web statisticsweb stats

Business Phone Systems

Previous Thread
Next Thread
Print Thread
Rate Thread
#89462 04/08/08 05:40 AM
Joined: Mar 2006
Posts: 5
Member
OP Offline
Member
Joined: Mar 2006
Posts: 5
OK: I have attempted to set up one SIP line and one incomming call route for the SIP line.

The inbound SIP provider URL is: 207.210.100.227
The IP public IP address of the recieving network is: 71.43.176.52

I have pasted below info from Monitor on what happens when an inbound SIP call hits the IP500.

No matter which number is called via the inbound SIP trunk - the IP500 ends up saying "SIP Service Unavailable".

I would appreciate your expert opinion on what I can do to troubleshoot further:

Via: SIP/2.0/UDP 207.210.100.227:5060;branch=z9hG4bK62A29C1C

From: "17875386938" <sip:[email protected]>;tag=194DC317-1B88

To: <sip:[email protected]>;tag=c1f1ba1de698d4c4

Call-ID: [email protected]

CSeq: 101 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO

Content-Length: 0


3244187mS SIP Tx: UDP 192.168.30.10:5060 -> 207.210.100.227:5060

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 207.210.100.227:5060;branch=z9hG4bK62A29C1C

From: "17875386938" <sip:[email protected]>;tag=194DC317-1B88

To: <sip:[email protected]>;tag=c1f1ba1de698d4c4

Call-ID: [email protected]

CSeq: 101 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO

Content-Length: 0


3244187mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 0

3244187mS SipDebugInfo: SIP: ProcessInbound Message

3244187mS SipDebugInfo: Find End Point [email protected]

3244188mS SipDebugInfo: SIP Line (17): Cannot free Txn Key 2015

3244477mS SipDebugInfo: Sip_sendToNetwork packet of length 359

3244477mS SipDebugInfo: SIP Line (17): SendToTarget cfd264e3, 5060

3244477mS SIP Trunk: 17:Tx

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 207.210.100.227:5060;branch=z9hG4bK62A22107D

From: "8001" <sip:[email protected]>;tag=194D9B2A-D22

To: <sip:[email protected]>;tag=e6fa78dd92e2566c

Call-ID: [email protected]

CSeq: 101 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO

Content-Length: 0


3244478mS SIP Tx: UDP 192.168.30.10:5060 -> 207.210.100.227:5060

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 207.210.100.227:5060;branch=z9hG4bK62A22107D

From: "8001" <sip:[email protected]>;tag=194D9B2A-D22

To: <sip:[email protected]>;tag=e6fa78dd92e2566c

Call-ID: [email protected]

CSeq: 101 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO

Content-Length: 0


3245965mS SIP Rx: UDP 207.210.100.227:49408 -> 192.168.30.10:5060

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 207.210.100.227:5060;branch=z9hG4bK62A22107D

From: "8001" <sip:[email protected]>;tag=194D9B2A-D22

To: <sip:[email protected]>

Date: Mon, 07 Jul 2003 12:05:56 GMT

Call-ID: [email protected]

Supported: 100rel,timer

Min-SE: 1800

Cisco-Guid: 696802366-2947158487-2732653801-833627292

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Remote-Party-ID: "8001" <sip:[email protected]>;party=calling;screen=no;privacy=off

Timestamp: 1057579556

Contact: <sip:[email protected]:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 371


v=0

o=CiscoSystemsSIP-GW-UserAgent 6929 3557 IN IP4 207.210.100.227

s=SIP Call

c=IN IP4 207.210.100.227

t=0 0

m=audio 16468 RTP/AVP 18 0 8 101 101

c=IN IP4 207.210.100.227

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

3245966mS SIP Trunk: 17:Rx

INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 207.210.100.227:5060;branch=z9hG4bK62A22107D

From: "8001" <sip:[email protected]>;tag=194D9B2A-D22

To: <sip:[email protected]>

Date: Mon, 07 Jul 2003 12:05:56 GMT

Call-ID: [email protected]

Supported: 100rel,timer

Min-SE: 1800

Cisco-Guid: 696802366-2947158487-2732653801-833627292

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Remote-Party-ID: "8001" <sip:[email protected]>;party=calling;screen=no;privacy=off

Timestamp: 1057579556

Contact: <sip:[email protected]:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 371


v=0

o=CiscoSystemsSIP-GW-UserAgent 6929 3557 IN IP4 207.210.100.227

s=SIP Call

c=IN IP4 207.210.100.227

t=0 0

m=audio 16468 RTP/AVP 18 0 8 101 101

c=IN IP4 207.210.100.227

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

3245966mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget

3245969mS SipDebugInfo: Sip_sendToNetwork packet of length 359

3245970mS SipDebugInfo: SIP Line (17): SendToTarget cfd264e3, 5060

3245970mS SIP Trunk: 17:Tx

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 207.210.100.227:5060;branch=z9hG4bK62A22107D

Avaya IP Office Help & Support Website
IP Office Help

Avaya IP Office Help & Support Website


FAQs, documentation, videos, updates, and support for the Avaya IP Office business phone system!
Everything you need to know about installing, upgrading, and troubleshooting IP 500v2 and IPO Server Edition systems.

Joined: Apr 2006
Posts: 254
Member
Offline
Member
Joined: Apr 2006
Posts: 254
The first question would be.... do you have a SIP license. Silly I know, but people forget these things sometimes.

The other thing to check would be your "Network topology" settings. The IPO is sending out the private IP address to the other end.

Joined: May 2004
Posts: 1,663
Likes: 4
Moderator-Avaya
*****
Offline
Moderator-Avaya
*****
Joined: May 2004
Posts: 1,663
Likes: 4
Aske the SIP provider if you are registering with their servers. Also, do you have the SIP ports opened up?


Link Copied to Clipboard
Forum Statistics
Forums84
Topics94,284
Posts638,772
Members49,765
Most Online5,661
May 23rd, 2018
Popular Topics(Views)
211,461 Shoretel
188,329 CTX100 install
187,093 1a2 system
Newest Members
Nadisale, andreww, gohunt, Darrick, telecopippo
49,764 Registered Users
Top Posters(30 Days)
Toner 23
teleco 5
jc2it 4
dans 3
Who's Online Now
0 members (), 145 guests, and 289 robots.
Key: Admin, Global Mod, Mod
Contact Us | Sponsored by Atcom: One of the best VoIP Phone Canada Suppliers for your business telephone system!| Terms of Service

Sundance Communications is not affiliated with any of the above manufacturers. Sundance Phone System Forums - VOIP & Cloud Phone Help
©Copyright Sundance Communications 1998-2024
Powered by UBB.threads™ PHP Forum Software 7.7.5