Maximum sip calls supported by CUBE 3945E
Hi All,
According to the link below the router 3945E supported maximum sessions is 2500.
http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-border-element/order_guide_c07_462222.html
My question is what is the command to see the active sessions on the CUBE, I use show sip-ua calls summary but I'm not sure if it is the same count for the CUBE sessions.
Kind regards!
Please refer this URL
https://supportforums.cisco.com/document/123281/how-check-active-call-count-sip
//Suresh
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Similar Messages
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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: -
Cisco ISR G2 SIP Calls Capacity
Dear all,
We're planning for Cisco Voice Gateway configuration with SIP trunk, till now no E1s are used.
I would like to know how can we calculate the number of simulataneous calls that a cisco ISR G2 router (1921. 2921.3945,etc...) can support ?
How much sip simultaneous calls each ISR G2 model can support ?
Is it better to use SIP or we must get into E1 PRI ?
Regards,The Q and A below has the call capacity you are looking for
Table 1. Number of IP-to-IP Calls per Platform
Platform
Maximum Number of Simultaneous Calls (Flow-Through)
Cisco 3945E
2500
Cisco 3925E
2100
Cisco 3945
950
Cisco 3925
800
Cisco 2951
500
Cisco 2921
400
Cisco 2911
200
Cisco 2901
100
Cisco ASR 1004; and Cisco ASR 1006 Router Processor 2 (RP2)
5000; 16000*
Cisco ASR 1002, ASR 1004, and ASR 1006 RP1
1750
Cisco AS5350XM and AS5400XM
600
Cisco 3845
500
Cisco 3825
400
Cisco 2851
225
Cisco 2821
200
Cisco 2811
110
Cisco 2801
55
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/prod_qas09186a00801da69b.html
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"opportunity is a haughty goddess who waste no time with those who are unprepared" -
Callcentric SIP Trunk (ITSP -- 2811 CUBE -- CUCM 8.6
I have a SIP trunk from call centric that goes into my lab gear - they appear to be a good sip service due to cost but I'm having some trouble getting calls to route correctly. The call flow is Callcentric.com ITSP (SIP) --> 2811 (acting as cube) -->SIP Trunk --> CUCM 8.6. Phones are registered to CUCM.
I have the sip trunk registered and calls come in to the router (I see them in ccsip message/call debugs) The 2811 running 15.1(4)M7). Callcentric sends the username of the customer in the sip Invite instead of the called number, the called number is in the TO field. I have several DID’s from Callcentric (18452055544, 18452055545, 18452055546) for my lab. There are a few configs on here for CME where the customer number (17772253754) is simply translated to their phone DN - which is fine if you only have 1 DN with callcentric but more than 1 and thats not feasible since every inbound did will be matched to that 17772253754 translation/phone dn.
I’m using the a guide from http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/ using the Copy function as described http://www.cisco.com/c/en/us/products/collateral/ios-nx-os-software/ios-software-release-15-1-3-t/product_bulletin_c25-635704.html
I haven’t been able to find anything where they actually explain all the header fields so Its mostly trial and error.. so far mostly error. I think I’m close.. but who knows. Any assistance would be greatly appreciated
voice class sip-profiles 1
request INVITE peer-header sip TO copy ".sip:(.*)@." u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
CUCM (single/pub)- 192.168.1.200
2811 acting as cube - 192.168.1.203
Calling Number - 18165297500
Called Number - 18452055544
vrtr1#show sip register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
17772253754 -1 20 yes
vrtr1#
The Call Setup Information is:
Call Control Block (CCB) : 0x49646C28
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 18165297500
Called Number : 17772253754 (my customer number not called number)
Source IP Address (Sig ): 192.168.1.203 (my 2811 router)
Destn SIP Req Addr:Port : 204.11.192.159:5080
Destn SIP Resp Addr:Port : 204.11.192.159:5080
Destination Name : 204.11.192.159
Feb 14 11:20:53.303: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>
i: [email protected]
CSeq: 1 INVITE
Max-Forwards: 8
m: <sip:[email protected]:5080;transport=udp>
Supported: timer
c: application/sdp
l: 350
v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.159
s=sip call
c=IN IP4 204.11.192.159
t=0 0
m=audio 61094 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
From: <sip:[email protected]>;tag=3601387252-874282
To: <sip:[email protected]>
Date: Fri, 14 Feb 2014 17:20:53 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
From: <sip:[email protected]>;tag=3601387252-874282
To: <sip:[email protected]>;tag=35399D8-63
Date: Fri, 14 Feb 2014 17:20:53 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=1
Content-Length: 0
Feb 14 11:20:53.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>;tag=35399D8-63
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 10
l: 0
u all
Feb 14 11:20:57.067: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-6bceae47efe9f53b4234698a32ac8beb
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>;tag=35399D8-63
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 8
l: 0
************************** Running Config **************************
sh run
vrtr1#sh running-config
Building configuration...
Current configuration : 4189 bytes
! Last configuration change at 00:34:03 CST Fri Feb 14 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
version 15.1
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
no service password-encryption
hostname vrtr1
boot-start-marker
boot system flash:
boot system flash flash:c2800nm-ipvoicek9-mz.151-4.M7.bin
boot-end-marker
card type t1 0 0
logging buffered 4096 notifications
enable password cisco
no aaa new-model
memory-size iomem 5
clock timezone CST -6 0
clock summer-time CST recurring
no network-clock-participate wic 0
dot11 syslog
ip source-route
ip cef
ip name-server 192.168.1.9
no ipv6 cef
multilink bundle-name authenticated
voice service voip
ip address trusted list
ipv4 192.168.1.0 255.255.255.0
ipv4 204.11.192.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server expires max 1800 min 1800
localhost dns:callcentric.com
outbound-proxy dns:callcentric.com
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class sip-profiles 1
request INVITE peer-header sip TO copy ".sip:(.*)@." u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2811 sn FTX1133A4QR
controller T1 0/0/0
cablelength long 0db
interface FastEthernet0/0
description ** LAN **
ip address 192.168.1.203 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.1.203
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 192.168.1.1
snmp mib persist circuit
control-plane
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/1/2
voice-port 0/1/3
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 192.168.1.200
ccm-manager config
mgcp
mgcp call-agent 192.168.1.200 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface FastEthernet0/0
mgcp bind media source-interface FastEthernet0/0
mgcp profile default
dial-peer voice 999100 pots
service mgcpapp
port 0/1/0
dial-peer voice 999101 pots
service mgcpapp
port 0/1/1
dial-peer voice 999102 pots
service mgcpapp
port 0/1/2
dial-peer voice 999103 pots
service mgcpapp
port 0/1/3
dial-peer voice 999010 pots
service mgcpapp
port 0/1/0
dial-peer voice 6 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 17772253754
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 7 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 1845205554[4-5]
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
sip-ua
credentials username 17772253754 password 7 106C1B49111F17194D realm callcentric.com
authentication username 17772253754 password 7 08035E1E1D11000553 realm callcentric.com
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:callcentric.com expires 3600 port 5060 transport udp
registrar dns:callcentric.com expires 3600
sip-server dns:callcentric.com
host-registrar
line con 0
line aux 0
line vty 0 4
password cisco
login
transport input all
scheduler allocate 20000 1000
ntp server 199.102.46.72
ntp server 23.227.162.123 prefer
end
exitThank you for the reply. I've updated the dial-peers as sugested. I'm now seeing an invite go out to my CUCM however the call fails with a 403 (forbidden) which appears to come from the ITSP (Callcentric). I've included a new set of ccsip message debugs and the dial-peers as adjusted. Please let me know what you think.
dial-peer voice 6 voip
description ## INBOUND CALL from ITSP ##
session protocol sipv2
session target sip-server
incoming called-number 17772253754
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
dial-peer voice 100 voip
description ## INBOUND DID to CUCM ##
destination-pattern 17772253754
session protocol sipv2
session target ipv4:192.168.1.200
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
dial-peer voice 7 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 1845205554[4-5]
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
Feb 15 10:18:11.424: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
f: ;tag=3601469891-655
t: [email protected]>
i: [email protected]
CSeq: 1 INVITE
Max-Forwards: 8
m:
Supported: timer
c: application/sdp
l: 350
v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.164
s=sip call
c=IN IP4 204.11.192.164
t=0 0
m=audio 61782 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
Feb 15 10:18:11.456: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
From: ;tag=3601469891-655
To: [email protected]>
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 15 10:18:11.460: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392481091
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273
v=0
o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18168 RTP/AVP 18 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Feb 15 10:18:11.552: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35;rport=57100;received=24.123.98.94
f: [email protected]>;tag=8408644-12C8
t:
i: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="8ae6b7b1cea74cf401e8a26fd3c7371b", opaque="", stale=TRUE, algorithm=MD5
l: 0
Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392481091
Contact:
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="17772253754",realm="callcentric.com",uri="sip:[email protected]:5060",response="a381f10fbbfbd255b444569fef0dddfe",nonce="8ae6b7b1cea74cf401e8a26fd3c7371b",opaque="",algorithm=MD5
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273
v=0
o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18168 RTP/AVP 18 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Feb 15 10:18:11.648: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Incorrect Authentication
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3;rport=57100;received=24.123.98.94
f: [email protected]>;tag=8408644-12C8
t:
i: [email protected]
CSeq: 102 INVITE
l: 0
Feb 15 10:18:11.660: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Feb 15 10:18:11.660: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
From: ;tag=3601469891-655
To: [email protected]>;tag=8408714-B60
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=57
Content-Length: 0
Feb 15 10:18:11.752: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
apsc-vrtr1#ACK sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
f: ;tag=3601469891-655
t: [email protected]>;tag=8408714-B60
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 10
l: 0
vrtr1#u al
Feb 15 10:18:14.776: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-e437c2c5cac5f1a6e147c1cd7c98aad7
f: ;tag=3601469891-655
t: [email protected]>;tag=8408714-B60
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 8
l: 0 -
Can someone explain how h323 to SIP calls work & vice versa.
The following messages are mapped:
SIP <---> H323
INVITE - SETUP
100 Trying - Call Proc
180 Ringing - Alerting
183 Session Progress - Progress
200 OK (for INVITE) - Connect
BYE - Release Complete
With H323 to SIP CUBE, if fast start occurs on one leg, early offer needs to happen on the other (and vice versa). Most SIP devices these days to early offer (SDP in invite) so you typically need fast start enabled on both directions of the H323 leg for this design.
Check out this link for more information:
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-h323sip_ps5640_TSD_Products_Configuration_Guide_Chapter.html -
Outbound SIP calls to invalid numbers
what is the expected behavior for a sip call to an invalid number? and what would cause it to behave differently?
Hi Ronald,
Whats the call flow ex IP Phone --SCCP-> CUCM --SIP --> CUBE -SIP-> Telco
If called number is invalid then call should not ring. Why is far end responding back with ringing.
Can you grab below debugs from the VG
++ debug voice ccapi inout
++ debug ccsip messages
HTH,
Regards,
Mohammed Noor -
Help needed! Just want to cancel my year subscription. No answer from support via mail, noone answers when I call, support chat don't work.
Hi there
I'll pass your details to our Russian team and they'll contact you to assist further.
Kind regards
Bev -
Hi there,
I'm having problems modifying the 'Dialed Number (DN)' text box under 'Advanced Configuration->Patterns for RNA timeout on outbound SIP calls' of the SIP tab in the Cisco Unified Customer Voice Portal 8.5(1) opsconsole. In a nut shell, I need to change the RNA timeout but some reason when typing into the Dialed Number text box, the input is not taken. The reason I want to change this settings is because my ICM Rona is not working with CVP:
https://supportforums.cisco.com/thread/2031366
Thanks in advance for any help.
Carlos A Trivino
[email protected]Hello Dale,
CVP doesn't allow you to exceed the RNA more than 60 Seconds. If you want to configure the timer for DN Patterns you should do it via OPS console, It would update the sip.properties files in correct way, the above way is incorrect.
Regards,
Senthil -
SIP to SIP Call Failures on CME to CME - sip-ua conflict/issue?
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.
Phones DN22xx - London CME 2801 - PIX505 --- Internet ---ASA5505 - India CME 2801 - Phones DN400x
I have configured dial peers on both CME's and the IPSec VPN. I can ping between both systems. The VPN allows traffic between the interface IP's of the CME systems only.
London CME (local SCCP phones 22xx):
interface FastEthernet0/0.100
encapsulation dot1Q 100 native
ip address 10.0.10.250 255.255.255.0
voice class codec 101
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 25 voip
description *** SIP Peer to India ***
answer-address 400.
destination-pattern 400.
voice-class codec 101
session protocol sipv2
session target ipv4:192.168.15.10
incoming called-number 400.
no vad
India CME (Local SSCP phones 400x):
interface FastEthernet0/0
ip address 192.168.15.10 255.255.255.0
voice class codec 100
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 10 voip
description *** SIP Peer to London UK ***
answer-address 22..
destination-pattern 22..
voice-class codec 100
session protocol sipv2
session target ipv4:10.0.10.250
incoming called-number 22..
no vad
The CME system at India also has an existing SIP dial peer to a service provider and has sip-ua configured (username, password, realm and registrar).
A call from India (4005) to London (DN2207) fails, the ccsip debug attached. I'm assuming its because the sip-ua configuration is being used for these calls to when I don't want it to be. The from field shows âFrom: <sip:[email protected]â when I need this to be the internal IP 192.168.15.10.
Can anyone offer any assistance with this?
Regards,
ChrisHi,
thanks for your input however thats not the problem. 201.196.128.56 isn't an address on the router, it only has one IP and its 192.168.15.10.
The 201.196.128.56 address is the NAT'd address on the firewall. So that when a SIP call is made to the internet with sip-ua the from address is the public IP.
Chris -
Hi,
I have many questions, and have had a very frustrating time to get in touch with the correct person.
I have called support twice, both times even though it was a technical question about cloud, they refused to forward my calls and told me there is NO technicall support for Cloud and that I have to place my questions here in the forum.
I am CONTINUALLY told to change my password, 2 and 3 times a week. I am a sysAdmin, my passwords are a MINIMUM of 13 Char/UP/Lower/Number/Special Combo, always have been. They work out to a 100% approval in any algorythm checker. Yes, I was one of the persons who had their password stolen which has caused me a tad bit of inconvenience, but such is life. Is This Normal? I have gone on twitter a few times to verify the emails, I have got them so often.
My "Cloud Sync" Hasn't worked since PhotoShop's Site Closed, I know this beause I stored my 20gigs of files, .ai, and .eps, .psd eveything that I couldn't move to Revel (which btw, I never got a refund for having to purchase the revel account 20gigs, when photoshop was closed, which also was never refunded.)
I actually wasn't too worried about the cloud issue because I only work at home, if I go somewhere else.. work's not going with me
Since about three weeks ago and currently I am having NOTHING but problems with cloud, The errors in my Event Log are NUMEROUS
Activation context generation failed for "C:\Program Files (x86)\Adobe\Adobe Dreamweaver CC\Dreamweaver.exe".Error in manifest or policy file "" on line . A component version required by the application conflicts with another component version already active. Conflicting components are:. Component 1: C:\Windows\WinSxS\manifests\amd64_microsoft.windows.common-controls_6595b64144ccf1df_6.0. 7601.17514_none_fa396087175ac9ac.manifest. Component 2: C:\Windows\WinSxS\manifests\x86_microsoft.windows.common-controls_6595b64144ccf1df_6.0.76 01.17514_none_41e6975e2bd6f2b2.manifest.
In xml form:
- <Event xmlns="http://schemas.microsoft.com/win/2004/08/events/event">
- <System>
<Provider Name="SideBySide" />
<EventID Qualifiers="49409">80</EventID>
<Level>2</Level>
<Task>0</Task>
<Keywords>0x80000000000000</Keywords>
<TimeCreated SystemTime="2014-03-28T08:07:04.000000000Z" />
<EventRecordID>2269809</EventRecordID>
<Channel>Application</Channel>
<Computer>Celtic-Surfer</Computer>
<Security />
</System>
<EventData>
<Data>C:\Windows\WinSxS\manifests\amd64_microsoft.windows.common-controls_6595b64144ccf1df_6.0.7 601.17514_none_fa396087175ac9ac.manifest</Data>
<Data>C:\Windows\WinSxS\manifests\x86_microsoft.windows.common-controls_6595b64144ccf1df_6.0.760 1.17514_none_41e6975e2bd6f2b2.manifest</Data>
<Data />
<Data />
<Data />
<Data />
<Data />
<Data />
<Data />
<Data />
<Data>C:\Program Files (x86)\Adobe\Adobe Dreamweaver CC\Dreamweaver.exe</Data>
<Data />
<Data />
<Data />
<Data />
<Data />
<Data />
<Data />
<Data />
<Data />
</EventData>
</Event>
manifest for x86/64bit at both dll files compiled so I cannot review their content.
However it should be noted at this time, I did not invoke Dreamweaver, the Cloud did, and this is not the first time.
This particular error occured 5 times in 3 seconds.
Same two errors at 3:49pm on the 28th.
Today, I went and bought a new solid state drive. Brand New Fresh OS, Downloaded Cloud, CC, and of course The Edge Products, which I dearly love btw. Everything seems to be working fine, EXCEPT,
1.) where did CS6 Products Go?
2.) Why after being a Faithful Product User/Cloud Supporter for the Last two years, after going through all the trials and tribulations with Adobe, am I being asked to renew my license for a "One Time Per Year Fee" which is MORE than my monthly amount? ( deal: 599.80; payments at 49.95 per month is 599.40 Go Figure? ) So, I'm being asked to pay the same amount, make it a single payment and get half the software, litterally. Did I miss a paticularly important anouncement?
3.) EVERYONE OF MY IRREPLACABLE FILES IN MY CLOUD ALL MY AI FILES, are gone. Yesterday, it would seem My having the SYNC TURNED OFF was over-ridden by the system (Adobes) and they have upload 'garbage' files, and my original files, the caligraphy files are no longer there. My gorgeous image of a little boy with flowers is no longer there. I PAID ALMOST A GRAND for these files. Stored them on Adobe, because, well, you deal with images, so you're the best right, that's why I bought extra storage on photoshop.com, then was told I would have to move to revel, but not the ai, eps, or pdf and i don't remember about png's. files, so I moved everything to revel (purchased the extra storage there.) AND put everything on the cloud and turned off my file sync. Not everyone is an idiot, maybe something is turned off, there's a reason for it, how dare someone override MY SETTINGS. They could have emailed, called, chat, google chat, irc, or snail mailed me, it's pretty easy to get my attention.
I could just cry.
lost my files,
asked to pay one time instead on monthy, and there's not even a discount and no additional incentive. Lost the ability to work without having to be connected to the internet, which will be delimited here very shortly since I'm on comcast.
and the worst, the very absolutely worst part is this is the SECOND time in a row now, I've had an issue with Adobe that has gone on for over a year while support TELLS ME A PAYING CUSTOMER, that 'gee, sorry, no support for cloud" , "No I can't forward you to a tech" (direct quote) and "The Only support for any cloud issues is the forum"
How can I handle this in the future, my nerves just can't take it anymore. Seriously, I'm just shaking now I'm so upset about this and I really hate writing something like this, or airing issues in public that really should be, imho, private.Hi Ken,
Wow, I've logged into adobe and never seen your response until now.
I need to make the forums more of a daily stop I guess.
Thank you for your response, When I go to the Archives, it states I have none. In addition, previously when I'd tried to move items to the archive, they just disappeared
Currently my desktop is not connecting at all, Everytime I connect, it's states "We have now logged you out" and requests I login again.
I have of course rebooted,
Gone into taskmanager, killed all the relevant threads, tried again to login through the desktop, no luck, I'll have to uninstall it, make sure the threads are killed, then reboot it and then reinstall it, but I've not had the time to do that.
On another note, and just as a point of reference, myself, personally, I hate these "cloud" file repositories (not to be confused with adobe cloud services which I love) but the file repositories themselves imho, are a blackhole of resource usage when one doesn't/isn't using them,
Is there a way we can use it more of a "ftp" sort of thing, when I want to I can put files there? There is no way to "Download" the repository, no way to download the folders.. only individual files and then it takes I think three different steps before the download starts.. I find this very inhibiting. Just an fyi. for what it's worth.
Please, feel free to contact me,
I'm on g+ chat [email protected] -
On my iPad Mini, when I delete an email from the inbox (and it goes into Trash), it immediately deletes that email from the server (ATT/Yahoo). I've called support for both ATT and Apple and can't find a way to have the emails remain on the server.
My husband and I share an email address and if he misses a day checking email on his notebook, he doesn't receive the items that I have sent to Trash on my iPad Mini.
Appreciate suggestions.
[The curious thing is, if I "move" the deleted emails from Trash on my iPad Mini into the Inbox, I can go to my notebook, open my online email and they are back in my ATT/Yahoo Inbox.]Is your Yahoo account set up as POP3 or IMAP?
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I have tried since Chrismas to use my Ipod touch and have not been able to! I have uninstalled and reinstalled itunes more often then I care to admit! I called support and when I found out there is a charge ($59.95) I declined and hung up. I now can't even download itunes! I guess I find that a little fishy. I was so happy with this gift and now I am so frustrated I am about to return it and never purchase another apple product again! I have gotten nothing but the run around when trying to get assistance for a very expensive music player! HELP PLEASE
Start here:
iPhone, iPad, or iPod touch: Device not recognized in iTunes for Windows -
Error in application accurred please call support quoting error code 2907
error in application accurred please call support quoting error code 2907
I suggest you call support and quote error code 2907.
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Dear Sir, i have problem in my Iphone 5 that during receiving any calls always appeare message that error in application occurred please call support quoting error code 2907. I need to solve my problem.
The problem this message always appeare when receiving the calls and requesting either accept or cancel.
Can you advice me if i have wrong application or i need to remove some applications. -
I for got the answers to my security questions and my rescue address is misspelled is there anyway to fix my rescue address so i can reset my security question answers? I called support and they said it was nothing they could do! Can somebody please help!?
Did you not call the Account Security Team? It's pretty new.
Check the AppleCare number for your country here:
http://support.apple.com/kb/HE57
Call them up, and let them know you would like to be transferred to the Account Security Team.
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