6300i and VoIP/SIP connections

I' v just bought 6300i and try to configure connection to my VoIP/SIP provider and was surprised that a can only add providers listed by NOKIA.
Can anyone know how to add account to VoIP/SIP provider not listed on NOKIA web page (display when you try to add new account)?
Solved!
Go to Solution.

Yes there is a method to add any VoIP provider to 6300i. Please read this thread for more information,
http://discussion.forum.nokia.com/forum/showthread.php?t=136775

Similar Messages

CME 7.1 with SCCP 7940G phones and SIP connection to a VOIP provider - inbound outbound fails

Here's a quick and dirty diagram of a CME 7.1 configuration. The phone can all call each other but something is not quite right with the SIP provider. The registrar and SIP registration pieces are working but most of the configuration examples that I've seen make me think that the CME router was being used as the edge device to the internet. From my drawing, you can see that is not the case here. My edge device is a Cisco ASA5505 with 9.2.x software running. I might be missing something in the SIP gateway knowledge department. Without diving into the configuration, I'm wondering if SIP messages are failing for calls because of NAT'ing? Trying to do searches has been tricky because I keep running into information that is more about setting up CME for SIP phones or just getting SIP to work between CME and a SIP provider. I have that part working. I'm just a bit unsure about how an SCCP 7940G gets an outbound call or even gets one to come in.
When I dial from my cell phone to the pilot number, there are no rings, it just goes to the VOIP provider's voice mail. When I try to dial out, I get a fast busy.
So, is NAT a consideration? Will the SIP gateway set up a call (forward) via the pre-established SIP connection? Yeah, I do sound like a newb.
If anyone has good information about, let's say, an inbound call and how that traffic flow works.
Thanks!

Have you configured your ASA to either NAT the IP address of the CME router or to do port forwarding for port 5060?

X5800 how to use SIP and VOIP

Hello, I have a X5800 music, model 5800d-1 and I can enter all the necessary SIP data. I get the message (settings, connectivity, admin settings, sip settings) registered which is fine. My provider is freephonie. Question: 1) Although on the freebox the SIP is activated, I don't receive incoming call on my nokia. 2) How do I tell my nokia to call on the internet ? Is it possible only using SIP tel numbers which I can enter in the contacts list? Help much appreciated. Pierre.

SIP and VoIP are not synonymous. VoIP can indeed use SIP (and often does), but other services, such as video sharing for example, also use SIP.
It so happens that the 5800 has no native support for VoIP, which is why you can't make SIP calls using it.
Your only solution for SIP is to use third-party services such as fring, for example. Once logged into your fring account you can subscribe to a SIP service. Fring's systems will register to the SIP account and relay the comminucation through your fring account. There's lag and the audio quality isn't as good as a direct SIP connection but at least it works.
The alternative is to use a phone that supports VoIP/SIP natively.
Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you!

Trying to use an SPA3102 to connect a SIP and non-SIP Telephone System

I have a new Mitel 3300 system which is working very well, but for 12 months or so ( until I can replace it with another MITEL ) I need to connect it to an old INDEX system.
I have got an SPA3102, and for the moment it is connected to an extension on the INDEX and a phone, which I will be replacing with an analogue trunk once testing is complete.
The SIP portion is giving me a huge headache. The MITEL side is nice and easy, poking the right range of extensions to a SIP gateway won't take long
The SPA3102 is a real pain, the Manual I have for the SPA just seems to list ranges of values and does not seem to show how to configure it to talk to my MITEL, I can't work out how to get it to make SIP calls across the ethernet interface at all.
Can anyone help, or perhaps provide a decent guide on how to do this.

Headbolt wrote:
This is what I want to achieve.
From the MITEL, when you dial a 4 digit extension 47xx the MITEL will make a SIP call to the SPA3102, which will then in turn use the port that is plugged into an INDEX extension to make an internal call on the INDEX to extension 47xx.
You will be using the pstn tab and the voip-to-pstn gateway on the SPA3102. There are two different ways you can configure the SPA3102 for the voip-to-pstn gateway.
One way is with no authentication and the Mitel will send a call to the sip uri represented by the userid@ipaddress: port_number and the SPA3102 will return a dial tone to the caller and the caller will then enter the INDEX extension they wish to dial.
The other way is with http_digest authentication where the Mitel system will send a Sip Invite to the SPA at the extnumber@ipaddress: port number with the INDEX extension to be dialed and will respond with the authid and the password at the appropriate time.
Without a personal understanding the capabilities of the MITEL system to send and receive sip calls I can't give more specific instructions.
From the INDEX when we dial a 4 digit 65xx, the INDEX will send the call out an Analogue trunk that is connected to the SPA3102 which will then make a SIP call to the MITEL which will ring it's extension 65xx
The forwarding of the call from the INDEX system, after it is answered by the pstn-to-voip gateway of the SPA3102 is done thru the appropriate dial plan of the 3102. If the INDEX system is sending the dial digits after the 3102 pstn-to-voip gateway answers the call then the call will be sent to the userid and ip_address in the proxy/userid on the spa3102's pstn line tab.
Without a personal understanding of the capabilities of the MITEL system to send and receive sip calls I can't give more specific instructions.

CUCM and Avaya CS1000 SIP connection

                   Hello - looking for some help on a SIP trunk configuration between the 2 devices. Currently we are running CUCM 9.1 with Avaya Session Manager 6.3. We are having issues with the call completing from the CS 1000 to the CUCM. Below are the session traces from both the Session Manager and CUCM. The CS1000 currently has 4 digit extensions and the CUCM has 7 digit extensions. We translate the number in the Session Manager to send the 7 digits. If you could lead in the right direction I would appreciate it. WOuld this have something to do with the context coming out of Session Manager? It looks like CDP.UDP and then only the 4 digits and the CUCM needs the 7. I also attached the configurations guide used for this.
Session Manager trace:
   mil-ss-01                 CUCM 9
                       SM100                10.101.2.75             10.174.2.75
13:04:53.730 |<--OPTIONS-|           |           |           |           | (1) sip:10.5.1.30
13:04:53.731 |--200 OK-->|           |           |           |           | (1) 200 OK (OPTIONS)
13:04:53,734 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:53,734 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:56.183 |--INVITE-->|           |           |           |           | (2) T:7317;phone-context=cdp.udp F:anonymous@anonymous U:7317;phone-context=cdp.udp
13:04:56.184 |<--Trying--|           |           |           |           | (2) 100 Trying
13:04:56,185 |                  Remote host is trusted                   | Trusted
13:04:56,185 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:56,186 |                Applied ingress Adaptation                 | P-Asserted-Identity=<sip:[email protected]>, Request-URI=sip:[email protected], History-Info=<sip:[email protected]>;index=1, <sip:[email protected]>;index=1.1
13:04:56,186 |                Originating Location found                 | Location: mil-cs1000m-01
13:04:56,186 |        Try routing to determine if emergency call         | Location: mil-cs1000m-01
13:04:56,186 |                Request Dial Pattern route                 | for: sip:[email protected] Location: mil-cs1000m-01
13:04:56,186 |               Dial Pattern route parameters               | URI Domain: company.com Location: mil-cs1000m-01
13:04:56,186 |                 Trying Dial Pattern route                 | Domain: company.com Location: mil-cs1000m-01
13:04:56,186 |                    Dial Pattern found                     | for: 7317 Pattern: 7317
13:04:56,186 |                    Route Policy found                     | Pattern: 7317 RoutePolicyList: to_CUCM9
13:04:56,187 |                        Route found                        | for: sip:[email protected] SIPEntity: CUCM 9
13:04:56,187 |                     Entity Link found                     | SIPEntity: CUCM 9 EntityLink: mil-sessionmgr-01->TCP, biDirId=null, deny=false:5060
13:04:56,187 |                    Request Adaptation                     | Adapter: CUCM 9
13:04:56,188 |                 Applied egress Adaptation                 | NoAdaptationModuleExists=true, Request-URI=sip:[email protected];routeinfo=0-0, Remote-Party-ID=<sip:[email protected]>;party=calling;screen=no;privacy=off,
13:04:56,188 |                    Routing SIP request                    | SipEntity: CUCM 9 EntityLink: mil-sessionmgr-01->TCP:5060
13:04:56,189 |              No hostname resolution required              | Routing to: sip:10.5.131.12;transport=tcp;lr;phase=terminating
13:04:56.191 |           |--INVITE-->|           |           |           | (2) T:7317;phone-context=cdp.udp F:anonymous@anonymous U:7657317 P:terminating
13:04:56.196 |           |<--Trying--|           |           |           | (2) 100 Trying
13:04:56.198 |           |<--Not Fou-|           |           |           | (2) 404 Not Found
13:04:56.199 |           |----ACK--->|           |           |           | (2) sip:[email protected]
13:04:56,200 |                    Request Adaptation                     | Adapter: CUCM 9
13:04:56,201 |                    Request Adaptation                     | Adapter: CUCM 9
13:04:56,201 |                    Request Adaptation                     | Adapter: mil-ss-01
13:04:56.202 |<--Not Fou-|           |           |           |           | (2) 404 Not Found
13:04:56.203 |----ACK--->|           |           |           |           | (2) sip:7317
13:05:07,597 |                Remote host is not trusted                 | Host not trusted
13:05:07,597 |                Originating Location found                 | Location: mil-cs1000m-01
13:05:12.657 |           |<--------OPTIONS-------|           |           | (3) sip:10.5.2.51
13:05:12,659 |                Remote host is not trusted                 | Host not trusted
13:05:12,659 |                Originating Location found                 | Location: mil-cs1000m-01
13:05:12.660 |           |--------200 OK-------->|           |           | (3) 200 OK (OPTIONS)
13:05:16.877 |           |<--------------OPTIONS-------------|           | (4) sip:10.5.2.51
13:05:16,879 |                  Remote host is trusted                   | Trusted
13:05:16,879 |                    Request Adaptation                     | Adapter: mil-ss-01
13:05:16,879 |                Applied ingress Adaptation                 | P-Asserted-Identity=<sip:[email protected]>
13:05:16,879 |                Originating Location found                 | Location: sas-cs1000e-01
13:05:16.880 |           |--------------200 OK-------------->|           | (4) 200 OK (OPTIONS)
13:05:24.463 |           |<--------------------OPTIONS-------------------| (5) sip:10.5.2.51
13:05:24,465 |                Remote host is not trusted                 | Host not trusted
13:05:24,465 |                Originating Location found                 | Location: mil-cs1000m-01
13:05:24.466 |           |--------------------200 OK-------------------->| (5) 200 OK (OPTIONS)
CUCM Trace Invite:
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 3188 bytes:
[1179,NET]
INVITE sip:[email protected] SIP/2.0
P-AV-Message-Id: 1_1
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
History-Info: <sip:[email protected]>;index=1, <sip:[email protected]>;index=1.1
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=no;privacy=off
Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, PRACK, OPTIONS, INFO, SUBSCRIBE, UPDATE
Contact: <sip:00000000;[email protected]:5060;maddr=10.5.1.30;transport=tcp;user=phone;gsid=68cac530-5d21-11e3-8b45-78e3b505dc88>
Alert-Info: <cid:[email protected]>
Supported: 100rel, x-nortel-sipvc, replaces
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
Via: SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140
Via: SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
Record-Route: <sip:[email protected];transport=tcp;lr>
Record-Route: <sip:10.5.2.50:15060;transport=tcp;ibmsid=local.1372169047609_2400497_2400521;lr>
Record-Route: <sip:[email protected];transport=tcp;lr>
P-Charging-Vector: icid-value="68cac530-5d21-11e3-8b45-78e3b505dc88"
User-Agent: Nortel CS1000 SIP GW release_7.0 version_linux-6.50.00 AVAYA-SM-6.3.1.0.631004
P-Asserted-Identity: <sip:[email protected]>
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
Max-Forwards: 66
CSeq: 1 INVITE
Content-Type: multipart/mixed;boundary=unique-boundary-1
Content-Length: 1063
Av-Global-Session-ID: 68cac530-5d21-11e3-8b45-78e3b505dc88
P-Location: SM;origlocname="mil-cs1000m-01";origsiglocname="mil-cs1000m-01";origmedialocname="mil-cs1000m-01";termlocname="Cisco BE6K";termsiglocname="Cisco BE6K";smaccounting="true"
--unique-boundary-1
Content-Type: application/sdp
SDP Message
====================================================
v=0
o=- 746 1 IN IP4 10.5.1.30
s=-
c=IN IP4 10.5.1.36
t=0 0
m=audio 5234 RTP/AVP 18 0 8 101 111
c=IN IP4 10.5.1.36
a=tcap:1 RTP/SAVP
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=pcfg:1 t=1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=ptime:20
a=sendrecv
--unique-boundary-1
Content-Type: application/x-nt-mcdn-frag-hex;version=linux-6.50.00;base=x2611
Content-Disposition: signal;handling=optional
0500bc05
0107130081900000a200
09090f00e9a4830001004000
1315070011fa0f00a10d02010102020100cc040000c56000
1e0403008183
4a1c0100180001001a011404000067353505000004000000000048710000
--unique-boundary-1
Content-Type: application/x-nt-epid-frag-hex;version=linux-6.50.00;base=x2611
Content-Disposition: signal;handling=optional
011201
3c:4a:92:f4:84:f4
--unique-boundary-1--
CUCM Trying Message:
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1180,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Content-Length: 0
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1181,NET]
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 623 bytes:
[1182,NET]
ACK sip:[email protected] SIP/2.0
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1180,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Content-Length: 0
CUCM not found message:
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.5.2.51 on port 46314 index 6
[1181,NET]
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3,SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974,SIP/2.0/TCP 10.5.2.50:15060;rport;ibmsid=local.1372169047609_2400497_2400521;branch=z9hG4bK129750805377494,SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb-AP;ft=19;received=10.5.2.51;rport=19140,SIP/2.0/TCP 10.5.1.30:5060;branch=z9hG4bK-3fa2ea5-894636b6-7d11effb
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Date: Wed, 04 Dec 2013 19:12:41 GMT
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
CSeq: 1 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0
CUCM ACK message:
SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.5.2.51 on port 46314 index 6 with 623 bytes:
[1182,NET]
ACK sip:[email protected] SIP/2.0
Route: <sip:10.5.131.12;transport=tcp;lr;phase=terminating>
Call-ID: abef5b48-1e01050a-13c4-55013-3fa2ea5-5d08551f-3fa2ea5
From: <sip:[email protected];user=phone>;tag=ac493fe8-1e01050a-13c4-55013-3fa2ea5-2b7d5ad-3fa2ea5
To: <sip:7317;[email protected];user=phone>;tag=580~35abe322-3212-479c-8a2c-ab75ead590d5-21575914
Via: SIP/2.0/TCP 10.5.2.51;branch=z9hG4bK138498682095974-AP;ft=3
Via: SIP/2.0/TCP 10.5.2.50:15060;rport=30522;ibmsid=local.1372169047609_2400498_2400522;branch=z9hG4bK138498682095974
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0
Thanks.

This document worked for us between CUCM BE6000 ver 9.0 and the Avaya.
The main focus on the Cisco side is this: Page 37 - 41
5.4. Define SIP Trunk Security Profile
Expand System  Security Profile and select SIP Trunk Security Profile. Click to
configure a SIP Trunk Security Profile.
Enter the following values and use defaults for remaining fields:
 Name Enter name
 Description Enter a brief description
 Incoming Transport Type Verify “TCP+UDP” is selected
 Outgoing Transport Type Verify “TCP” is selected
 Accept Out-of-Dialog REFER Enter
 Accept Unsolicited Notification Enter
 Accept Replaces Header Enter
Click . The screen below shows SIP Trunk Security Profile for the sample configuration
5.5. Define SIP Profile
Expand Device  Device Settings and select SIP Profile. Click to configure a SIP
Profile.
Under SIP Profile Information section, enter the following values and use defaults for
remaining fields:
 Name Enter name
 Description Enter a brief description
 Default MTP Telephony Event Payload Type Enter “120”
 Disable Early Media on 180 Enter
Note: Disabling Early Media allows local ringback to be used.
Under Parameters used in Phone section, scroll to end of section and enter the following values
and use defaults for remaining fields:
 RFC 2543 Hold Enter
Click . The screen below shows SIP Profile for the sample configuration.

Home Hub 3 Security exposure allowing VOIP (SIP) t...

OK its a bit tech but BT don't seem interested but it worries me!
I have a HH3 configured with with Firewall set to Default (to block unsolicited incomming traffic), DMZ disabled (so no default routing of inbound connections) and UnNP off (So no dynamic port opening stuff). Also, the internal DHCP server has been disabled and the internet network is not on the default IP range.
This configuration should block ALL inbound traffic. I.E. traffic that originates from the public Interenet. Packets that are replies to data sent out, will be allowed through.
I have an internal VOIP server running Asterisk which connects to SIPGate on the Internet and SIP phones in the house.
My Asterisk server is logging inbound SIP connections that have their source IP address as the Public IP address of the HH3. Somehow, inbound SIP connections are getting through the HH3 and then hitting my Asterisk server. The SIP connection then attempts to call an number in Israel (00972592653787) a few times (with different call prefixes). The Asterisk server is correctly configured to drop these calls. BUT it should never get them in the first place!
So the question is, why is the HH3 allowing these connections through in the first place?
And what else might it be letting in that I haven't spotted?
PS - Latest firmware running 4.7.5.1.83.8.94.1.11 (Type A) 29/12/12
Anybody seen this or know the reason?
I'll guess not, so just say hello so I know you've seen the post :-)
Richard.

Part of a setup such as Asterix/SIPGate is the use of a STUN server. The purpose of STUN is to enable incoming connections without the need for configuration of firewalls etc. It is needed if you are to get any incoming calls, whether malicious or not. It is only when a call arrives at Asterix that a decision can be made as to whether it is acceptable. If you really don't need to accept any incoming calls to the server, you could remove the STUN configuration.
http://en.wikipedia.org/wiki/STUN

C3-01 WiFi / WLAN & VoIP / SIP Question

Hello,
I am very interested in purchasing a C3-01, as I don't want a smart phone, but I like the advanced features this phone offers. Specifically, I want to make use of the VoIP SIP client that comes built in.
My question is this: I will be using VoIP / SIP on the WLAN (Wi-Fi) connection. In another post on this forum, I saw a user saying that Wi-Fi gets turned off automatically when the phone is locked.
So, does this mean that if I've got a VoIP account set up, and the C3-01 locks (either manually or automatically), that the WLAN will drop out and I won't be able to get incoming VoIP calls?
If this is how it works, this is a pretty major oversight, and I will look for something else.
Thanks
Solved!
Go to Solution.

Yeah, Sadly Some Country still have the v05.68 as the latest firware, But the reason might be that the Firmware released in your Country may contain the fixes included in the Latest Firware for Our Country.
So thats ok, If Nokia Finds any fixes to be done then, they will release a new Firware Update for your Country. Be Rest Assured.
And Buy C3-01.
If I've helped in any way, a click upon the White star to the left would always be appreciated.
If however my answer also solves your problem clicking below " ACCEPT AS SOLUTION " it will benefit other users!

Nokai E75 VOIP/SIP Setup for WLAN: sipgate.de - ...

Had problems setting up the VOIP Client on my E75 for the SIP Provider sipgate.de: Test calls to their 10005 or 10000 number worked without problems, but for outgoing calls to normal landline numbers no audio stream was connected through (ringback was ok):
Setup:
   - sipgate.de configured via MENU -> CTRL PANEL -> NET SETTINGS -> Download
   - SIP Profile (user/password etc. )configured based on customized config from sipgate.com
   - WLAN (linksys or AVM-Fritzbox)
   - Provider Tcom/Avego or 1&1
After trying lot of different things & settings it finally turned out that deleting the NAT FW settings fixed the issue:
   - MENU -> CTRL PANEL -> NET SETTINGS -> ADVANCED VOIP SETTINGS
   - NAT FIREWALL SETTINGS -> DOMAIN PARAMETERS
       -> select sipgate.de
       -> OPTIONS -> DELETE PARAMETERS
if you don't have the "Advanced VOIP Settings" Menu, you need to load a special sip addon application. Documentation and application is available at
    forum.nokia.com
Search for "SIP VOIP SETTINGS"
If the above tweak also works for other WLAN Router or even UMTS Settings, I don't know.
With regards to sipgate.de --> havn't used it yet a lot, but the good thing is it's not a standalone client, but as it's reusing the Nokia VOIP Client it nicely integrates into the phone (same contact list, button to choose internet vs. mobile dialing, ..
hope the above saves some people some time
jo

Native VoIP/SIP is the main reason I was so pleased to switch to the E72 (I use voiptalk.org myself).
Until just over a year ago I had an N95, which also had built-in VoIP. I lost that feature when I "upgraded" to an N96 and then a 5800XM, but am pleased as Punch to have it again with the E72. It makes life so much easier having the same device acting as both mobile and "landline" (my entire phone setup at home is VoIP-based with Asterisk).
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VOIP: SIP VoIP 3.1 Nokia Suite begins install, E71...

I have a Nokia E71 and would like to begin using our wifi to make calls, (our carrier signal is week/dropped calls/connection on-hold). When I try to install the file, SIP_VoIP_3_1_Settings_Symbian_3_v1_0_en, Nokia Suite begins the installation, says to continue on phone, phone pauses and then a red screen appears, "File Corrupted".
Can you help?

@oneofmarysboys
You are trying to install a version for Symbian S^3 devices, click on the blue upturned arrow head to reveal the other available versions of which you require SIP_VoIP_3x_Settings_v2.0_en.sis (164 kb).
Once installed you may like to look at this resource:http://mywebexperience.com/voip/sip-call-settings-for-e71-voip-calls-through-your-nokia-mobile/
Happy to have helped forum with a Support Ratio = 42.5

TOIP/VOIP SIP on Nokia X, X+, XL ?

This is an usual feature on Nokia devices that have wifi and/or 3G, I mean even some basic S40 phones, except Microsft Windows Phone OS who needs a software but has unsupported STUN features that cannot be relayed through OS Notification System.
So basically on Windows Phone (including 8.1) you can't receive a call and your device would be notified in the background, would activated and app and would ring as a normal call. This is also true about an app run in the background, since it's frozen there... so this is a real pain.
There is no such problem on any foster Symbian S60/S^3 device, neither on most and even recent S40 and Asha phones.
This feature would be very usefull to receive a VOIP SIP (you know - the globall IP standard) call from any wifi connection and some time even over 3G (or 4G if the device has the capability).
Now, Android has an inner implementation of TOIP/VOIP SIP agregated to the standard phone calls (this is ususlly found in "Internet calls accounts" option, just like on Symbian devices. The Notification system isn't that great, needs maintained full connection (so sucks the battery) but however much better than the non existing Windows Phone feature. Not all the Android Smartphones have this standard feature implemented, but there are applications with full SIP support (not those only configuring the inner Android feature if present).
Where does the Nokia pkanteform 1.0 stand. I mean Nokia X, X+, XL.
The VOIP SIP is a most wanted feature, not only professional but quite common since the developpement of SIP phone lines including on ADSL home accounts.
Can I buy a Nokia XL (because of the 5Mpix CAM) for that feature or do I need to keep my Nokia C7 + Nokia E65 for both my personnal and office mobile lines, + my VOIP personnal line + my professional VOIP line ?
Solved!
Go to Solution.

This is VERY GOOD news.
If CSIPSimple works, it means Android VOIP/TOIP API through SIP protocole is preserved and supported into Nokia Android OS.
CSIPSimple is only some kind of interface to that protocole.
On the contrary Linphone is a full application with full inner protocole support, which is bad because not connected to the OS.
Good to know Nokia has it when many recent Android devices lacks it (like Motorola Moto G with Android Kitkat 4.4 made by Google - sold only after that to Lenovo).
Can you tell if the G729 is present ?

Netgear WNDR3700, Tp-Link WPS510U, Epson Photo 1400 and PowerRIPX: wireless connection

Hi All
I need some advice regarding a wireless connection to my inkjet printer.
I have recently replaced my router with a Natgear WNDR3700 dual band device (2.4G and 5G). Connected and working wirelessly are my MacBook Pro, iPhone, Zeppelin Air, smart TV. Connected and working wired are my satellite modem, VoIP ’phones and laser printer.
However, I have an Epson Photo 1400 with a Tp-Link WPS510U which isn’t working. It worked with the old router.
Netgear’s Genie application shows the printer to be available under AirPrint.
The printer driver I use is PowerRIPX PostScript.
The TP-Link has the correct IP address from the previous configuration (192.168.1.10).
Printing shows the document processing in the print queue, but there is no flashing green LED on the front of the printer and no output results.
So … how do I actually get this to work? Advice gratefully received.
Cheers
Paul

VoIP (SIP) configuration guide for Nokia 5800xm

hi.everybody i would like to ask how to configure this
VoIP (SIP) configuration for my n5800, because when i saw and follow this link http://www.elisanet.fi/craig/sipvoip/nokia_n97.html ,i only configure my SIP setting and not my net setting which i tried to find in my phone. can someone help me find this Net setting in my handset and can you put a link on how to call from phone to phone using this "VoIP" and is this free of charge. thank..more power

Hi, don't have an answer to your question, but Skype for S60.5 was announced today, works fine, will give Skype to Skype calls free, and low cost calls to non Skype numbers, but both Skype or Voip will incur data charges if you don;t have a data plan !
Good Luck
Skype.com/m with your phones browser will take you to Skype
If I have helped at all, a click on the White Star is always appreciated :
you can also help others by marking 'accept as solution'

Doubt about integration CUCM and VoIP Provider

Hi Guys,
I have the follow doubt: Is possible to do integration CUCM and VoIP Provider using authentication? Is there necessary another equipment to do this? CUBE for example? or another alternative
Thanks,
Wilson

If you are referring to authentication over sip trunk, then NO! CUCM doesn't do authentication you will need a CUBE for that

Nokia 5800 VOIP SIP client that works!

I have just installed the V Phone VoIP SIP client onto my UK Nokia 5800, and it works very well, "straight out of the box". I have it set up to use my sipgate.co.uk account, and it registers via WiFi immediately, with good clear call quality. Sometimes there is a bit of echo, but generally the call is clean.
So here's a Nokia 5800 SIP VoIP application that does work - I thought that such a thing didn't exist. V Phone are an Australian outfit, found here: http://www.thevphone.biz/Products.html. I bought their Premium version for about £5.50 equivalent of Australian dollars, so it isn't expensive either.
Recommended (and I don't have anything to do with them, if you were wondering).
Tom

Does it really work? My operator requires proxy. Is there an option in the settings to enter proxy server? I went on their website and saw there is only registrar, user and password. Also, when I try to purchase Premium Edition with G729 codec, I get an option to purchase V Phone - s60 - Standard Edition for 8.95 dollars. The price is really good but I wonder if it really works.

Nokia N95 SIP connection disconnecting

I have been using a SIP provider on my Nokia N95 for months.
Since two days it started disconnecting the Wireless LAN and would not connect to the SIP profiles. I have reconfigured the settings as well. Same service works fine on computer.
Could it be a bug or there is a solution for this behaviour?

Make sure you've entered the realm correctly in the registrar settings. That's a classic gotcha.
And before you ask, no, I don't know what the realm is for your system.
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6300i and VoIP/SIP connections

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