5060>
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 61d1a623041e6ad00455947911f1dd93@66.241.96.109
CSeq: 102 INVITE
User-Agent: packetrino
Max-Forwards: 70
Date: Wed, 08 Aug 2012 01:28:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 27878 27878 IN IP4 66.241.96.109
s=session
c=IN IP4 66.241.96.109
t=0 0
m=audio 19970 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv
333954mS SIP Tx: UDP 172.16.8.6:5060 -> 66.241.96.109:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK05c34dc2
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as1cdaf159
To: <sip:9205602433@172.16.8.6:5060>;tag=f934e1eede18e07f
Call-ID: 61d1a623041e6ad00455947911f1dd93@66.241.96.109
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (43)
Content-Length: 0

333956mS SIP Tx: UDP 172.16.8.6:5060 -> 66.241.96.109:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK05c34dc2
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as1cdaf159
To: <sip:9205602433@172.16.8.6:5060>;tag=f934e1eede18e07f
Call-ID: 61d1a623041e6ad00455947911f1dd93@66.241.96.109
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Server: IP Office 8.1 (43)
Content-Length: 0

334015mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
ACK sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK05c34dc2
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as1cdaf159
To: <sip:9205602433@172.16.8.6:5060>;tag=f934e1eede18e07f
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 61d1a623041e6ad00455947911f1dd93@66.241.96.109
CSeq: 102 ACK
User-Agent: packetrino
Max-Forwards: 70
Content-Length: 0

334248mS SIP Rx: UDP 66.241.96.109:5060 -> 172.16.8.6:5060
INVITE sip:9205602433@172.16.8.6:5060 SIP/2.0
Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK2b11c83c
From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as381f08f1
To: <sip:9205602433@172.16.8.6:5060>
Contact: <sip:7154983971@66.241.96.109>
Call-ID: 2630a8c95141f09835d6e29d349d3a5c@66.241.96.109
CSeq: 102 INVITE
User-Agent: packetrino
Max-Forwards: 70
Date: Wed, 08 Aug 2012 01:28:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 336

tlpeter · Aug 7, 2012

I see this:

Code:

SIP/2.0 404 Not Found
 Via: SIP/2.0/UDP 66.241.96.109:5060;rport;branch=z9hG4bK05c34dc2
 From: "7154983971" <sip:7154983971@66.241.96.109>;tag=as1cdaf159
 To: <sip:9205602433@172.16.8.6:5060>;tag=f934e1eede18e07f

Usually a 404 means no incoming URI programmed.

BAZINGA!

I'm not insane, my mother had me tested!

cruzades · Aug 7, 2012

What have you got with your ITSP and Proxy?

e.g.

Code:

ITSP Domain=64.2.142.93
Proxy=66.241.96.109

clarkstyx18 · Aug 8, 2012

have you added an incoming call router and place . as the incoming number?

hairlessupportmonkey · Aug 8, 2012

> have you added an incoming call router and place . as the incoming number?

Eh?

A blank route from the respective LG ID to a valid extn or group will suffice so long as you have */*/* in your URI

ACSS - SME
General Geek

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