Blocking inbound call in CM with MGCP Voice gateways

I'm sure someone has posted on this: We have MGCP voicegateways and CM 4.02aSR2b. How can I setup CM to block a specific inbound call? If the gateways were H323 then no problem, but with MGCP we are stumped. Any suggestions? Thx!

It cannot be done with MGCP. It can be done with H.323 using reject translation rules on the GW.
With MGCP your only option is to route all calls via Unity or IPCC first where logic can be build to block calls.
HTH, please rate posts!
Chris

Similar Messages

Block Inbound Calls

Most of us are probably aware of the limited ability, via H323 gateways and translation rules, to block inbound calls from the PSTN.
Two questions:
1.) Is there another method I may not be aware of?
2.) I was reading about External Call Control Profiles this morning and it seems like a logical fit. Anyone experimented with this?
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_0_1/ccmfeat/fsextcallctrl.html

Here is great Doc from CCO that outline Voice Translation Profiles and has examples of
various situations. Letting the GW handle it to me seems most straight forward if its H323 GW
Why make CUCM have to even deal with it?
Has example for blocking ext
http://www.cisco.com/en/US/customer/tech/tk652/tk90/technologies_configuration_example09186a00803f818a.shtml
HTH, if it does not answer what your after pls give me some more details on what you want to accomplishand I will see if I can help
George

Call Manager register fxs port with voice gateway- problem

I have a CUCM 6 and a Voice Gateway V224. I've configured the voice gateway's voice FXS ports as MGCP.
I have a Voip connected and registered to the CUCM and a Pots phone connected to the Voice Gateway.
If i dial from the Voip to the Pots phone it rings. The problem is that i cannot ring from the Pots to the Voip phone.
I have no dial tone.
If i write no shut down on the voice port i have a tone. If i configure mgcp on the voice port i have a busy ringtone.
I've entered no mgcp and mgcp commands and i've reset the voice gateway.
How can i call from the pots to the voip phone?
The ios version on the voice gateway is Version 12.4(22)T4.
Here is an outghtput from the Voice gateway.
ccm-manager mgcp
ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 10.1.1.33
ccm-manager config
mgcp
mgcp call-agent CCMIOSS 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp validate domain-name
mgcp rtp payload-type g726r16 static
mgcp profile default
timeout tone busy 600
timeout tone dial 600
dial-peer voice 999223 pots
service mgcpapp
port 2/23
dial-peer voice 999222 pots
service mgcpapp
port 2/22
dial-peer voice 999888 pots
service mgcpapp
port 2/23
The CUCM 6 is registered with the voice gateway.

Is your campaign using CPA? If so, what's the behavior if CPA is not enabled?
I think the best thing to do is to run a trace...
Call Manager > Cisco Unified Serviceability > Trace > Configurations
Select a CUCM server - any subscriber would work.
Service Group - CM Services
Cisco CallManager (Inactive)
Enable SIP Stack Trace and apply to all nodes. Download and install RTMT
Make a bunch of outbound tests and then open RTMT > Trace & Log Central > Collect Files > Check "All Servers" for Cisco CallManager > Next > Next > Relative Range if you made the test calls within the last X minutes, otherwise you can set a From and To datetime. Click Finish and go through your SDL logs and see what errors you find and post them here.
Also, make sure your phone is in the correct CSS in Call Manager

Cisco 2911 Voice Gateway SIP PSTN Calls Fail

Hello All,
    I am having trouble with outboud SIP PSTN calls through a Cisco 2911 Voice Gateway. 2911 VG terminates PSTN SIP Traffic and connects to Avaya CS1000M via QSIG PRI Trunks. When calls are attempted outbound fron the PBX the caller gets a fast busy. Debug ISDN q931 shows the call hitting the 2911 properly, debug voip ccapi inout shows the call matching the correct dial peers and debug ccsip shows the invite to the PSTN Provider SBC, however within the invite the "from" address incorrectly shows the calling number with the provider SBC address (see below). does anyone have any insight on how to correct this? Attached are VG config and Debug isdn q931, voip ccapi inout, ccsip messages and ccsip call. Thanks in advance for any help!!
From: <sip:[email protected]>:tag=6166CDC4-882
To: <sip:[email protected]>
Shawn C. Smith

i have same problem my cucm ip is 192.168.200.53
my Voice Gateway is SIP by ip 192.168.200.86 for internal
and 172.29.7.94
and my SIP Server is 10.208.9.69
if its oky can yuo take a look at my problem please
this is the syslog from debug
May 30 20:19:34.284: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.200.53:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
Session-Expires: 1800
P-Asserted-Identity: "Aysar Mohamed" <sip:[email protected]>
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
May 30 20:19:34.284: //-1/B5A494800000/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=2217156
   ----- ccCallInfo IE subfields -----
   cisco-ani=2217156
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=90555769123
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
May 30 20:19:34.288: //-1/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x30CF41D4, Call Info(
   Calling Number=2217156,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=90555769123(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=465
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 1
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832953048,feature_id:85
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=90555769123(TON=Unknown, NPI=Unknown))
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
   Event=0x2B82D890
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 90555769123
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetContext:
   Context=0x2ABC2E44
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 465 with tag 0 to app "_ManagedAppProcess_Default"
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=802, Params=0x2ABC19D4, Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
   In: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
   Out: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Destination Pattern=9T, Called Number=0555769123, Digit Strip=FALSE
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0555769123(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=Aysar Mohamed
   Account Number=2217156, Final Destination Flag=TRUE,
   Guid=B5A49480-0001-0000-0000-11C535C8A8C0, Outgoing Dial-peer=802
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=2217156
   ----- ccCallInfo IE subfields -----
   cisco-ani=2217156
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=0555769123
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x30CF41D4, Interface Type=3, Destination=, Mode=0x0,
   Call Params(Calling Number=2217156,(Calling Name=Aysar Mohamed)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0555769123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=802, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 2
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832952824,feature_id:86
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccCallSetContext:
   Context=0x2ABC1984
May 30 20:19:34.292: //465/B5A494800000/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=802
May 30 20:19:34.292: //466/B5A494800000/CCAPI/cc_api_call_proceeding:
   Interface=0x30CF41D4, Progress Indication=NULL(0)
May 30 20:19:34.292: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
May 30 20:19:34.292: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1401481174
Contact: <sip:[email protected]:5060>
Call-Info: <sip:172.29.7.94:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
May 30 20:19:34.300: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
CSeq: 101 INVITE
Content-Length: 0
May 30 20:19:34.612: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Contact: <sip:[email protected]:5060;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Content-Length: 328
Content-Type: application/sdp
v=0
o=- 17192647 17192647 IN IP4 10.208.9.69
s=SBC call
c=IN IP4 10.208.9.69
t=0 0
m=audio 39910 RTP/AVP 8 0 102 102 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 AMR/8000
a=rtpmap:102 AMR/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=fmtp:116 0-15
a=fmtp:18 annexb=yes
May 30 20:19:34.612: %SIP-3-UNSUPPORTED: Unsupported ptime value
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=466,
   Caps(Codec=0x2, Fax Rate=0x2, Vad=0x1,
   Modem=0x0, Codec Bytes=160, Signal Type=2)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event=170, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event=98, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
   Interface=0x30CF41D4, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),
   Cause Value=0
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
   Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
   Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0
   Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
   Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccConferenceCreate:
   (confID=0x30C11410, callID1=0x1D1, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
   (confID=0x30C11410, callID2=0x1D2, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   Conference Id=0x30C11410, Call Id1=465, Call Id2=466, Tag=0x0
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
May 30 20:19:34.616: ccConferenceCreate: ret1=0, codecMask1=2, bytes1=160, negot1=0, dtmf1=0
                    ret2=0, codecMask2=2, bytes2=160, negot2=1, dtmf2=6,
                    tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
                    tx_dynamic_pt2=8, rx_dynamic_pt2=8, codec_mode2=0, params_bitmap2 =0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   delay media to slow start case, codec negotation is not done
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_api_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
   Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/B5A494800000/CCAPI/cc_api_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x16, Destination Call Id=466)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x16, Destination Call Id=465)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_process_notify_bridge_done:
   Conference Id=0x16, Call Id1=465, Call Id2=466
May 30 20:19:34.616: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 233
v=0
o=CiscoSystemsSIP-GW-UserAgent 2639 5276 IN IP4 192.168.200.86
s=SIP Call
c=IN IP4 192.168.200.86
t=0 0
m=audio 18288 RTP/AVP 8 0 19
c=IN IP4 192.168.200.86
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Reason: Q.850;cause=127;text="interworking unspecified"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
   Cause Value=41, Interface=0x30CF41D4, Call Id=466
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccCallReleaseResources:
   release reserved xcoding resource.
May 30 20:19:34.680: //466/B5A494800000/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=466
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccConferenceDestroy:
   Conference Id=0x16, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
   Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=466,
   Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: :cc_free_feature_vsa freeing 31A5D9F0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: vsacount in free is 1
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=41
Content-Length: 0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=465,
   Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: :cc_free_feature_vsa freeing 31A5DAD0
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: vsacount in free is 0
May 30 20:19:36.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.7.94:5060 SIP/2.0
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
May 30 20:19:36.048: //467/8DAABF6C8278/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>;tag=739BBC-1CE2
Date: Fri, 30 May 2014 20:19:36 GMT
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 446
v=0
o=CiscoSystemsSIP-GW-UserAgent 3496 1601 IN IP4 172.29.7.94
s=SIP Call
c=IN IP4 172.29.7.94
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.7.94
m=image 0 udptl t38
c=IN IP4 172.29.7.94
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
My SIP GW internal ip address is 192.168.200.86
and the Public IP is : 172.29.7.94
My CUCM is 192.168.200.53
my GW Config is :
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 3
rule 1 /^9/ /\1/
voice translation-rule 4
rule 4 /^22217/ /7/
rule 5 /^2217/ /7/
rule 6 /^022217/ /7/
rule 7 /^0122217/ /7/
voice translation-rule 5
rule 1 /^5/ /905/
rule 2 /^1/ /901/
rule 3 /^2/ /902/
rule 4 /^3/ /903/
rule 5 /^4/ /904/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 10 /^00/ /900/
rule 11 /'+'/ /900/
voice translation-profile OUT
translate called 3
voice translation-profile REDIAL
translate calling 5
voice translation-profile SIP-NEW
translate called 4
application
service mva http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
service ccm http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
license udi pid CISCO2921/K9 sn FCZ164960G0
hw-module pvdm 0/0
hw-module pvdm 0/1
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.200.86 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
ip address 172.29.7.94 255.255.255.252
duplex auto
speed auto
ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip route 0.0.0.0 0.0.0.0 192.168.200.1
ip route 10.208.9.0 255.255.255.0 172.29.7.93
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 192.168.200.53 identifier 1 priority 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register NAGHI-MTP
dspfarm profile 2 mtp
codec g711alaw
maximum sessions hardware 25
associate application SCCP
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern 9T
session protocol sipv2
session target ipv4:10.208.9.69:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay sip-notify rtp-nte sip-kpml
no vad
dial-peer voice 811 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 812 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 813 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 814 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 815 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 816 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 817 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 818 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
Please i need ur help ASAP

Video Calls on Voice gateway enabled router

Hi All,
I am having cisco 3925 router which is acting as MGCP voice gateway for PSTN calls,
Is it possible to use same MGCP voice gateway router to do the video calls .
thanx in advance
Regards
Deepak

No, we can only use H.323 / SIP for video calls across ISDN.
H.323 - H.320
http://www.cisco.com/en/US/partner/docs/video/milticomm/h320Bonding.html
SIP - H.320
http://www.cisco.com/en/US/partner/docs/ios/voice/sip/configuration/guide/sip_cg-rsvp_ps10592_TSD_Products_Configuration_Guide_Chapter.html
Arun

Voice Gateway & Gatekeeper On a single Router?

Hello All,
Has anyone had experience with putting voice gateway and voice gatekeeper functionality into one router? We will have three clusters initially (with about 1,000 users worldwide though) and was considering one gatekeeper per cluster for CAC and tail end hop off routing for LD/International calls.
Any guidance on what sort of platform these should be running on if kept separate? DSP resources in them, etc.?
The voice gateways will be a mixture of 2851 and 3800 series routers. Thanks in advance for the help.
Thanks,
Dave

Hi,
I have done this many times in lab, dont know if that will be ok in production if you have a good powerfull problem i dont think that it will be a problem.
About the DSP's maybe you can use the DSP calculator - anyway this refers to many parameters on the network.
BR,
Teo

7920 Unable to answer with two inbound calls ringing

I have a CME customer reporting that when their 7920s have two inbound calls ringing at the same time they are unable to answer either call.
The 7920s are a shared extension, on Dual Line DNs, with their desktop phones.
CME is 4.0(2), and the 7920 phone load is 4.0-03-00.
Has anyone else experienced this problem, or have any idea what might cause it?
Thanks,
Mike

What type of ap are you using? Controller or autonomous?
If controller is pre-4.0.217.0 version then would want to set the 802.1p tag for the platinum QoS profile to 7 vs 6 or upgrade to 4.0.217.0.
Also would advise to upgrade the 7920 to 03-02.
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser-crypto
You may be hitting CSCsi02523?
Release notes @ http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/wip7920/relnotes/rn302.htm#wp98206.

Does anyone configure cisco router with MGCP to link Call agent Clarent ?

hi,
We require to configure As5300 with MGCP to link Clarent call agent. Does anyone have cisco router configuration ?
thanks.
best regards.
fred.

Below is the sample configuration for the 5300 to Call-Agent. This is also dependant on which package is configured on the call-agent so we can configure it accordingly. Hope this helps.
version 12.3
no service pad
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
hostname AS5300-5
boot system tftp c5300-is-mz.123-2.T1 171.68.191.135
logging buffered 100000 debugging
enable password xxxx
backhaul-session-manager
set bh5300-vsc1 client nft
group bhgrp1 set bh5300-vsc1
session group bhgrp1 172.16.20.35 7007 172.16.20.28 7007 0
isdn switch-type primary-ni
isdn voice-call-failure 0
no scripting tcl init
no scripting tcl encdir
voice call carrier capacity active
voice class codec 1
codec preference 1 g723r63
codec preference 2 g711ulaw
no voice hpi capture buffer
no voice hpi capture destination
dial-control-mib retain-timer 240
dial-control-mib max-size 600
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24 service mgcp
controller T1 1
framing esf
clock source line secondary 1
linecode b8zs
ds0-group 0 timeslots 1-24 type none service mgcp
controller T1 2
framing esf
clock source line secondary 2
linecode b8zs
controller T1 3
framing esf
clock source line secondary 3
linecode b8zs
interface Ethernet0
no ip address
no ip mroute-cache
shutdown
interface Serial0
no ip address
no ip mroute-cache
shutdown
clockrate 2015232
no fair-queue
interface Serial1
no ip address
no ip mroute-cache
shutdown
clockrate 2015232
no fair-queue
interface Serial2
no ip address
no ip mroute-cache
shutdown
clockrate 2015232
no fair-queue
interface Serial3
no ip address
no ip mroute-cache
shutdown
clockrate 2015232
no fair-queue
interface Serial0:23
no ip address
isdn switch-type primary-ni
isdn bind-l3 backhaul bh5300-vsc1
no cdp enable
interface FastEthernet0
ip address 172.16.20.28 255.255.255.192
no ip mroute-cache
duplex full
speed auto
no cdp enable
ip classless
ip route 0.0.0.0 0.0.0.0 172.16.20.1
no ip http server
radius-server host 172.21.59.165 auth-port 1645 acct-port 1646
radius-server key xxxxxxxx
radius-server vsa send accounting
voice-port 0:23
voice-port 1:0
mgcp
mgcp call-agent 172.16.20.35 2427 service-type mgcp version 0.1
mgcp quarantine mode loop
mgcp package-capability dtmf-package
mgcp package-capability rtp-package
mgcp package-capability as-package
mgcp default-package gm-package
mgcp profile default
timeout tsmax 100
no max1 lookup
dial-peer cor custom

I have an iphone 3gs, and i keep getting blocked number calling me (prank calls) im getting really fed up with it all now i have had 12 calls from the same person! i was just wondering if there is any way i could block this caller from calling me?

i have an iphone 3gs, and i keep getting blocked number calling me (prank calls) im getting really fed up with it all now i have had 12 calls from the same person! i was just wondering if there is any way i could block this caller from calling me?

Contact your carrier and ask them.

MGCP problem with c881 voice

Hi,
I want to use a cisco "881 voice" with my CUCM v9.1.
This router, have a mgcp configuration and it's "ok", but i can't configure the BRI and the FXS with the CUCM.
It's possible to use the 881 with mgcp and configure fxs and bri?
Thanks.

Hi,
check these links
https://supportforums.cisco.com/discussion/11967296/cisco881v-c881-voice-bri-ports-remain-unregistered-cucm91
CUCM Add additional subunits to C887VA-V and C881VA-V SCCP gateway page
CSCul59124
-->
Description
Symptom:
Unable to add both the BRI and FXS ports on a C887VA-V SCCP voice gateway. When adding the FXS ports, the devicename is defined incorrectly, causing registration failure
Conditions:
CUCM 8.6.x, 9.x and 10.x
Cisco 887VA-V and 881VA-V voice gateway defined as SCCP controlled
Workaround:
Add the gateway under the 'Cisco 881/886/887' gateway entry. Add the FXS ports to 'Subunit 1'
Known Affected Releases:
(1)
9.1(2.12023.1
regds,
aman

Block unwanted calls with Lumia 520

How can I block unwanted calls with my Lumia 520 ,
Solved!
Go to Solution.

You use THIS.
Click on the blue Star Icon below if my advice has helped you or press the 'Accept As Solution' link if I solved your problem..

Deal with multiple inbound calls on 2 x BRI to one extension number

This feels like a really dumb question but I have RTFM and still am stuck.. This is my first UC setup, although I do a lot of IP networking...
We have a UC540W with 2 x BRI so up to 4 PSTN calls.
I want the inbound calls to go to extension 100 which should appear on a line key on all phones so any user can & should pick up incoming calls.
Phones are SPA500 Series (504 and 508)
I am working in CCA.
If I set it up with Line 100 as a shared number on all phones then only 2 calls can be presented and the 2nd call is only presented to the phone with the first call on it, the line key just shows as in use (red) on all other extensions.
The phones are not octal line compatible otherwise this would seem to be the answer to the problem.
I tried setting up (Via CLI) multiple ephone-dn with the same extension number and these appeared in the list for setting an overlay button, but CCA then overwrote these which makes it an unsuitable solution.
The other option would seem to be to set up a multiple extensions (101-104) and put these into a hunt group and under an overlay button, but as these numbers won't be assigned to an extension so can't see how to create these within CCA - if I put them in as floating extension then they appear in the list for an overlay button but not in the available list for setting up a hunt group....
This doesn't seem like such an unusual setup that it should be so difficult...
TIA
David

AFAIK you can have one iCloud account per user account and I can't find any suggestion that there is a limit to the number you can have.
However if in doing all this on the Mac you are talking about you have already created three iCloud accounts then you can't create any more, and there is no workaround for that. You can still sign into an existing iCloud account which has been created on another device, so is the sixth account you are talking about one which already exists? If it isn't, then you would have to use anoter device to create it, then you should be able to sign into it.
As to 'dissociating', just signing out in System Preferences>iCloud should do it. AFAIK the accounts are not tied to the hardware.
All this is to do with iCloud - iTunes accounts are a different can of digital worms and may behave differently.

CUCM 10.5 + MGCP VG Blocking 'Anonymous' calls

Hello to all.
I currently have a CUCM 10.5 installation with a 2921 VG configured with MGCP.
The problem experienced is that when someone calls with hidden caller id, the provider sends the Calling Party Number as 'anonymous'.
This causes CUCM not to process the call and reject it.
The odd thing is that I have another customer with CUCM 5.X with a similar setup (MGCP VG & anonymous as the caller ID ) where calls are processed without a problem.
I've searched the forums and saw other experincing the same issue as we are, even with CUCM 7+, however no answer or solution was provided.
Changing to H323 is not an option at this point.
Any input would be highly appreciated.
Many thanks,
Chris.

Hello Carlo,
Here is the output from the VG when dialing inwards through a phone with CallerID disabled.
I also just spoke with Cisco TAC and they have told me there is no current work-around and that we need to change the VG to H323.
Please let me know if you have any suggestions.
Thanks,
Chris.
Nov 4 11:47:21.977: ISDN Se0/1/0:15 Q931: RX <- CONNECT pd = 8 callref = 0x9300
        Date/Time i = 0x0E0B040D2F15
                Date (dd-mm-yr)   = 14-11-04
                Time (hr:mnt:sec) = 13:47:21
Nov 4 11:47:21.985: ISDN Se0/1/0:15 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x1300
VG2921#
Nov 4 11:47:24.413: ISDN Se0/1/0:15 Q931: RX <- SETUP pd = 8 callref = 0x15DC
        Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98381
                Exclusive, Channel 1
        Calling Party Number i = 0x0180, 'anonymous'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x81, '4321586251'
                Plan:ISDN, Type:Unknown
Nov 4 11:47:24.417: MGCP Packet received from 192.168.3.10:2427--->
CRCX 52335 S0/SU1/DS1-0/[email protected] MGCP 0.1
C: D000000001ab4c5e000000F5800015dc
X: 1
L: p:20, a:PCMU, s:off, t:b8
M: recvonly
R: D/[0-9ABCD*#]
Q: process,loop
<---
VG2921#
Nov 4 11:47:24.421: MGCP Packet sent to 192.168.3.10:2427--->
200 52335 OK
I: 28B9
v=0
c=IN IP4 192.168.3.9
m=audio 21444 RTP/AVP 0 100
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
<---
Nov 4 11:47:24.425: ISDN Se0/1/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x95DC
        Channel ID i = 0xA98381
                Exclusive, Channel 1
Nov 4 11:47:24.425: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x95DC
        Cause i = 0x8095 - Call rejected
Nov 4 11:47:24.449: ISDN Se0/1/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x15DC
Nov 4 11:47:24.449: MGCP Packet received from 192.168.3.10:2427--->
VG2921#DLCX 52336 S0/SU1/DS1-0/[email protected] MGCP 0.1
C: D000000001ab4c5e000000F5800015dc
I: 28B9
X: 1
S:
<---
Nov 4 11:47:24.469: MGCP Packet sent to 192.168.3.10:2427--->
250 52336 OK
P: PS=0, OS=0, PR=0, OR=0, PL=0, JI=0, LA=0
<---
Nov 4 11:47:24.469: ISDN Se0/1/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x95DC
VG2921#
Nov 4 11:47:40.401: MGCP Packet sent to 192.168.3.10:2427--->
NTFY 902475903 *@VG2921.cucm.local MGCP 0.1
X: 0
O:
<---
Nov 4 11:47:40.401: MGCP Packet received from 192.168.3.10:2427--->
200 902475903
<---
VG2921#
Nov 4 11:47:48.249: MGCP Packet received from 192.168.3.10:2427--->
MDCX 52337 S0/SU1/DS1-0/[email protected] MGCP 0.1
C: D000000001ab4c4a000000F5000012fb
I: 28B3
X: 1e
M: recvonly
R: D/[0-9ABCD*#]
Q: process,loop
<---
Nov 4 11:47:48.249: MGCP Packet sent to 192.168.3.10:2427--->
200 52337 OK
<---
Nov 4 11:47:48.253: ISDN Se0/1/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x12FB
        Cause i = 0x8090 - Normal call clearing
Nov 4 11:47:48.289: ISDN Se0/1/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x92FB
        Cause i = 0x8190 - Normal call clearing
Nov 4 11:47:48.289: MGCP Packet received from 192.168.3.10:2427--->
VG2921#DLCX 52338 S0/SU1/DS1-0/[email protected] MGCP 0.1
C: D000000001ab4c4a000000F5000012fb
I: 28B3
X: 1e
S:
<---
Nov 4 11:47:48.305: MGCP Packet sent to 192.168.3.10:2427--->
250 52338 OK
P: PS=5510, OS=881600, PR=5497, OR=879520, PL=0, JI=0, LA=0
<---
Nov 4 11:47:48.309: ISDN Se0/1/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x12FB
VG2921#
Nov 4 11:47:54.449: MGCP Packet sent to 192.168.3.10:2427--->
NTFY 902475904 *@VG2921.cucm.local MGCP 0.1
X: 0
O:
<---
Nov 4 11:47:54.449: MGCP Packet received from 192.168.3.10:2427--->
200 902475904
<---
VG2921#

Blocking incoming collect calls in the voice gateway

Hello
I am using a C3825 router and I want to block incoming collect calls. I tried the command "double-answer" under cas-custom but it is not working. Does anyone have an alternative? I am using an E1 R2 digital.
Thank you
Marcos

Jonathad,
OP is not in the US, and does not have ISDN, has E1 R2 instead
E1 R2 has a method to block collect calls called double-answer. This method is supported and documented by Cisco.
But for some reason it doesn't work for OP.
In these case, it is necessary to involve an experienced consultant, if not TAC escalation directly.

Is it possible to record parts of a phone call on Skype into Logic, then play back the audio while still on the Skype call? I'm a voice teacher coaching students in other cities, and Skype is not helpful with this idea.

Is it possible to record parts of a phone call on Skype into Logic, then play back the audio while still on the Skype call? I'm a voice teacher coaching students in other cities, and Skype can't offer any suggestions.

Hi
What I'm saying is:
Make a new MONO audio track (use the + button). Assign it to input 1
Make a second new MONO track, assign it to input 2
Name both tracks, record-enable both tracks, and record.
You could, of course, make both tracks in one go via the new track dialog box, and hav ethem automatically assigned to ascending inputs.
CCT

Blocking inbound call in CM with MGCP Voice gateways

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