Re: [OSL | CCIE_Voice] HQ call SC and transfer to CUE
Dear ViK , thanks a lot for you r response it is work with me , after 2 days troubleshooting , the issue was from ip blue , when i use Crisco ip communicator work fine thanks for your time Subject: Re: [OSL | CCIE_Voice] HQ call SC and transfer to CUE From: vma...@ipexpert.com Date: Mon, 12 Mar 2012 22:51:11 -0700 CC: ccie_voice@onlinestudylist.com To: mercy_for_...@hotmail.com Make sure you are not using ANY voice-class codec on the dial-peer from GK and the dial-peer to CUE. Also make sure you allow H323 to SIP connections. If this does not help send me the entire config. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 12, 2012, at 7:45 AM, mercy forall wrote: Hi all tried to call cue form HQ , i can not give me dissconect , the call use codeck g729 , i install transcoder 3 session in site c voice mail work in sc and from pstn , but if the call come through GK disconnect , give me disconnect code 47 i review all configuration , and also my frind review it , no issue in configuratin , i dont know why ? is this hardware issue , or miss conf debug ccsip mess 2-R3# Mar 12 07:18:41.612: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:4220@177.3.11.2:5060 SIP/2.0 Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 Remote-Party-ID: HQPH2 sip:2002@177.3.11.1;party=calling;screen=yes;privacy=off From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 To: sip:4220@177.3.11.2 Date: Mon, 12 Mar 2012 07:18:41 GMT Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2160937878-1352913397-469769730-16575 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1331536721 Contact: si BR2-R3#p:2002@177.3.11.1:5060 Expires: 180 Allow-Events: telephone-event Content-Length: 0 Mar 12 07:18:41.628: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 0 Timestamp: 1331536721 Contact: sip:4220@177.3.11.2:5060 Mar 12 07:18:41.628: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 0 Contact: sip:4220@177.3.11.2:5060 Mar 12 07:18:41.632: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 Ok Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 174 Contact: sip:4220@177.3.11.2:5060 Content-Type: application/sdp Call-Info: sip:177.3.11.2:5060;method=NOTIFY;Event=telephone-event;Duration=2000 Allow-Events: telephone-event v=0 o=CiscoSystemsSIP-Workflow-App-UserAgent 1350 1350 IN IP4 177.3.11.2 s=SIP Call c=IN IP4 177.3.11.2 BR2-R3# t=0 0 m=audio 16904 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 Mar 12 07:18:41.636: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:4220@177.3.11.2:5060 SIP/2.0 Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 Remote-Party-ID: HQPH2 sip:2002@177.3.11.1;party=calling;screen=yes;privacy=off From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 To: sip:4220@177.3.11.2 Date: Mon, 12 Mar 2012 07:18:41 GMT Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2160937878-1352913397-469769730-16575 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1331536721 Contact: sip:2002@177.3.11.1:5060 Expires: 180 Allow-Events: telephone-event Content-Length: 0 Mar 12 07:18:41.644: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 Ok Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 174 Contact: sip:4220@177.3.11.2:5060 Content-Type: application/sdp Call-Info: sip:177.3.11.2:5060;method=NOTIFY;Event=telephone-event;Duration=2000 Allow-Events: telephone-event v=0 o=CiscoSystemsSIP-Workflow-App-UserAgent 1350 1350 IN IP4 177.3.11.2 s=SIP Call c=IN IP4 177.3.11.2 t=0 0 m=audio 16904 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 Mar 12 07:18:41.980: //-1
[OSL | CCIE_Voice] HQ call SC and transfer to CUE
Hi all tried to call cue form HQ , i can not give me dissconect , the call use codeck g729 , i install transcoder 3 session in site c voice mail work in sc and from pstn , but if the call come through GK disconnect , give me disconnect code 47 i review all configuration , and also my frind review it , no issue in configuratin , i dont know why ? is this hardware issue , or miss conf debug ccsip mess 2-R3# Mar 12 07:18:41.612: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:4220@177.3.11.2:5060 SIP/2.0 Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 Remote-Party-ID: HQPH2 sip:2002@177.3.11.1;party=calling;screen=yes;privacy=off From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 To: sip:4220@177.3.11.2 Date: Mon, 12 Mar 2012 07:18:41 GMT Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2160937878-1352913397-469769730-16575 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1331536721 Contact: si BR2-R3#p:2002@177.3.11.1:5060 Expires: 180 Allow-Events: telephone-event Content-Length: 0 Mar 12 07:18:41.628: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 0 Timestamp: 1331536721 Contact: sip:4220@177.3.11.2:5060 Mar 12 07:18:41.628: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 0 Contact: sip:4220@177.3.11.2:5060 Mar 12 07:18:41.632: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 Ok Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 174 Contact: sip:4220@177.3.11.2:5060 Content-Type: application/sdp Call-Info: sip:177.3.11.2:5060;method=NOTIFY;Event=telephone-event;Duration=2000 Allow-Events: telephone-event v=0 o=CiscoSystemsSIP-Workflow-App-UserAgent 1350 1350 IN IP4 177.3.11.2 s=SIP Call c=IN IP4 177.3.11.2 BR2-R3# t=0 0 m=audio 16904 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 Mar 12 07:18:41.636: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:4220@177.3.11.2:5060 SIP/2.0 Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 Remote-Party-ID: HQPH2 sip:2002@177.3.11.1;party=calling;screen=yes;privacy=off From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 To: sip:4220@177.3.11.2 Date: Mon, 12 Mar 2012 07:18:41 GMT Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2160937878-1352913397-469769730-16575 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1331536721 Contact: sip:2002@177.3.11.1:5060 Expires: 180 Allow-Events: telephone-event Content-Length: 0 Mar 12 07:18:41.644: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 Ok Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 174 Contact: sip:4220@177.3.11.2:5060 Content-Type: application/sdp Call-Info: sip:177.3.11.2:5060;method=NOTIFY;Event=telephone-event;Duration=2000 Allow-Events: telephone-event v=0 o=CiscoSystemsSIP-Workflow-App-UserAgent 1350 1350 IN IP4 177.3.11.2 s=SIP Call c=IN IP4 177.3.11.2 t=0 0 m=audio 16904 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 Mar 12 07:18:41.980: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 Ok Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To: sip:4220@177.3.11.2;tag=cue7b03fb81 From: HQPH2 sip:2002@177.3.11.1;tag=5DD76C-1051 Call-ID: 6DB359A6-6B4A11E1-804CE032-8E5D7E0@177.3.11.1 CSeq: 101 INVITE Content-Length: 174 Contact: sip:4220@177.3.11.2:5060 Content-Type: application/sdp Cal BR2-R3#l-Info: sip:177.3.11.2:5060;method=NOTIFY;Event=telephone-event;Duration=2000 Allow-Events: telephone-event v=0 o=CiscoSystemsSIP-Workflow-App-UserAgent 1350 1350 IN IP4 177.3.11.2 s=SIP Call c=IN IP4 177.3.11.2 t=0 0 m=audio 16904 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 Mar 12 07:18:42.984: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 Ok Via: SIP/2.0/UDP 177.3.11.1:5060;branch=z9hG4bK6974 To:
Re: [OSL | CCIE_Voice] OUTVIA INVIA
Hi Vik , thank you for your great way to explain , Now it is clear for me. thanks all for your effort. From: Vik Malhi Sent: Wednesday, February 08, 2012 8:49 AM To: datucha123 datucha123 Cc: mercy forall ; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] OUTVIA INVIA viazones have always been one of the most misunderstood topics- hence this email to provide some clarity. In a nutshell- Invia is always checked (for ARQ and for LRQ) before outvia but outvia is used more often. Let's look at an example. Imagine that we have UCM and CME each registered in their own independent local zone and also a third remote Backbone zone defined on another gatekeeper. Extension 4XXX is routed to the CME and international calls (numbers beginning with 011) are routed to the backbone zone. Config below: gatekeeper zone local zoneUCM abc.com zone local zoneCME abc.com zone remote BB abc.com 1.1.1.1 1719 zone prefix zoneCME 4... zone prefix BB 011* no shut Let's look at two calls. UCM GK CME and also UCM GK BB. Note - in both cases zoneUCM is the source zone and zoneCME/BB are the destination zones for the two calls respectively. If we add a CUBE to the config we can invoke the CUBE in two ways. INVIA gatekeeper zone local zoneUCM abc.com invia VIAZONE zone local zoneCME abc.com zone local VIAZONE abc.com zone remote BB abc.com 1.1.1.1 1719 zone prefix zoneCME 4... zone prefix BB 011* no shut In this instance the CUBE will be invoked for both types of calls since the source zone has been configured with an invia command. And in both types of calls that we are making the source zone is zoneUCM. Note- If we configure outvia for zoneUCM the CUBE will not be invoked since it is the invia that is used on source zones. OUTVIA With the invia configuration above we invoke CUBE for any call coming from the UCM zone. We don't care where the call is destined for- as long as the call comes from zoneUCM we invoke the CUBE. If we only wanted to invoke CUBE for calls to the backbone (and not for calls from UCM GK CME) then do as follows: gatekeeper zone local zoneUCM abc.com zone local zoneCME abc.com zone local VIAZONE abc.com zone remote BB abc.com 1.1.1.1 1719 outvia VIAZONE zone prefix zoneCME 4... zone prefix BB 011* no shut For the call UCM GK BB the CUBE is invoked since the destination zone has been configured with an outvia. For the call UCM GK CME the CUBE is not invoked since neither the source zone (zoneUCM) nor the destination zone (zoneCME) has been configured with an invia/outvia. One last thing to mention- if the source zone has been configured with an invia AND the destination zone has been configured with an outvia, the invia trumps the outvia and the outvia is not used (CUBE is not invoked twice). On Feb 7, 2012, at 12:03 PM, datucha123 datucha123 wrote: Outvia is more accurate. Invia, in most cases, is used for incoming LRQs. On Tue, Feb 7, 2012 at 11:13 PM, mercy forall mercy_for_...@hotmail.com wrote: Hi All now in outvia and invia ,, Are is it deference if i use it in local zone or remote zone ? As per Doc, outvia for any traffic leave this zone , so are this same if i use outvia in local or remote zone I need to send the call form local zone to remote zone through CUBE as local zone , what is the correct [zone remote with outvia OR with invia CUBE ] ? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MGCP Registration
Hi, thanks for your support and good link\ this is my GW configuration , also it is connected to other cisco GW as PSTN GW through E1 cross cable sh run :Current configuration : 15381 bytes! !version 15.0service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname !boot-start-markerboot-end-marker!logging buffered 51200 warnings no aaa new-model network-clock-participate wic 2 !dot11 syslogip source-route!ip cef!!no ipv6 cefmultilink bundle-name authenticated!!!isdn switch-type primary-qsig!voice-card 0!!voice rtp send-recv!voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip redirect ip2ip h323 sip header-passing no call service stop!voice class codec 1 codec preference 1 g711ulaw!voice class custom-cptone dualtone disconnect frequency 425 cadence 250 250http client cache memory pool 15000http client cache memory file 500http client connection timeout 60http client connection idle timeout 10http client response timeout 30 mrcp client timeout connect 10mrcp client timeout message 10mrcp client rtpsetup enablevxml tree memory 500vxml audioerrorvxml version 2.0!crypto pki trustpoint TP-self-signed-3307538538 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-3307538538 revocation-check none rsakeypair TP-self-signed-3307538538 !controller E1 0/2/0 pri-group timeslots 1-4,16 service mgcp! interface GigabitEthernet0/0 no ip address duplex auto speed auto media-type rj45! interface GigabitEthernet0/0.1 encapsulation dot1Q ip address X.X.X.X 255.255.255.0! !interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-qsig isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable!ip forward-protocol nd!!ip http serverip http access-class 23ip http authentication localip http secure-serverip http timeout-policy idle 60 life 86400 requests 1ip route 0.0.0.0 0.0.0.0 X.X.X.X! !!control-plane!call threshold global cpu-5sec low 70 high 85!voice-port 0/2/0:15!voice-port 0/3/0!voice-port 0/3/1!ccm-manager switchback immediateccm-manager fallback-mgcp ccm-manager redundant-host X.X.x.xccm-manager mgcpccm-manager config server x.x.x.x !mgcpmgcp call-agent x.x.x.x service-type mgcp version 1.0mgcp bind control source-interface GigabitEthernet0/0.1mgcp bind media source-interface GigabitEthernet0/0.1!mgcp profile default!!gateway timer receive-rtp 1200!sip-ua retry invite 1 retry bye 1 retry cancel 1 timers expires 6 reason-header override!!telephony-service max-conferences 12 gain -6 transfer-system full-consult! thanks From: gogli...@gmail.com Date: Mon, 16 Jan 2012 13:29:49 +0100 Subject: Re: [OSL | CCIE_Voice] MGCP Registration To: mercy_for_...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi, The card is supported: http://www.cisco.com/en/US/prod/collateral/routers/ps259/product_data_sheet0900aecd8057f2e0.pdf Paste your voice gateway configuration, there must be a problem somewhere. Cheers, On Sun, Jan 15, 2012 at 8:45 PM, mercy forall mercy_for_...@hotmail.com wrote: Hi, I think the issue from card , i need to sure the CUCM support VWIC-1MFT-G703 card or not , because i did not fine it in cucm mgcp configuration and i use VWIC-1MFT-E1 i made all configuration , and made same configuration on other and it is work fine , from where i can check if is it support tor not ? From: ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 71, Issue 54 To: ccie_voice@onlinestudylist.com Date: Sun, 15 Jan 2012 06:01:19 -0500 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: AAR for UCCX (Mohammed Al Baqari) 2. BAT csv template for CUC 7 (donny f) 3. MGCP Registration (mercy forall) 4. Re: AAR for UCCX (datucha123 datucha123) 5. Re: MGCP Registration (datucha123 datucha123) -- Message: 1 Date: Sun, 15 Jan 2012 02:24:54 +0400 From: Mohammed Al Baqari baqari.voic...@gmail.com To: 'datucha123 datucha123' datucha...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] AAR for UCCX Message-ID: 078201ccd30b$597061a0$0c5124e0$@gmail.com Content-Type: text/plain; charset=us-ascii Hi, I haven't tested this. But my info is based on CUCM SRND. They haven't given any exception on AAR and CTI inter
Re: [OSL | CCIE_Voice] MGCP Registration
this is :- show isdn status \Global ISDN Switchtype = primary-qsig %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply ISDN Serial0/2/0:15 interfacedsl 0, interface ISDN Switchtype = primary-qsig Slave side configuration L2 Protocol = Q.921 0x L3 Protocol(s) = CCM MANAGER 0x0003 Layer 1 Status: ACTIVELayer 2 Status:TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNEDLayer 3 Status:0 Active Layer 3 Call(s)Active dsl 0 CCBs = 0The Free Channel Mask: 0x800FNumber of L2 Discards = 0, L2 Session ID = 6Total Allocated ISDN CCBs = 0 Date: Mon, 16 Jan 2012 17:44:32 +0400 Subject: Re: [OSL | CCIE_Voice] MGCP Registration From: datucha...@gmail.com To: mercy_for_...@hotmail.com CC: gogli...@gmail.com; ccie_voice@onlinestudylist.com What does the show isdn status shows up? On Mon, Jan 16, 2012 at 5:35 PM, mercy forall mercy_for_...@hotmail.com wrote: Hi, thanks for your support and good link\ this is my GW configuration , also it is connected to other cisco GW as PSTN GW through E1 cross cable sh run : Current configuration : 15381 bytes ! ! version 15.0 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname ! boot-start-marker boot-end-marker ! logging buffered 51200 warnings no aaa new-model network-clock-participate wic 2 ! dot11 syslog ip source-route ! ip cef ! ! no ipv6 cef multilink bundle-name authenticated ! ! ! isdn switch-type primary-qsig ! voice-card 0 ! ! voice rtp send-recv ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip redirect ip2ip h323 sip header-passing no call service stop ! voice class codec 1 codec preference 1 g711ulaw ! voice class custom-cptone dualtone disconnect frequency 425 cadence 250 250 ! ! ! ! http client cache memory pool 15000 http client cache memory file 500 http client connection timeout 60 http client connection idle timeout 10 http client response timeout 30 mrcp client timeout connect 10 mrcp client timeout message 10 mrcp client rtpsetup enable vxml tree memory 500 vxml audioerror vxml version 2.0 ! crypto pki trustpoint TP-self-signed-3307538538 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-3307538538 revocation-check none rsakeypair TP-self-signed-3307538538 ! controller E1 0/2/0 pri-group timeslots 1-4,16 service mgcp ! interface GigabitEthernet0/0 no ip address duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/0.1 encapsulation dot1Q ip address X.X.X.X 255.255.255.0 ! ! interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-qsig isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! ip forward-protocol nd ! ! ip http server ip http access-class 23 ip http authentication local ip http secure-server ip http timeout-policy idle 60 life 86400 requests 1 ip route 0.0.0.0 0.0.0.0 X.X.X.X ! ! ! control-plane ! call threshold global cpu-5sec low 70 high 85 ! voice-port 0/2/0:15 ! voice-port 0/3/0 ! voice-port 0/3/1 ! ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host X.X.x.x ccm-manager mgcp ccm-manager config server x.x.x.x ! mgcp mgcp call-agent x.x.x.x service-type mgcp version 1.0 mgcp bind control source-interface GigabitEthernet0/0.1 mgcp bind media source-interface GigabitEthernet0/0.1 ! mgcp profile default ! ! gateway timer receive-rtp 1200 ! sip-ua retry invite 1 retry bye 1 retry cancel 1 timers expires 6 reason-header override ! ! telephony-service max-conferences 12 gain -6 transfer-system full-consult ! thanks From: gogli...@gmail.com Date: Mon, 16 Jan 2012 13:29:49 +0100 Subject: Re: [OSL | CCIE_Voice] MGCP Registration To: mercy_for_...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi, The card is supported: http://www.cisco.com/en/US/prod/collateral/routers/ps259/product_data_sheet0900aecd8057f2e0.pdf Paste your voice gateway configuration, there must be a problem somewhere. Cheers, On Sun, Jan 15, 2012 at 8:45 PM, mercy forall mercy_for_...@hotmail.com wrote: Hi, I think the issue from card , i need to sure the CUCM support VWIC-1MFT-G703 card or not , because i did not fine it in cucm mgcp configuration and i use VWIC-1MFT-E1 i made all configuration , and made same configuration on other and it is work fine , from where i can check if is it support tor not ? From: ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 71, Issue 54 To: ccie_voice@onlinestudylist.com Date: Sun, 15 Jan 2012 06:01:19 -0500 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo
[OSL | CCIE_Voice] MGCP Registration
Hi, i have 3845 with VWIC-1MFT-G703 i configured mgcp but it is bending in registering on sh ccm-manager i did not find VWIC-1MFT-G703 on CUCM 7 , i use VWIC-1MFT-E1 , Are this card supported in mgcp ? thanks___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MGCP Registration
Hi, I think the issue from card , i need to sure the CUCM support VWIC-1MFT-G703 card or not , because i did not fine it in cucm mgcp configuration and i use VWIC-1MFT-E1 i made all configuration , and made same configuration on other and it is work fine , from where i can check if is it support tor not ? From: ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 71, Issue 54 To: ccie_voice@onlinestudylist.com Date: Sun, 15 Jan 2012 06:01:19 -0500 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: AAR for UCCX (Mohammed Al Baqari) 2. BAT csv template for CUC 7 (donny f) 3. MGCP Registration (mercy forall) 4. Re: AAR for UCCX (datucha123 datucha123) 5. Re: MGCP Registration (datucha123 datucha123) -- Message: 1 Date: Sun, 15 Jan 2012 02:24:54 +0400 From: Mohammed Al Baqari baqari.voic...@gmail.com To: 'datucha123 datucha123' datucha...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] AAR for UCCX Message-ID: 078201ccd30b$597061a0$0c5124e0$@gmail.com Content-Type: text/plain; charset=us-ascii Hi, I haven't tested this. But my info is based on CUCM SRND. They haven't given any exception on AAR and CTI inter-operability. Can you please highlight further. AAR does not support CTI route points as the origin or the destination of calls. Also, AAR is incompatible with the Extension Mobility feature when users roam across different sites. Regards, Mohammed Al Baqari From: datucha123 datucha123 [mailto:datucha...@gmail.com] Sent: Sunday, January 15, 2012 1:03 AM To: Mohd Baqari Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] AAR for UCCX AAR is supported with CTI, when it is registered. For instance the CTI Route Point supports AAR, when it is registered. But as soon as it will unregister (or some Dummy CTI Route Point, which never gets registered) does not support AAR. On Sat, Jan 14, 2012 at 10:29 PM, Mohd Baqari baqari.voic...@gmail.com wrote: Hi Datucha, In your scenario whats confusing me is how aar is triggered. As far as I know AAR isn't supported with CTI Regards, Mohammed Al Baqari Sent from my iPhone On Jan 8, 2012, at 9:11 PM, datucha123 datucha123 datucha...@gmail.com wrote: Hello, I have been making a test and find out the following issue: So, I set the AAR Mask to CTI Route Points Line Settings, and the calls from Branch1 to UCCX when there was no WAN bandwidth available were routed through PSTN successfully. I have not set any AAR Group and AAR CSS for CTI ports, but the HQ gateway has them. So when I called the UCCX IVR (simpla AA script) through HQ Gateway and dialed the Branch 1 Phones extensions, the call has been routed through PSTN based on CAC restrictions. Well that's ok with CAC. But I cannot understand how the AAR got worked, when neither CTI ports nor the CTI Route Point had any AAR configuration (AAR Mask has been also removed from the CTI RP Line). HQ Gateway had the AAR Group and CSS set. So based on that, UCCX took the HQ gateway setttings of AAR and made a call? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120115/75ebf1e2/attachment-0001.html -- Message: 2 Date: Sat, 14 Jan 2012 23:13:22 -0700 From: donny f f.faraday...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BAT csv template for CUC 7 Message-ID: CA+jB2JX19oA3d8=5mgzoqu08tg1isgz_jeqtqpkhvsvsi3t...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 hi all, anyone know where to find the BAT template for Unity connection 7 ? tks -- Message: 3 Date: Sun, 15 Jan 2012 08:16:16 + From: mercy forall mercy_for_...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MGCP Registration Message-ID: snt125-w55fc3dce00b1c1886511b6b5...@phx.gbl Content-Type: text/plain; charset=iso-8859-1 Hi, i have 3845
Re: [OSL | CCIE_Voice] incoming called-number .
if there are 2 dial peer for incoming mach dial-peer voice 1 pots incoming called-number . and dial-peer voice 2 pots answer-address 12345 and the ani is 12345 which dial peer will mach ?? From: ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 71, Issue 50 To: ccie_voice@onlinestudylist.com Date: Fri, 13 Jan 2012 20:49:33 -0500 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. CUE with CUCM (datucha123 datucha123) 2. Re: RSVP CAC (Ashraf Ayyash) 3. Re: incoming called-number . (Bill Lake) 4. Re: Caller ID Update (Mohd Baqari) 5. vol 2 lab 7 ---uccx script (donny f) -- Message: 1 Date: Fri, 13 Jan 2012 21:12:34 +0400 From: datucha123 datucha123 datucha...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUE with CUCM Message-ID: CAPZ0ugP+2L6mGzsKij0sH-_AhWOMkdjC=upqxrk0jnrsybq...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello, Do we need to configure the CUCME-SRST (or call-manager-fallback) as the Third Callmanager in JTAPI Subsystem configuration in CUE for SRST fallback? I think we do not need to configure the CUCME-SRST as the Third Callmanager, as CUCME-SRST does not use the JTAPI, but is integrated with SIP, and we have to configure the SIP Subsystem for that. Right? -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120113/f2a1d87f/attachment-0001.html -- Message: 2 Date: Fri, 13 Jan 2012 12:00:54 -0600 From: Ashraf Ayyash ash.ayy...@gmail.com To: study buddy studybudd...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP CAC Message-ID: CAEW==nuccyzqm5t1zlzwbgavt8lqwfdxfklsmqvglmd1gph...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 did you run qos on sc as well ? you know that you have to otherwise you will kill sc sub interface , do show traff and you will see it getting 56 K only , so the solution is to run the QOS on both sub interfaces on HQ and on sc as well Ash On Fri, Jan 13, 2012 at 12:09 AM, study buddy studybudd...@gmail.com wrote: Hi All, For the Lab topology, if I run QoS between HQ SB RSVP CAC between HQ SC set the ip rsvp band to 112 for 4 calls, I can actually make only two calls the third calls get re-routed. Now if I run qos between HQ SC routers as well with a BW of 1536 I can make 3 calls, but the 4th call still fails I get the following RSVP error *Jan 10 12:58:29.883:?? QoS Module: RESV ERROR received : Remote IP: 142.102.64.254 | Local IP: 142.102.66.254 *Jan 10 12:58:29.883: qos_rsvp_resv_notify_events: errCode 1, errVal 2, errFlag 0, errNode 10.10.112.1 *Jan 10 12:58:29.883: qos_rsvp_remove_reservation:? Removing RESV state for CallId? : 0xFFC5 ? Remove Resv: Source (142.102.64.254:17084), Dest (142.102.66.254:17178) Any thoughts on this? TR ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Message: 3 Date: Fri, 13 Jan 2012 12:03:35 -0600 From: Bill Lake whl...@gmail.com To: datucha123 datucha123 datucha...@gmail.com Cc: ccie_voice@onlinestudylist.com, Ken Wyan kew...@gmail.com Subject: Re: [OSL | CCIE_Voice] incoming called-number . Message-ID: CADpb93OumyUp-oCrL5ODLzphi_=-3Jz=7bi0QLut=4ooohr...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 BR1(config)#dial-peer hunt ? 0-7 Dial-peer hunting choices, listed in hunting order within each choice: 0 - Longest match in phone number, explicit preference, random selection. 1 - Longest match in phone number, explicit preference, least recent use. 2 - Explicit preference, longest match in phone number, random selection. 3 - Explicit preference, longest match in phone number, least recent use. 4 - Least recent use, longest match in phone number, explicit preference. 5 - Least recent use, explicit preference, longest match in phone number. 6 - Random selection. 7 - Least recent use. So if you want to really confuse someone change this to option 6 On Fri, Jan 13, 2012 at 9:09 AM, datucha123 datucha123 datucha...@gmail.com wrote:
