Integration between Cisco VOice gateway & Ericcsion MD110 EPBX

Hi
Need to know what parameters needs to match in between Cisco VG & Ericcsion PBX while integration(india off is going to connect through MPLS link(E1) via VG to US, in US we have avaya PBX)
We have a E1 connection (trunk) between VG & PBX.
Also need physical connectivity details like cross or stratght cable details.
Send important doc link also
Regards
Naga.

Hello,
the pinout for the router is documented at at:
http://cisco.com/en/US/products/hw/modules/ps2641/products_module_installation_guide_chapter09186a008007cb4c.html
You will need to configure both PBXs for ISDN PRI as described at;
http://cisco.com/en/US/partner/products/ps6350/products_configuration_guide_chapter09186a00800ca716.html
Note the correct command is "isdn incoming-voice voice" and not "modem" as the documentetion wrong in this.
Hope this helps, please rate post if it does!

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Hello All,
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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

Site to Site Connectivity Between BE6K and Voice Gateway 2901

Greetings,
Is an Ethernet handoff required for site to site connectivity between BE6K and a voice gateway 2901. My vendor is suggesting that it's required in order for both sites to see the BE6K as one phone system. However, here in lies the problem. I have a point-to-point T-1 between the sites that does not have an Ethernet handoff, just the smartjack to the T-1.
What would I need to get this to work? Have a router at each site? If so, which model? Or is there a component I could add to the BE6K or voice gateway?
Any help would be greatly appreciated.
Thanks in advance.

Ethernet handoff just means that the provider will deliver the circuit using Ethernet. What circuit is your provider delivering for you? Do they manage your WAN? Like I said what you need is ip connectivity between your sites and BE6k. If your T1 connection provides WAN connectivity and you have ip connectivity between the sites, then I don't know what you need any handoff for. The question is do you have ip connectivity between the sites via your T1 connection

SIP trunking between Microsoft OCS server and Cisco Voice GW router.

Hello All,
I have a client with an existing Microsoft OCS (office communications server) environment with the OCS server in their head office. The OCS clients in the remote Office registers with the OCS server in the head office. The WAN connectivity between the remote office and the Head office is MPLS.I would like to facilitate local call (PSTN) features at the remote site through a newly proposed Voice gateway router.
Can I achieve this by doing a SIP trunk between the OCS server in the head office to the newly proposed voice GW router in the remote office through the existing MPLS link. If yes, Could any one please assist me in this regards or suggest any other best solution to achieve the same.
Thank you in advance,
Mohammed Ameen R

Hi David,
this is a normal behaviour. To CUCM, OCS is a remote destination (just like your mobile phone). When your mobile phone hangs up, the system will put the call on hold for 10 sec.
This is there for the mobile user to go to his desk to pick up the call and continue the conversation (part of single number reach feature)
The best practise will be for the user to ensure that the other party hangs up the call first before he hang up.
Please grade if you think it's useful =)

Cisco VG202 voice gateway implementations

I have used the Cisco VG202 to replace Multitech units at some of my remote locations.
The phone systems at the remote locations are Avaya Partner systems.
What i encounter, after placing the Cisco units, is that the time displayed on the Avaya phones goes wrong. Even if you manually correct it, time will go off within a couple hours.
Has ANYONE come across this? TIme on the Avaya Partner system is adversely affected with the introduction of the Cisco VG202.

Hi Peter,
This forum is for discussions about specific known bugs. You might find better discussion in the IP Telephony forum: https://supportforums.cisco.com/community/netpro/collaboration-voice-video/ip-telephony
Conceptually as an analog voice gateway the VG shouldn't generally provide any time information to an attached phone system. Is it possible the phone system is pointed to the VG as an NTP reference? In that situation the internal platform clock isn't terribly reliable and synch to an more reliable external NTP reference is recommended. The only other possibility that comes to mind is delivery of a timestamp information along with callerid over the analog voice ports. In that situation the router again depends on internal clock. Configuring an NTP reference on the router will help to keep the router clock better synchronized. These are all a bit of a stretch. The most likely, IMHO, is the pone system pointing to VG202 as NTP reference. In that case giving the VG202 a reliable NTP reference so the router can keep more accurate time would be my first approach.
Hope this helps,
Wes

Document on Cisco 3845 gateway with both Voice and VXML role

Dear all,
we are using single cisco 3845 gateway as Ingress as well as vxml gateway . Can you provide any documentation regarding the call volume capacity this gateway can handle. We have deployed this gateway in a UCCE parent- child model.

Anoop,
It can be found in Table 7-2 on page 148 of CVP SRND Guide
HTH
GP.
Pls rate helpful posts by clicking on the stars under the post !!

Voice gateways, SLT and PGW

Hi everyone,
I dont know if its the right place to start this discussion so forgive me if i am wrong.( I also have opened a discussion in IP telephony portion)
The company in which i am currently wokring is running a call center and IPT as well. i will try to explain the scenario
voicegateways (AS5400 and AS5350 routers with IOS Version 12.4(15)T7)
SLT (2651XM with IOS Version 12.2(8)T10)
PGW (PGW 2200 with SunOS 5.10, MGC)
I was under the impression that if you want to connect PSTN and a voip network you need a PSTN gateway. So why are we using 3 different types of hardware.
Follwing are the explainations in a doucment given to me
the Cisco PGW 2200 provides service providers with the capability to seamlessly route voice and data calls between the PSTN and New World packet networks.
Cisco 2611 Signaling Link Terminals
(E1 terminates, Take Signaling part and send B n D channels to Voice gateway, Signaling part is resolved by it self and PGW)
Voice Gateways
allows terminals of one type, such as H.323, to communicate with terminals of another type, such as a PBX, by converting protocols. Gateways connect an organization’s network to the PSTN
Any information is much appreciated

have you read this
End-of-Sale and End-of-Life Announcement for the Cisco PGW 2200 Softswitch and Software
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2027/end_of_life_notice_c51-676990.html
hope this help

VG202 Voice gateway

I have multiple environments with Avaya Partner phone systems in them. I used touse a multitech VOIP modem for connection back to Headquarters - this works perfectly fine. Situation I am having, when the Multitech units die, I replace them with the Cisco VG202 unit. When placed at the remote locations, the time in the windows of the Avaya phones never corrects it self...it always wrong. It is correct on the Avaya system at the location, but the integration of the Cisco unit seems to make it go awry. All other functionality is there, but the time display is a mess. Any ideas as to where to troubleshoot this? Manually establish the correct time from a console phone at the remote locations, and it goes awry within hours.

Hi Peter,
This forum is for discussions about specific known bugs. You might find better discussion in the IP Telephony forum: https://supportforums.cisco.com/community/netpro/collaboration-voice-video/ip-telephony
Conceptually as an analog voice gateway the VG shouldn't generally provide any time information to an attached phone system. Is it possible the phone system is pointed to the VG as an NTP reference? In that situation the internal platform clock isn't terribly reliable and synch to an more reliable external NTP reference is recommended. The only other possibility that comes to mind is delivery of a timestamp information along with callerid over the analog voice ports. In that situation the router again depends on internal clock. Configuring an NTP reference on the router will help to keep the router clock better synchronized. These are all a bit of a stretch. The most likely, IMHO, is the pone system pointing to VG202 as NTP reference. In that case giving the VG202 a reliable NTP reference so the router can keep more accurate time would be my first approach.
Hope this helps,
Wes

Integration between Cisco VOice gateway & Ericcsion MD110 EPBX

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