CUC 8.0.2 AA sends calls to operator straight to VM
Auto-attendant sends calls to the designated operator's number (hitting 0 or dialing the individual's extension) straight to VM. I temporarily changed the operator to my extension (which works dialing through AA as did every other extension I checked) and it worked. The user was originally a local account on CUC. I renamed this account and changed the extension to a nonexistent number, and then imported the operator's account from CUCM (like the majority of the accounts are setup). The symptoms remained the same as if the problem is tied to that extension. Any thoughts/suggestions/next steps?
Hi William,
Can you have a look at this page under User>Edit User>and check the Transfer Calls to setting
Unity Directory > Edit Transfer Rule
Edit Transfer Rule (Standard)
Search Users Edit User Basics (buncle) Transfer Rules Edit Transfer Rule (Standard)
User
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Save Transfer Rule
Transfer Rules
Edit
User Basics
Password Settings
Change Password
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Message Waiting Indicators
Transfer Rules
Message Settings
Caller Input
Mailbox
Phone Menu
Playback Message Settings
Send Message Settings
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Greetings
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Alternate Names
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Contents
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Rule Name
Status
Disabled
Enabled With No End Date and Time
Enabled Until:
Date January February March April May June July August September October November December 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 2010 2011 2012 2013 2014
Time 01 02 03 04 05 06 07 08 09 10 11 12 00 05 10 15 20 25 30 35 40 45 50 55 AM PM
Transfer Action
Transfer Calls To:
Greeting
Extension
Transfer Type
Release to Switch Supervise Transfer
Rings to Wait For
Play the "Wait While I Transfer Your Call" Prompt
If Extension is Busy:
Send Callers to Voice Mail Put Callers on Hold Without Asking Ask Callers to Hold
Call Screening
Tell Me When the Call Is Connected
Tell Me Who the Call Is For
Ask Me If I Want to Take the Call
Ask for Caller's Name
Cheers!Rob
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How can I restrict my iPhone 4S to only receive or send calls or send receive texts from specific contacts and not all contacts in the phone.
Contact the carrier and see if they offer such a service.
This is not a feature of the iPhone and no App could restrict calls/sms. -
Unified Mobility, no audio on Send call to Mobile Phone
I'm using UCM 9.1 with Unified Mobility (xfer to alternate number) with good success if I follow a typical call flow:
Inbound call -> Ext -> Rings deskphone x seconds -> Rings mobile phone -> Answer mobile phone -> hangup -> Resume call on desk phone.
But if I pick up a call on my desk phone, and use the Mobility button to xfer a call to my mobile phone I get no audio:
Inbound call -> Ext -> Pickup desk phone -> Mobility soft button, 'Send call to Mobile Phone' -> Answer mobile phone, no audio -> Hang up mobile phone -> Resume call on desk phone (two-way audio).
Device wise the call flow is:
ITSP SIP trunk -> CUBE -> CUCM -> 7965 IP Phone.
Recently I reconfigured CUCM to use the CUBE for any MTP resources instead of the software option and I think I may have missed something.
CUBE config:
voice-card 0 dspfarm dsp services dspfarm!!!voice service voip ip address trusted list ipv4 173.46.30.218 ipv4 173.46.30.202 ipv4 10.0.6.30 ipv4 10.0.6.31 ipv4 10.0.6.33 ipv4 10.0.6.32 ipv4 10.1.1.4 ipv4 10.0.250.0 255.255.255.0 mode border-element allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip redirect ip2ip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip midcall-signaling passthru media-change early-offer forced no call service stop registration passthrough!voice class codec 1 codec preference 1 g711ulawsccp local GigabitEthernet0/0.42sccp ccm 10.0.6.30 identifier 1 version 7.0 sccp!sccp ccm group 1 bind interface GigabitEthernet0/0.42 associate ccm 1 priority 1 associate profile 1 register MTP_2951-01!dspfarm profile 2 transcode universal codec pass-through codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 11 associate application SCCP shutdown!dspfarm profile 3 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP!dspfarm profile 1 mtp codec pass-through codec g711ulaw maximum sessions hardware 15 associate application SCCP!
In CUCM I removed the software MTP from the MRG:
Not sure where to start troubleshooting this problem, any help is appreciated.
SteveThanks for the help gents. I couldn't get to this until we're out of office hours on the weekend.
Interestingly, I have no mid-call option in my dial-peers. This is a 2951 running 15.2(4)M2.
I double checked the MRGL, my phone is associated with it.
Codec is G711 on ITSP side, and on phones - I'm not sure I fully understand the use cases for MTP, this is something I need to research more.
I've included two ccsip message debugs, the first one is the existing issue of no audio (in either direction).
The second I've changed the midcall-signaling passthru option, dropping the media-change bit and we get audio in both directions for mobility except we use Unity call handlers for IVR functionality, and now when an inbound caller is forwarded to an extension we get no audio - obviously this is a game stopper.
In Unity I have the port group configured as SCCP - Maybe I should be using SIP instead?
No Audio:
voice service voip
sip
midcall-signaling passthru media-change
early-offer forced
no call service stop
registration passthrough
Dec 8 21:07:35.842: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationCisco-Guid: 3244526336-0000065536-0000003600-0503709706Session-Expires: 1800Diversion: ;reason=follow-me;privacy=off;screen=yesP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: ;isFocusMax-Forwards: 70Content-Length: 0Dec 8 21:07:35.850: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 3244526336-0000065536-0000003600-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITETimestamp: 1386536855Contact: Expires: 180Allow-Events: telephone-eventMax-Forwards: 69Diversion: ;privacy=off;reason=follow-me;screen=yesContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 7885 9952 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29558 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:35.850: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:35.858: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Dec 8 21:07:41.986: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120698 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:41.986: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Length: 0Dec 8 21:07:41.990: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 183 Session ProgressVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:41.990: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 180 RingingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Server: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:43.938: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274BE17From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Call-ID: [email protected]: 101 INVITETimestamp: 1386536855Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120699 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:43.938: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1b7e642f53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Require: timerSession-Expires: 1800;refresher=uacSupported: timerContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:43.942: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274C4F2From: "Steve Dainard" ;tag=47582CDC-2165To: ;tag=as1502e8b4Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:07:43.958: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1d288c5e53From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 218v=0o=CiscoSystemsCCM-SIP 40265 1 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.6.30t=0 0m=audio 4000 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=inactivea=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:07:44.158: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: UPDATE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1f50a2749fFrom: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:35 GMTCall-ID: [email protected]: Cisco-CUCM9.1Max-Forwards: 70Supported: timer,resource-priority,replacesAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 102 UPDATESupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:07:44.158: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b1f50a2749fFrom: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 102 UPDATEAllow-Events: telephone-eventContact: Supported: timerContent-Length: 0Dec 8 21:07:44.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 103 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationSession-Expires: 1800;refresher=uacP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:07:44.162: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 103 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:44.162: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2028eab237From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 103 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Require: timerSession-Expires: 1800;refresher=uacSupported: timerContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 3614 6206 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29556 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:44.214: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2151dd5c40From: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249To: ;tag=475844D8-1CC2Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: 70CSeq: 103 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40265 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 25834 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:07:52.590: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: CSeq: 102 INVITEUser-Agent: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 253v=0o=root 959120698 959120700 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 40818 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:07:52.594: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 102 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:07:52.594: //13815/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK76uf9710bomh2kk6c350.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 102 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 7885 9953 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29558 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:07:52.610: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK8m4hbg10c8ag4kg723g0.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: CSeq: 102 ACKUser-Agent: Rogers SIP CoreContent-Length: 0Dec 8 21:07:52.630: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: BYE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK96bidp10cou05l89e611.1Max-Forwards: 68From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Call-ID: [email protected]: 103 BYEUser-Agent: Rogers SIP CoreX-RBS-SIP-HangupCause: Normal ClearingX-RBS-SIP-HangupCauseCode: 16Content-Length: 0Dec 8 21:07:52.634: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Sent: BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK274D1192From: ;tag=475844D8-1CC2To: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249Date: Sun, 08 Dec 2013 21:07:44 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2Max-Forwards: 70Timestamp: 1386536872CSeq: 101 BYEReason: Q.850;cause=16P-RTP-Stat: PS=0,OS=0,PR=420,OR=67200,PL=0,JI=0,LA=0,DU=8Content-Length: 0Dec 8 21:07:52.634: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK96bidp10cou05l89e611.1From: ;tag=as1502e8b4To: "Steve Dainard" ;tag=47582CDC-2165Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 103 BYEReason: Q.850;cause=16P-RTP-Stat: PS=2,OS=320,PR=100,OR=16000,PL=0,JI=0,LA=0,DU=8Content-Length: 0Dec 8 21:07:52.646: //13814/C1638B000000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK274D1192From: ;tag=475844D8-1CC2To: "Steve Dainard" ;tag=40265~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727249Date: Sun, 08 Dec 2013 21:07:52 GMTCall-ID: [email protected]: 101 BYEContent-Length: 0
bi-direcitonal audio:
voice service voip
sip
early-offer forced
midcall-signaling passthru
no call service stop
registration passthrough
Dec 8 21:09:44.331: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationCisco-Guid: 0239559040-0000065536-0000003601-0503709706Session-Expires: 1800Diversion: ;reason=follow-me;privacy=off;screen=yesP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: ;isFocusMax-Forwards: 70Content-Length: 0Dec 8 21:09:44.339: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITETimestamp: 1386536984Contact: Expires: 180Allow-Events: telephone-eventMax-Forwards: 69Diversion: ;privacy=off;reason=follow-me;screen=yesSession-Expires: 1800Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2507 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:44.339: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:44.347: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: Call-ID: [email protected]: 101 INVITETimestamp: 1386536984Dec 8 21:09:52.535: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674502 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:52.539: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 183 Session ProgressVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:53.007: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 180 RingingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Length: 0Dec 8 21:09:53.007: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 180 RingingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Server: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:54.967: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274E8E6From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 101 INVITETimestamp: 1386536984Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674503 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:54.967: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b267445467fFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:54.967: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK274F1060From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:09:54.979: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2876c28ab9From: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 218v=0o=CiscoSystemsCCM-SIP 40276 1 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.6.30t=0 0m=audio 4000 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=inactivea=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:09:55.007: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: UPDATE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2ada930ebFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:44 GMTCall-ID: [email protected]: Cisco-CUCM9.1Max-Forwards: 70Supported: timer,resource-priority,replacesAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 102 UPDATESupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:09:55.011: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2ada930ebFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 102 UPDATEAllow-Events: telephone-eventContact: Supported: timerContent-Length: 0Dec 8 21:09:55.011: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM9.1Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 103 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceSupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=calling;screen=yes;privacy=offContact: Content-Length: 0Dec 8 21:09:55.011: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 102 INVITEMax-Forwards: 70Timestamp: 1386536995Contact: Diversion: ;privacy=off;reason=follow-me;screen=yesExpires: 180Allow-Events: telephone-eventContent-Length: 0Dec 8 21:09:55.011: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 103 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:09:55.019: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 102 INVITETimestamp: 1386536995Dec 8 21:09:55.031: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK275014C3From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beCall-ID: [email protected]: 102 INVITETimestamp: 1386536995Contact: Server: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 279v=0o=root 1961674502 1961674504 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:09:55.035: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2be82180dFrom: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 103 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:09:55.175: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.6.30:5060;branch=z9hG4bK6b2c8e71c96From: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255To: ;tag=475A42CC-1387Date: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 70CSeq: 103 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40276 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 22546 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:09:55.179: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK2751A88From: "Steve Dainard" ;tag=475A22C4-26C5To: ;tag=as314346beDate: Sun, 08 Dec 2013 21:09:55 GMTCall-ID: [email protected]: 70CSeq: 102 ACKAllow-Events: telephone-eventContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2508 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.267: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: CSeq: 102 INVITEUser-Agent: Rogers SIP CoreAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replacesContent-Type: application/sdpContent-Length: 255v=0o=root 1961674502 1961674505 IN IP4 173.46.30.202s=Rogers SIPc=IN IP4 173.46.30.202t=0 0m=audio 37982 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvDec 8 21:10:05.271: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 100rel,timer,resource-priority,replaces,sdp-anatMin-SE: 1800Cisco-Guid: 0239559040-0000065536-0000003601-0503709706User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERCSeq: 101 INVITEMax-Forwards: 70Timestamp: 1386537005Contact: Expires: 180Allow-Events: telephone-eventContent-Type: application/sdpContent-Length: 241v=0o=CiscoSystemsSIP-GW-UserAgent 7438 7415 IN IP4 10.0.250.4s=SIP Callc=IN IP4 10.0.250.4t=0 0m=audio 29560 RTP/AVP 0 101c=IN IP4 10.0.250.4a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.271: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 100 TryingVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.4.M2Content-Length: 0Dec 8 21:10:05.275: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 100 TryingVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: presenceContent-Length: 0Dec 8 21:10:05.275: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27523EAFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYAllow-Events: presenceSupported: replacesSupported: X-cisco-srtp-fallbackSupported: GeolocationP-Asserted-Identity: "Steve Dainard" Remote-Party-ID: "Steve Dainard" ;party=called;screen=yes;privacy=offContact: Content-Type: application/sdpContent-Length: 232v=0o=CiscoSystemsCCM-SIP 40276 3 IN IP4 10.0.6.30s=SIP Callc=IN IP4 10.0.250.93b=TIAS:64000b=AS:64t=0 0m=audio 22546 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15Dec 8 21:10:05.279: //13819/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK3tuoo01070ag0lg7o5k0.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventContact: Supported: replacesSupported: sdp-anatServer: Cisco-SIPGateway/IOS-15.2.4.M2Supported: timerContent-Type: application/sdpContent-Length: 238v=0o=CiscoSystemsSIP-GW-UserAgent 4519 2508 IN IP4 10.0.1.67s=SIP Callc=IN IP4 10.0.1.67t=0 0m=audio 29562 RTP/AVP 0 101c=IN IP4 10.0.1.67a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20Dec 8 21:10:05.279: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: ACK sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK27531E8DFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 70CSeq: 101 ACKAllow-Events: telephone-eventContent-Length: 0Dec 8 21:10:05.295: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: ACK sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4d5qq910785h6ks9f3c0.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: CSeq: 102 ACKUser-Agent: Rogers SIP CoreContent-Length: 0Dec 8 21:10:05.295: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received: BYE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4tbtsi10785h6jcp71g1.1Max-Forwards: 68From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Call-ID: [email protected]: 103 BYEUser-Agent: Rogers SIP CoreX-RBS-SIP-HangupCause: Normal ClearingX-RBS-SIP-HangupCauseCode: 16Content-Length: 0Dec 8 21:10:05.295: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Sent: BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK275469CFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2Max-Forwards: 70Timestamp: 1386537005CSeq: 102 BYEReason: Q.850;cause=16P-RTP-Stat: PS=511,OS=81760,PR=505,OR=80800,PL=0,JI=0,LA=0,DU=10Content-Length: 0Dec 8 21:10:05.299: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 200 OKVia: SIP/2.0/UDP 173.46.30.202:5060;branch=z9hG4bK4tbtsi10785h6jcp71g1.1From: ;tag=as314346beTo: "Steve Dainard" ;tag=475A22C4-26C5Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: Cisco-SIPGateway/IOS-15.2.4.M2CSeq: 103 BYEReason: Q.850;cause=16P-RTP-Stat: PS=505,OS=80800,PR=634,OR=101440,PL=0,JI=0,LA=0,DU=10Content-Length: 0Dec 8 21:10:05.303: //13818/0E4761800000/SIP/Msg/ccsipDisplayMsg:Received: SIP/2.0 200 OKVia: SIP/2.0/TCP 10.0.250.4:5060;branch=z9hG4bK275469CFrom: ;tag=475A42CC-1387To: "Steve Dainard" ;tag=40276~d732e07f-799a-4d2b-9d6a-ae2aaf54507d-19727255Date: Sun, 08 Dec 2013 21:10:05 GMTCall-ID: [email protected]: 102 BYEContent-Length: 0 -
Incoming Phone Call DND (Do Not Disturb) or ability to send call to voicema
Does anyone know a good way to put a DND (do not disturb) on incoming phone calls? The idea is to send calls directly to voice mail while still leaving the phone on to receive email & text messages. While at work or in meetings, it would be nice to not be bothered by calls especially from specific people. Ideally it would be really nice to one or all of the following:
-Function to send all calls to voice mail without ringing
-An option when an incoming call is coming in to send the person to voicemail with the choice of a prerecorded message such as "In a meeting" or "At work" etc.
-Send specific people to voice mail without ringing, possibly with a prechosen message.The only option for the iPhone not ringing when a call is received and the same when a text message is received while in a meeting is by turning the silent switch on located above the volume buttons.
-
Username and password for SOAP sender call
Hello,
does anyone know how to provide the username and password to a SOAP sender call, e.g. XI receives the ws via soap and needs to know which userid and password to check. When you use a SOAP client they use basic authentication which sends the request first to XI and XI send a request back for password. This would work for an online app but not for ws from machine to machine. I read some docu about query strings but no where it has an example what to put either on the request URL, the adapter or the SOAP envelope. SOAP 1.1 seems to have left that open and IBM has an example using SOAP Header which did not work with XI.
Thanks
StefanHi,
do you use the javax.xml.rpc.Call class? Because then
you can supply username and password to the call via
the addParameter method. I think I did that with XI 3.0
and it worked. If you need more information please consult
the javax.xml.rpc.Call javadoc.
Best regards,
Hermann -
FaceTime connects ok and rings out when sending call but no ringing sound when receiving incoming calls
Using FaceTime http://support.apple.com/kb/ht4319
Troubleshooting FaceTime http://support.apple.com/kb/TS3367
The Complete Guide to FaceTime + iMessage: Setup, Use, and Troubleshooting
http://tinyurl.com/a7odey8
Troubleshooting FaceTime and iMessage activation
http://support.apple.com/kb/TS4268
Using FaceTime and iMessage behind a firewall
http://support.apple.com/kb/HT4245
iOS: About Messages
http://support.apple.com/kb/HT3529
Set up iMessage
http://www.apple.com/ca/ios/messages/
Troubleshooting Messages
http://support.apple.com/kb/TS2755
Setting Up Multiple iOS Devices for iMessage and Facetime
http://macmost.com/setting-up-multiple-ios-devices-for-messages-and-facetime.htm l
FaceTime and iMessage not accepting Apple ID password
http://www.ilounge.com/index.php/articles/comments/facetime-and-imessage-not-acc epting-apple-id-password/
Unable to use FaceTime and iMessage with my apple ID
https://discussions.apple.com/thread/4649373?tstart=90
My Facetime Doesn't Ring
https://discussions.apple.com/message/19087457
For non-Apple devices, check out the TextFree app https://itunes.apple.com/us/app/text-free-textfree-sms-real/id399355755?mt=8
How to Send SMS from iPad
http://www.iskysoft.com/apple-ipad/send-sms-from-ipad.html
Cheers, Tom -
When I call someone, people on the other end can't see my number. I checked the settings where you get 3 options. Send caller-ID, don't send caller-ID or let the network decide. Obviously "don't send" won't send it, but the other 2 options don't work either.
If I select "yes" the other end still sees "private number".
Is there any other option I'm overlooking?
Solved!
Go to Solution.yes call your operator and ask them to turn on caller id and always set the settings to yes anyway
If i have helped at all a click on the white star below would be nice thanks.
Now using the Lumia 1520 -
Skype sending calls on its own
Starting with my change from a note 2 to a note 4, Skype has begun randomly sending calls to my contacts. When I caught it doing this there was a faint buzz/ring much quieter than normal ringing and more the volume of things such as the keyboard tap noise.
The call attempts last about as long as a normal call attempt but then are marked as failed.
It does not bring up the 'call' screen it's all entirely in the background and I only know it attempted a call afterwards when it shows a 'call failed' message on that person's text history.
Since I only use Skype for texting is there a way to disable calling? Or any ideas what's causing it to attempt to call people?
I have reinstalled the app, changed my password, and disabled non-contact calling and messaging already.
Any help would be appreciated as the random calls are really irking my friends. I know it's nit me bumping the button during messaging because that brings up the call screen as normal.I too have experienced this. It is downright scary. It happened while I was on another call using the actual phone. The call ended screen came up and the log showed call failed.
-
Timing of NetStream.send() calls
Hello,
we use NetStream.send() calls to remote control clients
subscribed to a live stream. We observed that the live video &
audio stream can get out of sync with the remote calls sent via
NetStream.send(). In bad cases the remote calls where executed 4
seconds before the audio/video signal was shown.
My idea is that NetStream.send() events are created when the
stream "arrives" at the Flash Player, while the Audio/Video Data
might be buffered. Can anybody confirm this implementation detail
of FMS 3.0?
Any help on this is greatly appreaciated,
JuergenThanks for reporting. I can reproduce the bug in house. We will investigate.
Calise -
Problem sending calls from iPhone 4
My iPhone 4 does not receive calls from my fixed home line - but sending calls anywhere is apparently no problem - have been through complete Apple procedure for resolving problem (reinitialisation etc) but without success - any ideas out there ?
Thanks for responding so quickly !
In response to your queries:
- I have not yet consulted the carrier as yet (guess that is my next action tomorrow !)
- The fixed phone only seems to have a problem in calling my cellphone and no other !
- I have not tried to call my cellphone yet from other fixed phones
- My cellphone receives calls from other cellphones
Hope this gives you a lead ! -
4FXS-DID configuration problem using BRI to receive/send calls from/to PSTN
Hi to all.
Follow is the partial configuration of my CME.
It has a 4FXS card to use analog phones and faxes, and 4 ISDN Basic Rate interfaces.
On each port of the 4FXS DID card happens the following:
If I enable direct-inward-dial
I can receive calls from PSTN.
Off hooking the analog phone I cannot hear any free line tone. I cannot make any call from that phone.
If I disable direct-inward-dial
I cannot receive directly calls form PSTN.
Callers on PSTN after typing my number, they hear a free tone. Then just typing the extension desired the phone ring.
Off hooking the analog phone I hear a free line tone. I can make normal calls to everyone.
I have half the configuration working.
Where am I wrong?
I really appreciate your help.
Thanks to all.
Giorgio.
CME#sh ver
Cisco IOS Software, 2800 Software (C2800NM-IPVOICEK9-M), Version 12.4(24)T3, RELEASE SOFTWARE (fc2)
ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)
CME uptime is 2 days, 15 hours, 49 minutes
System returned to ROM by power-on
System restarted at 18:54:45 CDT Fri Aug 2 2013
System image file is "flash:/c2800nm-ipvoicek9-mz.124-24.T3.bin"
Cisco 2811 (revision 53.50) with 249856K/12288K bytes of memory.
Processor board ID FTX1120A08L
2 FastEthernet interfaces
4 ISDN Basic Rate interfaces
16 terminal lines
4 Voice FXS interfaces
DRAM configuration is 64 bits wide with parity enabled.
239K bytes of non-volatile configuration memory.
1948656K bytes of USB Flash usbflash0 (Read/Write)
497448K bytes of ATA CompactFlash (Read/Write)
Configuration register is 0x2102
CME#sh telephony-service
CONFIG (Version=7.1)
=====================
Version 7.1
Cisco Unified Communications Manager Express
isdn switch-type basic-net3
voice translation-rule 1
rule 1 /^2929091\(..\)/ /\1/
rule 2 /^2929091\(.\)/ /\1/
rule 3 /^02929091\(..\)/ /\1/
rule 4 /^02929091\(.\)/ /\1/
voice translation-rule 2
rule 2 /\(.*\)/ /02929091\1/
voice translation-rule 10
rule 1 /\(^......$\)/ /0\1/ type national national plan isdn isdn
rule 8 /\(^......$\)/ /00\1/ type international international plan isdn isdn
voice translation-profile PSTN-IN
translate calling 10
translate called 1
voice translation-profile PSTN-OUT
translate calling 2
interface BRI0/0/0
description Isdn channels 1 & 2
no ip address
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn static-tei 0
interface BRI0/0/1
description Isdn channels 3 & 4
no ip address
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn static-tei 0
interface BRI0/1/0
description Isdn channels 5 & 6
no ip address
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn static-tei 0
interface BRI0/1/1
description not used
no ip address
shutdown
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn static-tei 0
voice-port 0/0/0
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
compand-type a-law
description Connessione con CO Telecom channels 1 & 2
voice-port 0/0/1
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
compand-type a-law
description Connessione con CO Telecom channels 3 & 4
voice-port 0/1/0
translation-profile incoming PSTN-IN
translation-profile outgoing PSTN-OUT
compand-type a-law
description Connessione con CO Telecom channels 5 & 6
voice-port 0/1/1
description Disponibile
voice-port 0/2/0
input gain 14
connection plar 83
description Interphone 40
station-id name Citofono
station-id number 40
caller-id enable
voice-port 0/2/1
cptone IT
description Ced 35
station-id name CED
station-id number 35
caller-id enable
voice-port 0/2/2
cptone IT
description FAX_1 50
station-id name FAX_1
station-id number 50
caller-id enable
voice-port 0/2/3
cptone IT
description FAX_2 60
station-id name FAX_1
station-id number 60
caller-id enable
dial-peer voice 1001 pots
description **Sends call to PSTN line 1-2**
destination-pattern 0T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/0
dial-peer voice 2001 pots
description **Receives calls coming from PSTN line 1-2**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/0/0
dial-peer voice 1003 pots
description **Sends call to PSTN line 3-4**
destination-pattern 0T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0/1
dial-peer voice 2003 pots
description **Receives calls coming from PSTN line 3-4**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/0/1
dial-peer voice 1005 pots
description **Sends call to PSTN line 5-6**
destination-pattern 0T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/1/0
dial-peer voice 2005 pots
description **Receives calls coming from PSTN line 5-6**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/1/0
dial-peer voice 2035 pots
description **Receives calls coming from PSTN to Extension 35**
answer-address 35
destination-pattern 35$
incoming called-number 35
port 0/2/1
dial-peer voice 2040 pots
description **Receives calls coming from PSTN to Extension 40**
destination-pattern 40
incoming called-number 40
no digit-strip
direct-inward-dial
port 0/2/0
dial-peer voice 2050 pots
description **Receives calls coming from PSTN to Fax 50**
destination-pattern 50
incoming called-number 50
no digit-strip
direct-inward-dial
port 0/2/2
dial-peer voice 2060 pots
description **Receives calls coming from PSTN to Fax 60**
destination-pattern 60
incoming called-number 60
no digit-strip
direct-inward-dial
port 0/2/3
CME#Hi Paolo.
Sorry for the delay. I was on holiday.
I would like to keep separate incoming calls from outgoing calls.
So I decided to keep two dial peers for every BRI interface.
I followed your suggestion on eliminate commands:
"incoming called-number" on FXSs,
various "progress_ind" on BRIs
Also I eliminated "direct-inward-dial" on FXSs,
Today I reconfigured and right tested the dial peers as following:
! Bri Interfaces
dial-peer voice 1001 pots
description **Sends call to PSTN line 1-2**
destination-pattern 0T
port 0/0/0
dial-peer voice 2001 pots
description **Receives calls coming from PSTN line 1-2**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/0/0
dial-peer voice 1003 pots
description **Sends call to PSTN line 3-4**
destination-pattern 0T
port 0/0/1
dial-peer voice 2003 pots
description **Receives calls coming from PSTN line 3-4**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/0/1
dial-peer voice 1005 pots
description **Sends call to PSTN line 5-6**
destination-pattern 0T
port 0/1/0
dial-peer voice 2005 pots
description **Receives calls coming from PSTN line 5-6**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/1/0
! FXS interfaces
dial-peer voice 2035 pots
description **Receives calls coming from PSTN to Extension 35**
destination-pattern 35
no digit-strip
port 0/2/1
dial-peer voice 2040 pots
description **Receives calls coming from PSTN to Extension 40**
destination-pattern 40
no digit-strip
port 0/2/0
dial-peer voice 2050 pots
description **Receives calls coming from PSTN to Fax 50**
destination-pattern 50
no digit-strip
port 0/2/2
dial-peer voice 2060 pots
description **Receives calls coming from PSTN to Fax 60**
destination-pattern 60
no digit-strip
port 0/2/3
Thanks a lot Paolo. :-)
Giorgio. -
Recently I bought Iphone 4s. I'm having trouble receiving and sending calls and messages, in the Red Status appears INVALID SIM or without Signal... I already restored the original settings, changed sim card and still got nothing fixed... what else can I do? Is it possible to change my Iphone 4s.
I saw in yootube that is very common problem now. I paid for Iphone 4s 749 $ and it has such king of problems, I am very disappointed. Can you advice me how Can replace for new one.
-
Droid X receives and sends calls but neither party can hear
My Droid X would receive and send calls but nether party can hear each other, however I can send dial tones on the line. The receiving side can not send tones. Resetting to factory is dramatic. What can I do to fix this. Thanks!
Sounds like the software could be corrupt, how is hard reset dramatic?
It is either hardware or software, you won't know until you eliminate one. -
Send calls from overflow queue to Normal queue and prioritize call
I have 2 queues in my script: queue1(normal queue) queue2(overflow queue)
I am sending calls to the queue2 if the call threshold reaches 3 calls or 3 minutes in the queue1.
While the call is in the queue2, queue 2 checks queue 1 to see if there are any Ready agents in the queue1.
If it sees Ready agents in the queue1 send the call back to the queue1, when this call is in the queue1,
I want this call to be 1st position in the first queue even though if there are 10-15 people waiting in the queue1.
What is the best way to script this scenario?
I am testing this in the lab but still not successful.
Any help would appreciated.Hi,
Set call priority.
Regards,
Muhammad Amir Raza -
UCCX Scripting:Not able to tranfer calls to Operator.
HI Experts.
I am trying to make a script like
thank you for calling XYZ Company if you know the extension please dial it now or press 0 for assitance.
I have created a script in Cisco Unfied CCX 7.1 Editor and i am able to send the calls to Users Ext's but i am not able to transfer the calls to operator when i pressed 0.
I have attached the script where i am able to transfer the calls to Users ext successfully but i also want to transfer calls to operator when any body from outside press 0.
Please review my attached script and let me know how i can do it to complete the rest.I have attached My script and also the snapshot.
Waiting for reply
thanks and best regardsAdd an integer varible named intCount and set it to 0. I've added some lines and bolded them for you. The first time the script asks for a number, intCount will be 0, the second time it will be 1, so the test will be false the second time and terminate the call. I don't know what type of buisness you are in, but you may want to record a message telling them they are being disconnected and place it right before the terminate statement.
Start
Delay 2 sec
Accept (--Triggering Contact--)
startscript
PhoneNumber = get digit string (--Triggering Contact--)
Successful
Call Redirect (--Trigger Contact-- to PhoneNumber)
Successful
goto wrapup
Busy
goto startscript
Invalid
goto startscript
Unsuccessful
goto startscript
Timeout
Unsuccessful
If (PhoneNumber == "0") Then
True
Set PhoneNumber = "7263"
Call Redirect (--Trigger Contact-- to PhoneNumber)
Sucessful
goto wrapup
Busy
goto startscript
Invalid
goto startscript
Unsuccessful
goto startscript
False
Set intCount = intCount + 1
if (intCount > 1) Then
True
goto prewrapup
False
goto startscript
prewrapup
Terminate (--Triggering Contact--)
wrapup
end
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