Sorry the Phone will not register with the asterisk box. I have gotten a softphone to register and make outbound calls fine.
Main Topics
Browse All Topics
Loading Advertisement...
Question
Cisco and Sip
Wow I guess cisco and SIP are a sore subject here no one is answering anything I post. Anyways I have a setup like this
comcast --- asterisk box ----- LAN ---- MY PC ---- Cisco 7971G-GE --- Qwest
I have it setup in DHCP that everything goes out through quest except my VOIP which is controlled trough the asterisk box using NAT.
my problem is either I have bad configs or I the stupid phone which no one really seems to like
MY SIP image is SIP 70.8-3-1S
here is my log I'M going to sleep been at this for days
To: <sip:2000@192.168.0.100>;t
Call-ID: 00152bd2-02640002-0fcf0e0e
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7eef41bd"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '00152bd2-02640002-0fcf0e0
elastix*CLI>
<--- SIP read from 192.168.0.12:49175 --->
REGISTER sip:192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z
From: <sip:2000@192.168.0.100>;t
To: <sip:2000@192.168.0.100>
Call-ID: 00152bd2-02640002-0fcf0e0e
Max-Forwards: 70
Date: Fri, 04 May 2007 19:55:02 GMT
CSeq: 102 REGISTER
User-Agent: Cisco-CP7971G-GE/8.3.0
Contact: <sip:7b452e87-4496-4762-e1
Authorization: Digest username="2000",realm="ast
Supported: (null),X-cisco-xsi-6.0.1
Content-Length: 0
Expires: 3600
<------------->
--- (14 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.12 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.12:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z
From: <sip:2000@192.168.0.100>;t
To: <sip:2000@192.168.0.100>
Call-ID: 00152bd2-02640002-0fcf0e0e
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
elastix*CLI>
<--- Transmitting (no NAT) to 192.168.0.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z
From: <sip:2000@192.168.0.100>;t
To: <sip:2000@192.168.0.100>;t
Call-ID: 00152bd2-02640002-0fcf0e0e
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:7b452e87-4496-4762-e1
Date: Tue, 02 Jun 2009 02:46:59 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '00152bd2-02640002-0fcf0e0
Scheduling destruction of SIP dialog '29717d5f084401263e0f09365
Reliably Transmitting (no NAT) to 192.168.0.12:5060:
NOTIFY sip:7b452e87-4496-4762-e11
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=
From: "Unknown" <sip:Unknown@192.168.0.100
To: <sip:7b452e87-4496-4762-e1
Contact: <sip:Unknown@192.168.0.100
Call-ID: 29717d5f084401263e0f09365f
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:*97@192.168.0.100
Voice-Message: 0/0 (0/0)
---
elastix*CLI>
<--- SIP read from 192.168.0.12:49183 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=
From: "Unknown" <sip:Unknown@192.168.0.100
To: <sip:7b452e87-4496-4762-e1
Call-ID: 29717d5f084401263e0f09365f
Date: Fri, 04 May 2007 19:55:10 GMT
Warning: 399 Bad MWI NOTIFY
CSeq: 102 NOTIFY
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- Got SIP response 400 "Bad Request" back from 192.168.0.12
Really destroying SIP dialog '29717d5f084401263e0f09365
Really destroying SIP dialog '00152bd2-02640002-0fcf0e0
Reliably Transmitting (NAT) to 209.249.3.59:5060:
OPTIONS sip:209.249.3.59 SIP/2.0
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch
From: "Unknown" <sip:Unknown@xxx.11.27.1xx
To: <sip:209.249.3.59>
Contact: <sip:Unknown@xxx.11.27.1xx
Call-ID: 6035040551856d385a75ebf365
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Jun 2009 02:47:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
elastix*CLI>
<--- SIP read from 209.249.3.59:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch
To: <sip:209.249.3.59>;tag=345
From: "Unknown" <sip:Unknown@xxx.11.27.1xx
Call-ID: 6035040551856d385a75ebf365
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:209.249.3.59:5060>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '6035040551856d385a75ebf36
Reliably Transmitting (NAT) to 209.249.3.59:5060:
OPTIONS sip:209.249.3.59 SIP/2.0
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch
From: "Unknown" <sip:Unknown@xxx.11.27.1xx
To: <sip:209.249.3.59>
Contact: <sip:Unknown@xxx.11.27.1xx
Call-ID: 4b05409745a3073f5830ea292d
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Jun 2009 02:48:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
elastix*CLI>
<--- SIP read from 209.249.3.59:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch
To: <sip:209.249.3.59>;tag=345
From: "Unknown" <sip:Unknown@xxx.11.27.1xx
Call-ID: 4b05409745a3073f5830ea292d
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:209.249.3.59:5060>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '4b05409745a3073f5830ea292
Reliably Transmitting (NAT) to 209.249.3.59:5060:
OPTIONS sip:209.249.3.59 SIP/2.0
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch
From: "Unknown" <sip:Unknown@xxx.11.27.1xx
To: <sip:209.249.3.59>
Contact: <sip:Unknown@xxx.11.27.1xx
Call-ID: 2ebf3ede51617e4e00dabdc90a
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Jun 2009 02:49:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
elastix*CLI>
<--- SIP read from 209.249.3.59:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.11.27.1xxx:5060;branch
To: <sip:209.249.3.59>;tag=345
From: "Unknown" <sip:Unknown@xxx.11.27.1xx
Call-ID: 2ebf3ede51617e4e00dabdc90a
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:209.249.3.59:5060>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '2ebf3ede51617e4e00dabdc90
-- Remote UNIX connection
-- Remote UNIX connection disconnected
elastix*CLI>
This Question has been solved and asker verified All Experts Exchange premium technology solutions are available to subscription members.
Subscribe now for full access to Experts Exchange and get
Instant Access to this Solution
- Plus...
- 30 Day FREE access, no risk, no obligation
- Collaborate with the world's top tech experts
- Unlimited access to our exclusive solution database
- Never be left without tech help again
Trusted by hundreds of thousands everyday for fast, accurate and reliable tech support.
- "The time we save is the biggest benefit of Experts Exchange to Warner Bros. What could take multiple guys 2 hours or more each to find is accessed in around 15 minutes on Experts Exchange." Mike Kapnisakis, Warner Bros.
- "Our team likes having a resource that is more secure than just using Google and most experts using this service really know their stuff. It's nice to look here first versus using Google." Dayna Sellner, Lockheed Martin
- "Anytime that I've been stumped with a problem, 9 out of 10 times Experts Exchange has either the accepted solution or an open discussion of the potential solution to the problem." Kenny Red, eBay Inc.
See what Experts Exchange can do for you.
Got a question?
We've got the answer.
Experts Exchange has been collecting answers to technology questions since 1996…3 million and counting! If you have a question, chances are we already have your answer.
Need individual assistance?
Our experts are ready to help.
If you can't find the exact answer you're looking for, ask our exclusive community of 50,000 experts. You’ll get a personalized answer from a trusted professional.
Want to learn from the best?
Read articles from industry experts.
Thousands of free tech tips, tricks, how-to’s and tutorials are available in our peer reviewed articles section. See for yourself how smart our experts are, no login required.
Working on a long term project?
Store your work and research.
Save solutions to your questions, answers you’ve discovered through searching plus helpful articles in your personal knowledgebase for easy future access.
Access the answers to your technology questions today.
30-day free trial. Register in 60 seconds.
What Makes Experts Exchange Unique?
Members of the expert community talk about why the experience at Experts Exchange is different than what you will find anywhere else.
Trusted by the world's most respected brands.
Faithfully serving IT professionals since 1996.
Try it out and discover for yourself.
30-day free trial. Register in 60 seconds.
Free Tech Articles
-
WARNING: 5 Reasons why you should NEVER fix a computer for free.
It is in our nature to love the puzzle. We are obsessed. The lot of us. We love puzzles. We love the challenge. We thrive on finding the answer. We hate disarray. It bothers us deep in our soul. W...
-
SCCM OSD Basic troubleshooting
SCCM 2007 OSD is a fantastic way to deploy operating systems, however, like most things SCCM issues can sometimes be difficult to resolve due to the sheer volume of logs to sift through and the dispe...
-
Migrate Small Business Server 2003 to Exchange 2010 and Windows 2008 R2
This guide is intended to provide step by step instructions on how to migrate from Small Business Server 2003 to Windows 2008 R2 with Exchange 2010. For this migration to work you will need the fo...
-
Create a Win7 Gadget
This article shows you how to create a simple "Gadget" -- a sort of mini-application supported by Windows 7 and Vista. Gadgets can be dropped anywhere on the desktop to provide instant information, ...
-
Outlook continually prompting for username and password
There have been a lot of questions recently regarding Outlook prompting for a username and password whilst using Exchange 2007. There are a few reasons why this would happen and I will try to cover t...
-
Backup Exchange 2010 Information Store using Windows Backup
There seems to be quite a lot of confusion around the ability to backup Exchange 2010 using the built in Windows Backup feature. This stems from the omission of this feature prior to Exchange 2007 s...
Cloud Class Webinars
- Avoiding Bugs in Microsoft Access
Alison Balter takes and in-depth look at avoiding bugs in Access. In this webinar you will learn about using the immediate window to debug your applications, invoking the debugger, using breakpoints to troubleshoot, stepping through code, setting the next statement to execute, ...
- Top 10 Best New Features in Visio 2010
Scott Helmers gives live demonstrations of the top 10 new features in Visio 2010. This webinar will teach you how to create compelling diagrams by adding shapes to the page with a single click, linking the shapes in a diagram to data in Excel (or SQL Server, or SharePoint), ...
- IT Consultant Business Secrets Revealed
Michael Munger, Experts Exchange tech pro and IT consultant, pulls back the curtain on his very successful businesses and answers question on every IT consultant and business owner should know about. He shares secrets on what he did to solve the 5 most common problems in IT, ...
- Disaster Recovery and Business Continuity
Quest CTO, Mike Billon, gives an overview of the steps involved in building a dunamic disaster recovery plan. Through case studies and an examination of software/hardware tooles for monitoring and testing, you'll gain a better understandin of where you are, where you want ...
- Organize Your Visio Diagrams with Containers and Lists
Scott Helmers uses cross functional flowcharts, wireframe diagrams, data graphic legends and seating charts to teach you: how to ustilize all three new structured diagram components in Visio 2010, the best practices for organizeing shapes in previous version of Visio, how to organize ...
- How to Us Objects, Properties, Events and Methods in Microsoft Access
Alison Dalter gives an in-depbth look at objects, properties, events and methods in Microsoft Access. In this webinar you will learn about using the object browser, referring to objects, working with properties and methods, working with object variables, understanding the ...
Join the Community
Give a Little. Get a Lot.
Join the community of experts here and help other tech pros by answering question in your area of expertise. You can earn FREE access to all Experts Exchange's premium features and resources.
Answers
...but your SIP trace shows that the Cisco phone is registering ok:
<--- Transmitting (no NAT) to 192.168.0.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z
From: <sip:2000@192.168.0.100>;tag
To: <sip:2000@192.168.0.100>;tag
Call-ID: 00152bd2-02640002-0fcf0e0e
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:7b452e87-4496-4762-e11
Date: Tue, 02 Jun 2009 02:46:59 GMT
Content-Length: 0
I was on the phone with broadvox and realized that my softphone could dial out from the pbx but not in. however the outside calls will route to the asterisk pbx but not into the LAN(tested by setting up voice mail which picked up). I assume its routing caused by the configuration of my network I have two network cards in the PBX the WAN interface is connected directly to the NIC via 172.12.23.143 the internal LAN is connected via 192.168.2.100. I assume I need to create a static route telling the box any VOIP calls from 172.12.23.143 need to be routed to 192.168.2.0/255.255.255.0
You maybe having a vlan issue base on my knowlegde of Cisco phones and asterisk. You question is confusing. You don't really state the nature of the problem. Are you having IP communication problem with the phone. Verify you can ping the phone on the network. Verify that you can ping the phone from you asterisk box. What is the purpose of the qwest link and the comcast link? Please provide more info so we can help.
The cisco phone uses a voice vlan when the get on the network. If you do not have a voice vlan define it may not take to anything on your network. You can configure it without the voice vlan, but that would require you to setup the switch to put both the data vlan and voice vlan on the same number. It is a slightly tricky setup.
You seem to be having more problems than you can explain. See my profile and contact me and I will help you.
I was told by one of the experts in a previous link that asterisk could be install as follows
I asterisk box two nic cards -- the first card eth0 is plugged into a switch in the lan
the second nic is plugged into the Comcast router with a static ip
I can successfully make calls out using a sip phone when I try to make calls in the get stuck at the box
and display this log
--- (10 headers 0 lines) ---
173-11-27-133-oregon*CLI>
<--- SIP read from UDP://209.249.3.59:5060 --->
INVITE sip:9713454107@172.23.217.
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Supported: timer, 100rel
To: <sip:9713454107@209.249.3.5
From: "My Name " <sip:5037234567@209.249.3.5
P-Asserted-Identity:"My Name "<sip:5037234567@64.152.60.
Call-ID: 5108895-3453053515-752092@
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 209.249.3.59:5060;branch=z
Contact: <sip:5037234567@209.249.3.5
Call-Info: <sip:209.249.3.59>;method="N
Content-Type: application/sdp
Content-Length: 249
v=0
o=NXT02 6744 21046 IN IP4 209.249.3.59
s=sip call
c=IN IP4 209.249.3.60
t=0 0
m=audio 18764 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 12 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 209.249.3.59 : 5060 (NAT)
Using INVITE request as basis request - 5108895-3453053515-752092@
Found peer 'Broadvox' for '5037234567' from 209.249.3.59:5060
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 209.249.3.60:18764
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 - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.249.3.60:18764
Looking for 9713454107 in from-trunk (domain 172.23.217.145)
list_route: hop: <sip:5037234567@209.249.3.5
<--- Transmitting (NAT) to 209.249.3.59:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.249.3.59:5060;branch=z
From: "My Name " <sip:5037234567@209.249.3.5
To: <sip:9713454107@209.249.3.5
Call-ID: 5108895-3453053515-752092@
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:9713454107@172.23.217.
Content-Length: 0
<------------>
-- Executing [9713454107@from-trunk:1] Set("SIP/Broadvox-09daedc0
-- Executing [9713454107@from-trunk:2] Gosub("SIP/Broadvox-09daed
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/Broadvox-09dae
-- Executing [s@app-blacklist-check:2] Return("SIP/Broadvox-09dae
-- Executing [9713454107@from-trunk:3] ExecIf("SIP/Broadvox-09dae
-- Executing [9713454107@from-trunk:4] Set("SIP/Broadvox-09daedc0
-- Executing [9713454107@from-trunk:5] Set("SIP/Broadvox-09daedc0
-- Executing [9713454107@from-trunk:6] Set("SIP/Broadvox-09daedc0
-- Executing [9713454107@from-trunk:7] Goto("SIP/Broadvox-09daedc
-- Goto (from-did-direct,2000,1)
-- Executing [2000@from-did-direct:1] Set("SIP/Broadvox-09daedc0
-- Executing [2000@from-did-direct:2] Macro("SIP/Broadvox-09daed
-- Executing [s@macro-exten-vm:1] Macro("SIP/Broadvox-09daed
-- Executing [s@macro-user-callerid:1] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/Broadvox-09dae
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/Broadvox-09dae
-- Executing [s@macro-user-callerid:4] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-user-callerid:5] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/Broadvox-09dae
-- Goto (macro-user-callerid,s,10)
-- Executing [s@macro-user-callerid:10]
-- Executing [s@macro-user-callerid:11]
-- Executing [s@macro-user-callerid:12]
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19]
-- Executing [s@macro-exten-vm:2] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-exten-vm:3] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-exten-vm:4] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-exten-vm:5] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-exten-vm:6] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-exten-vm:7] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-exten-vm:8] Macro("SIP/Broadvox-09daed
-- Executing [s@macro-record-enable:1] GotoIf("SIP/Broadvox-09dae
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/Broadvox-09daedc0
-- Launched AGI Script /var/lib/asterisk/agi-bin/
recordingcheck,20090603-14
-- Executing [s@macro-record-enable:5] MacroExit("SIP/Broadvox-09
-- Executing [s@macro-exten-vm:9] Macro("SIP/Broadvox-09daed
-- Executing [s@macro-dial:1] GotoIf("SIP/Broadvox-09dae
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/Broadvox-09daedc0
-- Launched AGI Script /var/lib/asterisk/agi-bin/
dialparties.agi: Failed to execute '/var/lib/asterisk/agi-bin
-- Executing [s@macro-dial:4] NoOp("SIP/Broadvox-09daedc
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/Broadvox-09dae
-- Executing [s@macro-exten-vm:11] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/Broadvox-09da
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/Broadvox-09da
-- Executing [s@macro-exten-vm:14] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-exten-vm:15] NoOp("SIP/Broadvox-09daedc
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/Broadvox-09dae
-- Executing [s@macro-exten-vm:17] NoOp("SIP/Broadvox-09daedc
-- Executing [s@macro-exten-vm:18] Macro("SIP/Broadvox-09daed
-- Executing [s@macro-vm:1] Macro("SIP/Broadvox-09daed
-- Executing [s@macro-user-callerid:1] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/Broadvox-09dae
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/Broadvox-09dae
-- Executing [s@macro-user-callerid:4] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-user-callerid:5] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/Broadvox-09dae
-- Goto (macro-user-callerid,s,10)
-- Executing [s@macro-user-callerid:10]
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19]
-- Executing [s@macro-vm:2] Set("SIP/Broadvox-09daedc0
-- Executing [s@macro-vm:3] GotoIf("SIP/Broadvox-09dae
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] GotoIf("SIP/Broadvox-09dae
-- Executing [vmx@macro-vm:2] Set("SIP/Broadvox-09daedc0
-- Executing [vmx@macro-vm:3] GotoIf("SIP/Broadvox-09dae
-- Goto (macro-vm,vmx,5)
-- Executing [vmx@macro-vm:5] NoOp("SIP/Broadvox-09daedc
-- Executing [vmx@macro-vm:6] GotoIf("SIP/Broadvox-09dae
-- Goto (macro-vm,s-,1)
-- Executing [2000@from-did-direct:3] Goto("SIP/Broadvox-09daedc
-- Goto (from-did-direct,vmret,1)
-- Executing [vmret@from-did-direct:1] GotoIf("SIP/Broadvox-09dae
-- Executing [vmret@from-did-direct:2] Hangup("SIP/Broadvox-09dae
== Spawn extension (from-did-direct, vmret, 2) exited non-zero on 'SIP/Broadvox-09daedc0'
Scheduling destruction of SIP dialog '5108895-3453053515-752092
173-11-27-133-oregon*CLI>
<--- Reliably Transmitting (NAT) to 209.249.3.59:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 209.249.3.59:5060;branch=z
From: "My Name " <sip:5037234567@209.249.3.5
To: <sip:9713454107@209.249.3.5
Call-ID: 5108895-3453053515-752092@
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
<------------>
173-11-27-133-oregon*CLI>
<--- SIP read from UDP://209.249.3.59:5060 --->
ACK sip:9713454107@172.23.217.
Max-Forwards: 70
To: <sip:9713454107@209.249.3.5
From: "My Name " <sip:5037234567@209.249.3.5
Call-ID: 5108895-3453053515-752092@
CSeq: 1 ACK
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 209.249.3.59:5060;branch=z
Contact: <sip:5037234567@209.249.3.5
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5108890-3453053514-777640
Really destroying SIP dialog '5108892-3453053514-995261
Really destroying SIP dialog '5108893-3453053515-232559
Really destroying SIP dialog '5108895-3453053515-752092
Reliably Transmitting (NAT) to 209.249.3.59:5060:
OPTIONS sip:209.249.3.59 SIP/2.0
Via: SIP/2.0/UDP 172.23.217.145:5060;branch
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.23.217.145
To: <sip:209.249.3.59>
Contact: <sip:Unknown@172.23.217.145
Call-ID: 363e13ba05cfcdce23feabbc26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.1.0
Date: Wed, 03 Jun 2009 21:33:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
---
173-11-27-133-oregon*CLI>
<--- SIP read from UDP://209.249.3.59:5060 --->
SIP/2.0 200 OK
Session-Expires: 3600
Require: timer
Via: SIP/2.0/UDP 172.23.217.145:5060;branch
To: <sip:209.249.3.59>;tag=34530
From: "Unknown" <sip:Unknown@172.23.217.145
Call-ID: 363e13ba05cfcdce23feabbc26
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:209.249.3.59:5060>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '363e13ba05cfcdce23feabbc2
Reliably Transmitting (NAT) to 209.249.3.59:5060:
OPTIONS sip:209.249.3.59 SIP/2.0
Via: SIP/2.0/UDP 172.23.217.145:5060;branch
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.23.217.145
To: <sip:209.249.3.59>
Contact: <sip:Unknown@172.23.217.145
Call-ID: 2b121cbb61d5b8bb1a2bc7491e
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.1.0
Date: Wed, 03 Jun 2009 21:33:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
---
173-11-27-133-oregon*CLI>
<--- SIP read from UDP://209.249.3.59:5060 --->
SIP/2.0 200 OK
Session-Expires: 3600
Require: timer
Via: SIP/2.0/UDP 172.23.217.145:5060;branch
To: <sip:209.249.3.59>;tag=34530
From: "Unknown" <sip:Unknown@172.23.217.145
Call-ID: 2b121cbb61d5b8bb1a2bc7491e
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:209.249.3.59:5060>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '2b121cbb61d5b8bb1a2bc7491
173-11-27-133-oregon*CLI>
As I said before, it appears that the phone is registering ok based on the SIP messages. Is the display on the phone showing not registered? What does Asterisk show for that phone if you type this command at the CLI:
sip show peers
Please can you also explain about your LAN because you have mentioned two different private subnets - 192.168.0.x and 192.168.2.x - but not explained which one is correct (or if you have both, then which one is Asterisk using and which one is the phone using and how are they linked)? Also, the questions asked by koszeqi are relevant and you have not answered them. We cannot help if you don't provide information when requested by experts.
It is showing on the phone as registering I have given up on the SIP for the phone and am testing using the SCCP protocol can you point me in a good direction of how to set it up. I have alreadt instralled the drivers but am still get the registering message for SCCP. If this doesn't work I am going to find a more friendly phone to work with and return this one
that is awhole different ball game. You want to enable and configure the Skinny protocol on your Asterisk box.
checkout http://www.voip-info.org and do a search for "configuring skinny on asterisk" and "setup sccp phone on asterisk"
Good luck.
Here's how to configure skinny "sccp" on asterisk box.
http://www.voip-info.org/w
Have fun.
3 Ways to Join
Business Accounts
- Perfect for organizations
- Multiple users
- Save over 36%
- Create a Business Account
Answer for Membership
- Earn free access
- Showcase your knowledge
- No credit card required
- Sign Up to Answer
Loading Advertisement...
by: feptiasPosted on 2009-06-02 at 06:36:34ID: 24526572
> "Wow I guess cisco and SIP are a sore subject here no one is answering anything I post."
Perhaps the problem is that people like me who know Asterisk, tend not to know Cisco. Those who know Cisco, probably don't know Asterisk. They are products on opposite ends of the cost spectrum and reflecting totally different commercial philosophies.
As for your question, you don't actually say what the problem is. The SIP trace shows:
1. A registration that is successful
2. Asterisk sending a MWI NOTIFY that is rejected with "400 Bad Request"/"399 Bad MWI NOTIFY"
3. Some OPTIONS requests being sent to the Internet and getting 200 OK returned
Are we to assume that the problem is the MWI notification? Perhaps it is because the Cisco 7971G does not recognise the message account: "Message-Account: sip:*97@192.168.0.100".