Rogue SIP calls on C40

Hello,
We have a VC system with public IP address registered to VCS-E receiving various rogue SIP calls. All of these SIP calls are 3 - 6 digit alias lasting for 32 sec each. The incoming call address is: alias@ our VC system's public IP address. All calls are video calls at 384k
To avoid the calls, I have turned off the SIP option and since then we have not received any of these calls. However, we need to connect to other devices using SIP. Is there another way to stop these incoming calls?
Thanks in advance for your help.
Amrit

Set the following on your endpoints:
xConfiguration SIP ListenPort: Off
xConfiguration SIP Profile 1 Outbound: On
See bug CSCue55239 for more details.
You'll also need to take steps to secure your VCS if you haven't already, turning off SIP UDP will stop the SIP calls. However, within the last year we've seen these calls come over H323 TCP, the only way to stop the H323 calls is either secure the endpoint behind your firewall, and use a CPL script on the VCS. See the sourceh323idcisco-incomingcalls discussion on how to setup a CPL.
FYI, searching the forums would be my first place to look, or bug search. It's been asked all over the place in the forums.

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[email protected]

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Hi,
I have two existing CME systems which I wish to allow internal calls between. These calls will go over an IPSec VPN. However the calls are failing.
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Hi,
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Sent:
SIP/2.0 400 Bad Request - 'Invalid Host'
Via: SIP/2.0/UDP 107.6.67.238:5060;branch=z9hG4bK000d3c36;rport
From: "+19144410197" <sip:[email protected]>;tag=as7439b9c1
To: <sip:[email protected]:1061>;tag=829C8-2532
Date: Sun, 31 Aug 2014 12:42:04 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=100
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
I also am getting this:
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SIP Call error tracing is enabled
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Aug 31 12:45:07.359: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr
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Sujish Sudhakar

Hi Sudhakaran,
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INVITE - SETUP
100 Trying - Call Proc
180 Ringing - Alerting
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Outbound SIP calls to invalid numbers

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Hi Ronald,
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Mohammed Noor

SIP Calls Drop. Receive Bye From Cube 15min,30min, 45min

Hello,
Running into an odd issue. I've seen several others having this problem with calls dropping after 15min duration. But this is a bit different. Sometimes long duration calls drop at 15min. Some at 30min, others at 45min. And sometimes not at all. Call flow is such.
8831-sip--CUCM--sip--Cube--ITSP
I'm convinced this is likely a problem with the refresh timer. But I can't explain why it wouldn't just fail only at 15min. It's also interesting to note I've only seen this on the 8831. I tried getting the issue with debugs from the cube but of course it didn't happen once I turned on ccsip message.
From the callmanager traces I see the bye arrive from cube with Reason Q.850 cause=102.
The CUCM version is 9.1.2 and cube is 15.2(4)M1. I did see some odd defect in 15.1 related to this where the refresh on the cube would send out 3 invites to the ITSP on an update. I guess it would have only 33% chance of getting it right. Any help someone could provide I'd appreciate it.

Thanks for the replies.
So was able to capture it while had debugs running. This time it disconnected after an hour. Same cause=102.
Now here is where it gets interesting in the debugs. I see an invite is sent 3 seconds from callmanager. I assume this is a refresher with the same call-id. Cube receives it and sends out to ITSP. With a new call-id. We then receive a bye from ITSP cause=86. Which then of course is sent to callmanager. Here are the relevent sections of debugs.
Received from cucm to cube:
820421: May 6 09:00:42.976: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:43           2365082: Received:
2014-05-06 09:00:43           2365083: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
2014-05-06 09:00:43           2365084: Via: SIP/2.0/TCP 10.38.246.136:5060;branch=z9hG4bK28bab16dbd5664
2014-05-06 09:00:43           2365085: From: "Marcos Vazquez" <sip:[email protected]>;tag=3831180~dfbf10b3-6c69-4443-852f-cbf609935a6f-35009402
2014-05-06 09:00:43           2365086: To: <sip:[email protected]>;tag=5EBA2282-19C8
2014-05-06 09:00:43           2365087: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:43           2365088: Call-ID: [email protected]
2014-05-06 09:00:43           2365089: Supported: 100rel,timer,resource-priority,replaces
2014-05-06 09:00:43           2365090: Min-SE: 1800
2014-05-06 09:00:43           2365091: User-Agent: Cisco-CUCM9.1
2014-05-06 09:00:43           2365092: Allow: INVITE, OPTIONS, I
2014-05-06 09:00:43           2365093: NFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
2014-05-06 09:00:43           2365094: CSeq: 106 INVITE
2014-05-06 09:00:43           2365095: Max-Forwards: 70
2014-05-06 09:00:43           2365096: Expires: 300
2014-05-06 09:00:43           2365097: Allow-Events: presence, kpml
2014-05-06 09:00:43           2365098: Supported: X-cisco-srtp-fallback
2014-05-06 09:00:43           2365099: Supported: Geolocation
2014-05-06 09:00:43           2365100: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:43           2365101: Remote-Party-ID: "Marcos Vazquez" <sip:[email protected]>;party=calling;screen=yes;privacy=off
2014-05-06 09:00:43           2365102: Contact: <sip:[email protected]:5060;transport=tcp>
2014-05-06 09:00:43           2365103: Content-Type: application/sdp
2014-05-06 09:00:43           2365104: Content-Length: 371
2014-05-06 09:00:43           2365105:
2014-05-06 09:00:43           2365106: v=0
2014-05-06 09:00:43           2365107: o=CiscoSystemsCCM-
2014-05-06 09:00:43           2365108: SIP 3831180 1 IN IP4 10.38.246.136
2014-05-06 09:00:43           2365109: s=SIP Call
2014-05-06 09:00:43           2365110: c=IN IP4 10.96.5.28
2014-05-06 09:00:43           2365111: b=TIAS:64000
2014-05-06 09:00:43           2365112: b=AS:64
2014-05-06 09:00:43           2365113: t=0 0
2014-05-06 09:00:43           2365114: m=audio 31146 RTP/AVP 18 0 116 101
2014-05-06 09:00:43           2365115: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:43           2365116: a=ptime:20
2014-05-06 09:00:43           2365117: a=rtpmap:116 iLBC/8000
2014-05-06 09:00:43           2365118: a=ptime:20
2014-05-06 09:00:43           2365119: a=maxptime:60
2014-05-06 09:00:43           2365120: a=fmtp:116 mode=20
2014-05-06 09:00:43           2365121: a=rtpmap:18 G729/8000
2014-05-06 09:00:43           2365122: a=ptime:20
2014-05-06 09:00:43           2365123: a=fmtp:18 annexb=no
2014-05-06 09:00:43           2365124: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:43           2365125: a=fmtp:101 0-15
2014-05-06 09:00:43           2365126: 5820422: May 6 09:00:42.978: //3024943/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
Sent to ITSP:
Sent: Which looks like 3 are sent.
2014-05-06 09:00:43           2365128: INVITE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:43           2365129: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D699C1126
2014-05-06 09:00:43           2365130: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:43           2365131: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:43           2365132: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:43           2365133: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:43           2365134: Call-ID: [email protected]
2014-05-06 09:00:43           2365135: Supported: 100rel,timer,resource-priority,replaces,sdp-an
2014-05-06 09:00:43           2365136: at
2014-05-06 09:00:43           2365137: Min-SE: 1800
2014-05-06 09:00:43           2365138: Cisco-Guid: 3422833792-0000065536-0000746374-2297832970
2014-05-06 09:00:43           2365139: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:43           2365140: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
2014-05-06 09:00:43           2365141: CSeq: 105 INVITE
2014-05-06 09:00:43           2365142: Max-Forwards: 70
2014-05-06 09:00:43           2365143: Timestamp: 1399388442
2014-05-06 09:00:43           2365144: Contact: <sip:[email protected]:5060>
2014-05-06 09:00:43           2365145: Expires: 60
2014-05-06 09:00:43           2365146: Allow-Events: telephone-event
2014-05-06 09:00:43           2365147: Content-Type: application/sdp
2014-05-06 09:00:43           2365148: Content-Length: 334
2014-05-06 09:00:43           2365149:
2014-05-06 09:00:43           2365150: v=0
2014-05-06 09:00:43           2365151: o=CiscoSystemsSIP-GW-UserAgent 8182 4488 IN IP4 12.17.223.243
2014-05-06 09:00:43           2365152: s=SIP Call
2014-05-06 09:00:43           2365153: c=IN IP4 1
2014-05-06 09:00:43           2365154: 2.17.223.243
2014-05-06 09:00:43           2365155: t=0 0
2014-05-06 09:00:43           2365156: m=audio 18760 RTP/AVP 18 0 100 101
2014-05-06 09:00:43           2365157: c=IN IP4 12.17.223.243
2014-05-06 09:00:43           2365158: a=rtpmap:18 G729/8000
2014-05-06 09:00:43           2365159: a=fmtp:18 annexb=no
2014-05-06 09:00:43           2365160: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:43           2365161: a=rtpmap:100 X-NSE/8000
2014-05-06 09:00:43           2365162: a=fmtp:100 192-194
2014-05-06 09:00:43           2365163: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:43           2365164: a=fmtp:101 0-15
2014-05-06 09:00:43           2365165: 5820423: May 6 09:00:42.978: //3024942/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:43           2365166: Sent:
2014-05-06 09:00:43           2365167: SIP/2.0 100 Trying
2014-05-06 09:00:43           2365168: Via: SIP/2.0/TCP 10.38.246.136:5060;branch=z9hG4bK28bab16dbd5664
2014-05-06 09:00:43           2365169: From: "Marcos Vazquez" <sip:[email protected]>;tag=3831180~dfbf10b3-6c69-4443-852f-cbf609935a6f-35009402
2014-05-06 09:00:43           2365170: To: <sip:[email protected]>;tag=5EBA2282-19C8
2014-05-06 09:00:43           2365171: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:43           2365172: Call-ID: [email protected]
2014-05-06 09:00:43           2365173: CSeq: 106 INVITE
2014-05-06 09:00:43           2365174: Allow-Events: telephone-event
2014-05-06 09:00:43           2365175: Server: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:43           2365176: Content-Length: 0
2014-05-06 09:00:43           2365177:
2014-05-06 09:00:43           2365178: 5820424: May 6 09:00:43.479: //3024943/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:43           2365179: Sent:
2014-05-06 09:00:43           2365180: INVITE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:43           2365181: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D699C1126
2014-05-06 09:00:43           2365182: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:43           2365183: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:43           2365184: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:43           2365185: Date: Tue, 06 May 2014 15:00:43 GMT
2014-05-06 09:00:43           2365186: Call-ID: [email protected]
2014-05-06 09:00:43           2365187: Supported: 100rel,timer,resource-priority,replaces,sdp-an
2014-05-06 09:00:43           2365188: at
2014-05-06 09:00:43           2365189: Min-SE: 1800
2014-05-06 09:00:43           2365190: Cisco-Guid: 3422833792-0000065536-0000746374-2297832970
2014-05-06 09:00:43           2365191: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:43           2365192: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
2014-05-06 09:00:43           2365193: CSeq: 105 INVITE
2014-05-06 09:00:43           2365194: Max-Forwards: 70
2014-05-06 09:00:43           2365195: Timestamp: 1399388443
2014-05-06 09:00:43           2365196: Contact: <sip:[email protected]:5060>
2014-05-06 09:00:43           2365197: Expires: 60
2014-05-06 09:00:43           2365198: Allow-Events: telephone-event
2014-05-06 09:00:43           2365199: Content-Type: application/sdp
2014-05-06 09:00:43           2365200: Content-Length: 334
2014-05-06 09:00:43           2365201:
2014-05-06 09:00:43           2365202: v=0
2014-05-06 09:00:43           2365203: o=CiscoSystemsSIP-GW-UserAgent 8182 4488 IN IP4 12.17.223.243
2014-05-06 09:00:43           2365204: s=SIP Call
2014-05-06 09:00:43           2365205: c=IN IP4 1
2014-05-06 09:00:44           2365206: 2.17.223.243
2014-05-06 09:00:44           2365207: t=0 0
2014-05-06 09:00:44           2365208: m=audio 18760 RTP/AVP 18 0 100 101
2014-05-06 09:00:44           2365209: c=IN IP4 12.17.223.243
2014-05-06 09:00:44           2365210: a=rtpmap:18 G729/8000
2014-05-06 09:00:44           2365211: a=fmtp:18 annexb=no
2014-05-06 09:00:44           2365212: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:44           2365213: a=rtpmap:100 X-NSE/8000
2014-05-06 09:00:44           2365214: a=fmtp:100 192-194
2014-05-06 09:00:44           2365215: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:44           2365216: a=fmtp:101 0-15
2014-05-06 09:00:44           2365217: 5820425: May 6 09:00:44.479: //3024943/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:44           2365218: Sent:
2014-05-06 09:00:44           2365219: INVITE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:44           2365220: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D699C1126
2014-05-06 09:00:44           2365221: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:44           2365222: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:44           2365223: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:44           2365224: Date: Tue, 06 May 2014 15:00:44 GMT
2014-05-06 09:00:44           2365225: Call-ID: [email protected]
2014-05-06 09:00:44           2365226: Supported: 100rel,timer,resource-priority,replaces,sdp-an
2014-05-06 09:00:44           2365227: at
2014-05-06 09:00:44           2365228: Min-SE: 1800
2014-05-06 09:00:44           2365229: Cisco-Guid: 3422833792-0000065536-0000746374-2297832970
2014-05-06 09:00:44           2365230: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:44           2365231: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
2014-05-06 09:00:44           2365232: CSeq: 105 INVITE
2014-05-06 09:00:44           2365233: Max-Forwards: 70
2014-05-06 09:00:44           2365234: Timestamp: 1399388444
2014-05-06 09:00:44           2365235: Contact: <sip:[email protected]:5060>
2014-05-06 09:00:44           2365236: Expires: 60
2014-05-06 09:00:44           2365237: Allow-Events: telephone-event
2014-05-06 09:00:44           2365238: Content-Type: application/sdp
2014-05-06 09:00:44           2365239: Content-Length: 334
2014-05-06 09:00:44           2365240:
2014-05-06 09:00:44           2365241: v=0
2014-05-06 09:00:44           2365242: o=CiscoSystemsSIP-GW-UserAgent 8182 4488 IN IP4 12.17.223.243
2014-05-06 09:00:44           2365243: s=SIP Call
2014-05-06 09:00:44           2365244: c=IN IP4 1
2014-05-06 09:00:44           2365245: 2.17.223.243
2014-05-06 09:00:44           2365246: t=0 0
2014-05-06 09:00:44           2365247: m=audio 18760 RTP/AVP 18 0 100 101
2014-05-06 09:00:44           2365248: c=IN IP4 12.17.223.243
2014-05-06 09:00:44           2365249: a=rtpmap:18 G729/8000
2014-05-06 09:00:44           2365250: a=fmtp:18 annexb=no
2014-05-06 09:00:44           2365251: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:44           2365252: a=rtpmap:100 X-NSE/8000
2014-05-06 09:00:44           2365253: a=fmtp:100 192-194
2014-05-06 09:00:44           2365254: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:44           2365255: a=fmtp:101 0-15
2014-05-06 09:00:45           2365256: 5820426: May 6 09:00:45.147: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:45           2365257: Received:
And then I don't see a response then send out a bye:
Sent:
2014-05-06 09:00:46           2365897: BYE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:46           2365898: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D69A54BC
2014-05-06 09:00:46           2365899: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:46           2365900: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:46           2365901: Date: Tue, 06 May 2014 15:00:44 GMT
2014-05-06 09:00:46           2365902: Call-ID: [email protected]
2014-05-06 09:00:46           2365903: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:46           2365904: Max-Forwards: 70
2014-05-06 09:00:46           2365905: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:46           2365906: Timestamp: 1399388446
2014-05-06 09:00:46           2365907: CSeq: 106 BYE
2014-05-06 09:00:46           2365908: Reason: Q.850;cause=86
2014-05-06 09:00:46           2365909: P-RTP-Stat: PS=180295,OS=3604444,PR=180354,OR=3607080,PL=0,JI=0,LA=0,DU=3603
2014-05-06 09:00:46           2365910: Content-Length: 0
2014-05-06 09:00:46           2365911:
2014-05-06 09:00:46           2365912: 5820458: May 6 09:00:46.479: //3024942/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:46           2365913: Sent:
2014-05-06 09:00:46           2365914: BYE sip:[email protected]:5060;transport=tcp SIP/2.0
2014-05-06 09:00:46           2365915: Via: SIP/2.0/TCP 10.38.246.166:5060;branch=z9hG4bK2D69A6E75
2014-05-06 09:00:46           2365916: From: <sip:[email protected]>;tag=5EBA2282-19C8
2014-05-06 09:00:46           2365917: To: "Marcos Vazquez" <sip:[email protected]>;tag=3831180~dfbf10b3-6c69-4443-852f-cbf609935a6f-35009402
2014-05-06 09:00:46           2365918: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:46           2365919: Call-ID: [email protected]
2014-05-06 09:00:46           2365920: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:46           2365921: Max-Forwards: 70
2014-05-06 09:00:46           2365922: Timestamp: 1399388446
2014-05-06 09:00:46           2365923: CSeq: 101 BYE
2014-05-06 09:00:46           2365924: Reason: Q.850;cause=102
2014-05-06 09:00:46           2365925: P-R
2014-05-06 09:00:46           2365926: TP-Stat: PS=180239,OS=3604780,PR=180295,OR=3604444,PL=0,JI=0,LA=0,DU=3603
2014-05-06 09:00:46           2365927: Content-Length: 0
2014-05-06 09:00:46           2365928:

SIP to SIP call on CME 8.6

Hi all, I'm trying to setup a video call between 9951 and IP door station 2N Helios IP that support H264 over SIP.
The audio call is working well but I see only black screen on my 9951, I don't see video also with other SIP client connected on the same CME 8.6.
This is my config:
voice service voip
no notify redirect ip2ip
clid network-provided
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco
modem passthrough nse codec g711alaw
sip
registrar server
asymmetric payload full
voice register global
mode cme
source-address 192.168.99.204 port 5060
bandwidth video tias-modifier 512000 negotiate end-to-end
max-dn 20
max-pool 10
load 9951 sip9951.9-1-1SR1
authenticate presence
authenticate register
authenticate realm cme
timezone 23
date-format D/M/Y
tftp-path flash:
file text
create profile sync 0010244609221862
network-locale IT
user-locale IT
ntp-server 192.168.99.254 mode directedbroadcast
camera
video
voice register dn 1
number 200
allow watch
name 200
no-reg
label 200
voice register dn 2
number 201
allow watch
name 201
no-reg
label 201
voice register dn 3
number 202
allow watch
name 202
no-reg
label 202
voice register pool 1
id mac 0000.0000.0000
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username 200 password xxxxx
codec g711ulaw
camera
video
voice register pool 2
id mac 0000.0000.0000
type 9951
number 1 dn 2
presence call-list
dtmf-relay sip-notify
username 201 password xxxxx
codec g711ulaw
camera
video
voice register pool 3
id mac 0000.0000.0000
number 1 dn 3
presence call-list
dtmf-relay rtp-nte
username 202 password xxxxx
codec g711ulaw
camera
video
Anyone have a CME with SIP video call working that can help me to debug my problem?
Thanks
Enrico.

Hi William
Thanks for the response.
i have attached screen shots from 2N Admin page.
can you please check and let me know the settings are correct.
door is not opening by dialing 11 during the call or after call.
Regards
shameer

E72 and SIP - unable to make sip calls

I have my E72 configured with SIP using the nokia sip client. Receiving calls work fine, however being called on my sip number causes the calling phone to go ' busy' when I pick up on the E72
Anyone a clue?
No software updates available for my E72
Thanks

Try another sim even if yours works on ither phones. Failing that, you could try resetting your phone. remenber to backup your spdata beforehand. If the reset does not fix the fault then I suggest having it checked out at your local Nokia care point. You can locate your nearest care point using the link below.
www.nokia.com/support

Configure goandcall SIP calls on E61i

Hi All,
I have configured goandcall SIP service and it got registerd on my nokia E61i. The problem is I cannot make outgoing calls. On the other hand when I configure the same on Fring with username, password, proxy : sip.goandcall.com
I am able to make both mobile and landline calls.
Please advice if there is any advance settings I have to make on my mobile. I have installed the nokia SIP_VoIP_Settings_v1_0_en software and set priority of codec G729 but no help.
Regards,
Abhi.

I found the solution for this.
On Nokia E series devices, you need to install a software from Nokia SIP_VoIP_Settings_v1_0_en at the URL : http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin...
When you install it, you need to go to Menu -> Installation -> SIP VOIP Setting -> NAT firewall Settings.
Here you need to configure the Domain parameters for sip.goandcall.com and put the value of STUN server as provided on your website.
Please note, we do not need to configure the outbound proxy in E61i ( referred as Proxy Server).
The reason we need to install the above software is because I could not find advance settings in Nokia to configure STUN server.
Hope this info helps.
-Abhi.

Internet (voip/SIP) calls on N97

Hi,
I was using Internet calls feature on N95. On N97 the only thing what I was able to do is to define SIP profile. It is defined. I am able to register to the SIP server ... but no way to use this server for Internet Calls. I simply do not have possibility to select Internet call. I may select Voice or Video but not Internet as it was on N95. Any ideas?
regards,

If you read the documentation that has been revised recently (yeah that is sneaky) you see alot of mentions of 'you may' or 'there might be' and somthing about a widget you might find that does not exist. Basically if Nokia wants to enable this with SIP Voip Settings App (like they had for 3.x and 2.x) they could do so quite easily.
See the attached for a screen snap of the new documentation.
The question is are they going to do this, or am I moving on to another company that is semi rational and has better communication skills than a 1 year old. I've been patient and silent since the day I unpacked this phone, but today I have lost patience with waiting for something i know they can enable in 2 minutes.
I love my purple screen, 1/2 day battery, sketchy touch screen and $800 price tag - I call it my Lumia 900
Attachments:
n97Capture.PNG ‏142 KB

Initiating a SIP call via Hyperlink

So we're using lync 2013, and I would like to be able to send out a hyperlink in an email that will actually initiate a SIP video call once clicked. Just so everyone understands the use case is wanting to dial into a cloud based service video service
that interworks h.323/SIP standards based systems with Lync
I can use:
Sip:[email protected] as a hyperlink and that will bring up the presence of the SIP contact since as we're federated with there domain. Ultimately it would be nice if it initiated the call just like clicking "online meeting"
This is the code that the join online meeting uses: conf:sip:https://meet.example.com/user/7322994 and I tried replacing the URL with a SIP address with no go, because it looks like it tries to find a meeting that is supposed to happen on the AVMCU.
So ultimately if there is a way to have a hyperlink, so Lync will initiate a SIP video call that would be great.

So we're using lync 2013, and I would like to be able to send out a hyperlink in an email that will actually initiate a SIP video call. Just so everyone understands the use case is wanting to dial into a cloud based service video service that interworks
h.323/SIP standards based systems with Lync
I can use:
Sip:[email protected] as a hyperlink and that will bring up the presence of the SIP contact since as we're federated with there domain. Ultimately it would be nice if it initiated the call just like clicking "online meeting"
This is the code that the join online meeting uses: conf:sip:https://meet.example.com/user/7322994 and I tried replacing the URL with a SIP address with no go, because it looks like it tries to find a meeting that is supposed to happen on the AVMCU.
So ultimately if there is a way to have a hyperlink, so Lync will initiate a SIP video call that would be great.

AAA and MD5 Configuration on SIP Calls

Olease can anyone help in AAA and MD5 configuration on Cisco 3640 running SIP. My carrier told me that the only way that my calls can be Authenticated is thru AAAor MD5, eg -
Host:
Authentication ID:
Secret:
Please I need your help thank you in advance.
Knmezi

MD5 authentication works similarly to plain text authentication, except that the key is never sent over the wire. Instead, the router uses the MD5 algorithm to produce a "message digest" of the key (also called a "hash"). The message digest is then sent instead of the key itself. This ensures that nobody can eavesdrop on the line and learn keys during transmission.
These protocols use MD5 authentication:
OSPF
RIP version 2
BGP
IP Enhanced IGRP
For AAA configuration refer to following url;
http://www.cisco.com/en/US/products/sw/secursw/ps2138/products_configuration_example09186a008017ee15.shtml

Rogue SIP calls on C40

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