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Grandstream Phone Says Not Registered

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Jag5x5 jag5x5 Newsterisk Posts: 33Joined: Thu Dec 21, 2006 10:56 pm Top by heartofyogesh » Tue Sep 05, 2006 12:08 am Yes it worked heartofyogesh Newsterisk Posts: 2Joined: Thu Need Help? There are several ways to determine whether or not the account has successfully registered to the SIP server. 1. The client MAY repeat the request without modifications at any later time. [1]:§21.4.9 409 Conflict User already registered.[9]:§7.4.10 Deprecated by omission from later RFCs[1] and by non-registration with the IANA.[3] 410 Check This Out

I have double checked all of the user ID's to make sure that they match between the CM and the sip phone. I have attached some screen shots of the phones configuration tabs. (SIP and the EXT tabs). Integration of Resource Management and Session Initiation Protocol (SIP). more stack exchange communities company blog Stack Exchange Inbox Reputation and Badges sign up log in tour help Tour Start here for a quick overview of the site Help Center Detailed

Grandstream Phone Says Not Registered

If it is needed, please enter the outbound proxy IP address or domain name. IETF. With VoIP service you can call anywhere at anytime for less money.

One of the 2 machines acts as an Asterisk server (under CentOS 4.2) the other is just a Windows XP Home box. It shows as "online" when the PBX is connected to the Netgear switch. A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP). Grandstream Ip Phone Configuration IETF.

Figure:GXV3140 registration (Please enter your corresponding login information) If the GXV3140 and the SIP server are on the same LAN, NAT traversal is not needed. Grandstream Not Registered Problem Unlike the 486 response, this response indicates the destination knows there are no alternative destinations (such as a voicemail server) able to accept the call.[1]:§21.6.1 603 Decline The destination does not i think it isnt a nat problem because i have asterisk server and the Xlite client in the same machine best regards nabil Newsterisk Posts: 3Joined: Wed Nov 29, 2006 If you are experiencing a similar issue, please ask a related question Suggested Solutions Title # Comments Views Activity Using an ASA5505 as a multicast router 2 56 168d failover for

https://tools.ietf.org/html/rfc6442#section-4.3. ^ a b c d Peterson, Jon; Jennings, Cullen (August 2006). Grandstream Gxp1405 Sip Registration Assigned it basic third part SIP protocol. In how many bits do I fit Why call it a "major" revision if the suggested changes are seemingly minor? If the login information is wrong, the SIP server will reject the registration request of the phone.

If all of these are correct, there may be a problem with NAT

Grandstream Not Registered Problem

Reboot your device and check if you have dial tone. 4.) Make sure again you entered your account number, password (passwords are case sensitive) and our servers domain name correctly. http://www.voipmechanic.com/sip-call-example.htm https://tools.ietf.org/html/rfc3265. ^ Fielding, Roy T.; Gettys, James; Mogul, Jeffrey C.; Nielsen, Henrik Frystyk; Masinter, Larry; Leach, Paul; Berners-Lee, Tim (June 1999). "202 Accepted". Grandstream Phone Says Not Registered I have also assigned a DN to the phone. Grandstream Device Not Registered This response is issued by UASs and registrars.[1]:§21.4.2 402 Payment Required Reserved for future use.[1]:§21.4.3 403 Forbidden The server understood the request, but is refusing to fulfil it.[1]:§21.4.4 Sometimes (but not

The way we have configured the accounts in the SIP channel driver, Asterisk will expect the phones to register to it. his comment is here You can use your existing high-speed Internet connection at home and your existing touch-tone phone to make all your calls at a fraction of the cost of analog phone service with here's my username and password. Why does the GXV3140 fail to register to the SIP server? Grandstream Sip Registration

A server may send multiple 182 responses to update progress of the queue.[1]:§21.1.4 183 Session in Progress This response may be used to send extra information for a call which is Please make changes one at a time and reboot your device each time to see if the problem is solved. 1.) Change the SIP registration port on VoIP device to 5060 as SIP client is getting registered, I am able to get the voip-to-landline context in extensions.conf by dialing an IP address... this contact form If the GXV3140 is on a LAN and needs to register to a SIP server on a public IP, NAT traversal must be enabled.

Not the answer you're looking for? Grandstream Device Configuration Default Password Here is a guide: https://supportforums.cisco.com/docs/DOC-11599 Hope it helps. 0 Message Author Comment by:TNRD ID: 369893102011-10-18 Hi, I tried running the RTMT as per the document guide. MenuExperts Exchange Browse BackBrowse Topics Open Questions Open Projects Solutions Members Articles Videos Courses Contribute Products BackProducts Gigs Live Courses Vendor Services Groups Careers Store Headlines Website Testing Ask a Question

You should ask a local voip person to have a look, difficult remotely ^^ –Olivier Dulac Jan 8 '13 at 16:25 I've looked in the PBX config, and the

SIP Tutorial For a good understanding of VoIP SIP messaging. If the NAT type is unknown, it is recommended to select “Auto”. The issue was that phone was registering and then unregistering. Grandstream Nat Traversal Session Initiation Protocol (SIP) Extension for Event State Publication.

The Session Initiation Protocol (SIP) Referred-By Mechanism. Press Save and restart the phone. https://tools.ietf.org/html/rfc5360#section-5.9.2. ^ a b Arkko, Jari; Torvinen, Vesa; Camarillo, Gonzalo; Niemi, Aki; Haukka, Tao (January 2003). navigate here Before registering, please make sure that you have valid login information with a valid SIP server.

Each transaction consists of a SIP request (which will be one of several request methods), and at least one response.[1]:p11 SIP requests and responses may be generated by any SIP user Before trying to resolve this issue on your router, try the following solutions on your VoIP device configuration. https://tools.ietf.org/html/rfc6228#section-Abstract. ^ a b Roach, Adam B. (June 2002). https://tools.ietf.org/html/rfc4028#section-6. ^ Polk, James; Rosen, Brian; Peterson, Jon (December 2011). "424 (Bad Location Information) Response Code".

Session Timers in the Session Initiation Protocol (SIP). However, upon softphone's startup, after a brief attempt to register, it shows a message on the display: -SIP not registered. This should be set to demo-alice on one phone and demo-bob on the other. VoIP Devices Support If you get one-way audio, or cannot get a dial tone or cannot make/receive calls with your VoIP device, you are probably behind firewall.

You can register the GXV3140 to a SIP server through the web interface or the LCD menu. Cisco Support Community Directory Network Infrastructure WAN, Routing and Switching LAN, Switching and Routing Network Management Remote Access Optical Networking Getting Started with LANs IPv6 Integration and Transition EEM Scripting Other dhcp switch voip share|improve this question edited Jan 8 '13 at 2:16 asked Jan 7 '13 at 21:34 Chase Florell 37511027 Well the port config screenshot shows interface g1 I do have sufficient licenses.

Privacy policy About Wikipedia Disclaimers Contact Wikipedia Developers Cookie statement Mobile view Asterisk Forums Please hold while I try that extension. External links[edit] Mapping SIP Error Messages to DSS1 codes Session Initiation Protocol (SIP) Parameters Contains a registry of different SIP parameters, including response codes Retrieved from "https://en.wikipedia.org/w/index.php?title=List_of_SIP_response_codes&oldid=750680183" Categories: VoIP protocolsInternet-related listsTelecommunications How to approach cyber security in today's business world! Is there a toy example of an axiomatically defined system/ structure?

How smart is the original Ridley Scott Xenomorph really? External link in |work= (help) ^ Holmberg, Christer (May 2011). To register the phone through the phone LCD screen, select [MENU]→[Settings]→[Accounts] →[Account], and enter the required information (similar to the web interface). But this changes depending on your phone types (and on their config, usually on the PBX server that send them their config, if they can connect to it).

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