[SIPForum-discussion] Soft phone can not call IP phone(Urgently, need your help)

Sat Jan 19 14:22:14 UTC 2008

Dear All,

Now the problem what I am facing is that: the softphone can connect to soft phone and the IP phone can connect to IP phone, while, the soft phone can not call IP phone(the IP phone can not call the soft phone either). Below is the description and any body have any idea on this?

1. Architecture
(1).Sip server: Minisipserver(Sever Adress: 192.168.1.73)
(2) Terminal:
 Softphone: 2pcs （X-Lite）Extension: 100, 101
 IP Phone:2pcs (Avaya 4610SW) Extention: 200(IP: 192.168.1.177), 202 (IP:192.168.1.137)

2.Problem
1．All the phone can register to the server

2.  Soft phone can call soft phone and IP phone can call IP phone( actually, even withou the server the IP phone can call IP phone).

3. Soft phone can not call IP phone(IP phone can not call soft phone either).

3. Information Traced:
When I used the soft phone(extension:100) call the IP phone( extention:200) there is no any response from the IP phone. Below is the information I traced: 

01/19/08 21:33:09 | REGISTER sip:192.168.1.73:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.177:5060;branch=z9hG4bKcd701523a
Max-Forwards: 70
Content-Length: 0
To: sip:200 at 192.168.1.73
From: sip:200 at 192.168.1.73;tag=fca1e4c1edb6180
Call-ID: 109e97c91c8528242e5b013492264154 at 192.168.1.73
CSeq: 362998185 REGISTER
Contact: <sip:200 at 192.168.1.177;srcadr=192.168.1.177>;q=1;expires=120
Authorization:Digest response="e5a54193623f30e7109fab40877ca767",username="200",realm="myvoipapp.com",nonce="610b71d773e45e343b0d440b048510b8",algorithm=MD5,uri="sip:192.168.1.73:5060"
User-Agent: Avaya 4600SW 7.2.2040 00073BC4BCF1 MxSF/v3.2.6.26

01/19/08 21:33:09 | MSS sends following SIP message to peer GW or UA(192.168.1.177:5060):
01/19/08 21:33:09 | SIP/2.0 200 OK
From: sip:200 at 192.168.1.73;tag=fca1e4c1edb6180
To: sip:200 at 192.168.1.73;tag=1b191fe1
CSeq: 362998185 REGISTER
Call-ID: 109e97c91c8528242e5b013492264154 at 192.168.1.73
Allow: INVITE, ACK, CANCEL, BYE, REGISTER
Via: SIP/2.0/UDP 192.168.1.177:5060;branch=z9hG4bKcd701523a
Contact: <sip:200 at 192.168.1.177;srcadr=192.168.1.177>;q=1;expires=120
Content-Length: 0

01/19/08 21:33:13 | REGISTER sip:192.168.1.73:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.177:5060;branch=z9hG4bK700f5c9c4
Max-Forwards: 70
Content-Length: 0
To: sip:202 at 192.168.1.73
From: sip:202 at 192.168.1.73;tag=d8a3460a8a45de5
Call-ID: 322bafbe9845965e2eee42780bc29bc4 at 192.168.1.73
CSeq: 937049281 REGISTER
Contact: <sip:202 at 192.168.1.177;srcadr=192.168.1.177>;q=1;expires=120
Authorization:Digest response="95441834796b1c11034add1cf115bb04",username="202",realm="myvoipapp.com",nonce="4f7862b420791bc81e362af2487b1102",algorithm=MD5,uri="sip:192.168.1.73:5060"
User-Agent: Avaya 4600SW 7.2.2040 00073BC4BCF1 MxSF/v3.2.6.26

01/19/08 21:33:13 | MSS sends following SIP message to peer GW or UA(192.168.1.177:5060):
01/19/08 21:33:13 | SIP/2.0 200 OK
From: sip:202 at 192.168.1.73;tag=d8a3460a8a45de5
To: sip:202 at 192.168.1.73;tag=46da4160
CSeq: 937049281 REGISTER
Call-ID: 322bafbe9845965e2eee42780bc29bc4 at 192.168.1.73
Allow: INVITE, ACK, CANCEL, BYE, REGISTER
Via: SIP/2.0/UDP 192.168.1.177:5060;branch=z9hG4bK700f5c9c4
Contact: <sip:202 at 192.168.1.177;srcadr=192.168.1.177>;q=1;expires=120
Content-Length: 0

Thanks
With Best Regards
Daniel Zeng

From: dyork at voxeo.comDate: Fri, 18 Jan 2008 11:29:09 -0500To: marc.robins at sipforum.orgCC: discussion at sipforum.orgSubject: Re: [SIPForum-discussion] Notice of Upcoming SIP Forum SIPconnect Compliance WorkshopMarc, 

I'll see you down there!  Hope to see other SIP Forum members as well.

Regards,
Dan

On Jan 9, 2008, at 5:35 PM, Marc Robins wrote:

Dear SIP Forum Members: 
The SIP Forum is proud to announce that it will be running a FREE half-day workshop on SIPconnect Compliance on January 25th, 2008, in conjunction with the Internet Telephony Conference and EXPO EAST 2008 and the Ingate SIP Trunking Seminar series.
Please note that although the workshop is free, you must be registered as an attendee of the show in order to attend the workshop. Here's the link to the conference website -- if you click "Register Now", you'll be presented with a number of options, including a free VIP pass:
http://www.tmcnet.com/voip/conference/ 
We hope to see you there! 
************************* Marc Robins Managing Director, SIP Forum www.sipforum.org marc.robins at sipforum.org 
Chief Evangelism Officer, RCG Tel: 718-548-7245 Fax: 484-952-2470 Mobile: 203-829-6307 SkypeMe! marcrobins www.robinsconsult.com 
************************* 
Full Workshop Agenda follows: 
******************************************************************************************************************** 
SIP Forum SIPconnect Compliance Workshop
Schedule, Speakers and Topics
Date: Friday, January 25, 2008 -- 10:00am-1:00pm 
Co-located with TMC’s Internet Telephony Conference & EXPO EAST 2008 The Miami Beach Convention Center, Room B114/115 Miami, FL 
The SIP Forum is an IP communications industry association that engages in numerous activities that advance and promote SIP technology, such as the development of industry recommendations, the SIPit interoperability and testing events, special interoperability workshops, and general promotion of SIP in the industry.
One of the Forum's most important technical activities is the development of the SIPconnect Technical Recommendation -- a standards-based recommendation that provides detailed guidelines for direct IP peering and interoperability (SIP Trunking) between IP PBXs and VoIP service provider networks, and the SIPconnect Compliant Certification Program through which eligible companies can license the use of the SIP Forum's 'SIPconnect Compliant' certification mark -- the official brand of the leading standard for SIP Trunking products and services.
This special educational workshop will provide a full review of the SIPconnect Technical Recommendation, real-world insights into current SIP Trunking deployments, and a full understanding of the process and requirements of SIPconnect Compliant Certification.
Detailed Schedule: 
10:00am-10:20am – Opening Remarks and Introduction: The Why and What of SIPconnect – Presenter: Marc Robins, Managing Director, SIP Forum
10:20am-10:40am -- The SIPconnect Value Proposition – Presenter: Marc Robins 
10:45am-11:15am -- SIPconnect Compliance Process Overview – Presenter: Chris Gatch, CTO, Cbeyond Topics include: 

A Discussion of the Application Process
The Compliance Survey
License Agreement Overview
Discussion of the Certification Committee
11:15am-11:45am -- Lessons Learned  – Presenter: Mark Enstrom, Broadsoft.         Get real-world feedback from actual SIP Trunking/SIPconnect deployments. Topics include: 

How to use SIPconnect in your existing interoperability program to significantly reduce resource requirements
How real-world service providers improved the economics of SIP Trunking services
NAT/Firewall Issues
The additional requirements needed to provide a successful SIPconnect Trunking service.
11:45am-12:00pm – Break 
12:00pm-12:45pm -- SIPconnect Recommendation Deep Dive – Presenter: Chris Gatch, CTO, Cbeyond. This session will provide a section-by-section review of the SIPconnect Technical Recommendation, with identification of key areas of interoperability and review of the latest SIPconnect Compliance survey results.
12:45pm-1:00pm—Closing Remarks and Q&A 
PRESENTER BIOS 
Mark Enstrom Mark Enstrom is a telecommunications professional with almost 14 years of experience in the industry, including roles in Systems Engineering, Product Line Management and Product/Field Marketing. In these roles, Mark has been instrumental in many product introductions. As a member of the Field Marketing team at BroadSoft, he works with service providers on their go-to-market efforts for Hosted PBX, Business Trunking and SIP trunking solutions.
  Before joining the Marketing team, he led the effort to define and document the SIP trunking solution, which is now BroadSoft’s fastest growing application. As BroadSoft’s representative to the SIP Forum, Mark was active with the SIPconnect Certification Committee and helped launch the SIPconnect Compliance Trademark. Prior to BroadSoft, Mark was with TTC/Acterna (now JDSU) and sentitO Networks (now Verso). Mark holds a bachelor’s degree in Electrical Engineering from MIT and an MBA from the R. H. Smith School of Business at the University of Maryland.
Chris Gatch Chris Gatch is the CTO and a founder of Cbeyond (NASDAQ: CBEY), a successful managed services provider started in 1999 and now publicly traded on the NASDAQ. Chris is responsible for Cbeyond's Engineering department as well as on-going network technology research and development. Chris has a bachelor's degree in computer engineering from Clemson University and a master's degree in the management of technology from the Georgia Institute of Technology. He has served on the board of the Cisco BTS 10200 Users Group and the Service Provider Board of the International Packet Communications Consortium (IPCC). Chris has been professionally committed to VoIP technology since 1998.
Chris helped lead the formation of the SIP Forum’s Service Provider to IP PBX Interoperability Task Group for which he submitted the proposal to create. This task group produced the SIPconnect SIP Forum recommendation of which Chris was an editor along with Chris Sibley.
Marc Robins Marc Robins currently serves as president of SIP Forum LLC (the operating U.S. subsidiary of the SIP Forum) and as the consulting managing director of the SIP Forum. Marc is also the founder and Chief Technology Evangelism Officer of Robins Consulting Group (RCG), an IP Communications industry consultancy founded in 2003 that offers advisory, market intelligence and strategy development; and comprehensive marketing and communications and services for companies in the IP communications industry. RCG has also served as a producer of a number of special workshops and industry conferences.
Prior to RCG, Marc served five years as Vice President of Publications and Trade Shows, Associate Group Publisher and Group Editorial Director at TMC (Technology Marketing Corporation).  He was the co-founder, conference architect and co-chairman of TMC conferences, such as the Internet Telephony Conference & EXPO, and Communications Solutions EXPO, and also managed the operations of a number of leading industry publications -- including Internet Telephony® magazine, Customer Inter at ction Solutions®, Communications Solutions®, and Communications ASP®. 
Over the course of his 25-year career, Marc has authored hundreds of articles, columns and special reports for leading telecommunications industry magazines. He continues to evangelize new IP communications technologies as a regular contributor to mainstream and business publications, and as the author of columns in various industry publications. Marc is also a frequent moderator and speaker at industry events and is a commentator for a variety of radio shows on the subject of VoIP and IP communications.
Best,   Marc   
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-- 
Dan York, CISSP, Director of Emerging Communication Technology
Office of the CTO    Voxeo Corporation     dyork at voxeo.com
Phone: +1-407-455-5859  Skype: danyork  http://www.voxeo.com
Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com

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