HI Everyone,   we want to integrate the UCCX 9.0.2 with MS Dynamics CRM 2013 in such a way that if a customer calls in, the agent should see the customer details in the CRM database. or if customer has called before, the call should pick all the previous details of the caller from CRM server.  Looking forward to your support. Thanks Regards, MWH ... View more

Dear All, I am trying to configure CME 4.1 on 1760-v router for my home purpose. you can find attach sh run, sh inventory and incomming call-log on fxo port, which i got from "debug vpm all" and "terminal monitor" commnads. call is coming to fxo port and moved to aap.tcl script ( inside the flash with all audio.au files), and then sometimes its disconnected immediately, and sometimes keep connected for 10 to 12 sec and then disconnected. Any idea, what going wrong? is it codec issue? PVDM? sh version is also attached ITCS#terminal moni ITCS# ITCS# ITCS# ITCS#debug vpm all All Voice Port Module debugging is enabled Note: DSP message debugging is not part of this command. Use debug vpm dsp to enable it ITCS# *Mar  3 22:46:16.062: htsp_process_event: [2/1, FXOLS_ONHOOK, E_DSP_SIG_0000]fxo ls_onhook_ringing *Mar  3 22:46:16.062: TGRM: reg_invoke_tgrm_call_update(0, 2, 1, 65535, 1, TGRM_ CALL_PENDING, TGRM_CALL_VOICE, TGRM_DIRECTION_IN) *Mar  3 22:46:16.062: htsp_timer - 125 msec *Mar  3 22:46:16.082: htsp_process_event: [2/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxo ls_onhook_ringing *Mar  3 22:46:16.082: TGRM: reg_invoke_tgrm_call_update(0, 2, 0, 65535, 1, TGRM_ CALL_PENDING, TGRM_CALL_VOICE, TGRM_DIRECTION_IN) *Mar  3 22:46:16.086: htsp_timer - 125 msec *Mar  3 22:46:16.174: htsp_process_event: [2/1, FXOLS_WAIT_RING_MIN, E_DSP_SIG_0 100]fxols_ringing_disc *Mar  3 22:46:16.174: htsp_timer_stop3 *Mar  3 22:46:16.174: htsp_timer_stop *Mar  3 22:46:16.174: TGRM: reg_invoke_tgrm_call_update(0, 2, 1, 65535, 1, TGRM_ CALL_IDLE, TGRM_CALL_VOICE, TGRM_DIRECTION_IN) *Mar  3 22:46:16.194: htsp_process_event: [2/0, FXOLS_WAIT_RING_MIN, E_DSP_SIG_0 100]fxols_ringing_disc *Mar  3 22:46:16.194: htsp_timer_stop3 *Mar  3 22:46:16.194: htsp_timer_stop *Mar  3 22:46:16.194: TGRM: reg_invoke_tgrm_call_update(0, 2, 0, 65535, 1, TGRM_ CALL_IDLE, TGRM_CALL_VOICE, TGRM_DIRECTION_IN) ITCS# ITCS#sh version Cisco IOS Software, C1700 Software (C1700-ADVENTERPRISEK9-M), Version 12.4(15)T RELEASE SOFTWARE (fc3) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2007 by Cisco Systems, Inc. Compiled Mon 25-Jun-07 18:39 by prod_rel_team ROM: System Bootstrap, Version 12.2(7r)XM2, RELEASE SOFTWARE (fc1) ITCS uptime is 1 day, 10 hours, 49 minutes System returned to ROM by reload System image file is "flash:c1700-adventerprisek9-mz.124-15.T.bin" This product contains cryptographic features and is subject to United States and local country laws governing import, export, transfer and use. Delivery of Cisco cryptographic products does not imply third-party authority to import, export, distribute or use encryption. Importers, exporters, distributors and users are responsible for compliance with U.S. and local country laws. By using this product you agree to comply with applicable laws and regulations. If you are unable to comply with U.S. and local laws, return this product immediately. A summary of U.S. laws governing Cisco cryptographic products may be found at: http://www.cisco.com/wwl/export/crypto/tool/stqrg.html If you require further assistance please contact us by sending email to export@cisco.com . Cisco 1760 (MPC860P) processor (revision 0x500) with 123157K/7915K bytes of mem ry. Processor board ID FOC08490ZUQ (3780485029), with hardware revision 0000 MPC860P processor: part number 5, mask 2 1 FastEthernet interface 2 Serial(sync/async) interfaces 1 ISDN Basic Rate interface 2 Voice FXO interfaces 32K bytes of NVRAM. 49152K bytes of processor board System flash (Read/Write) Configuration register is 0x3922 Thanks Waqas ... View more

Hi I have IPCC server 5.0 deployed in high availability ( 2 servers in Cluster ), I restarted the secondary server as it was UP since 160 days and after rebooting, all of services on server are still running but when i try login using http://ipadress/appadmin i get the error of "Page Cannot be Displayed" ? Any idea what had wrong? Looking forward to quick response. Thanks Regards Waqas ... View more

Hi All, When i turned on debugging at FXO port to troubleshoot an issue, when call lands on Gateway, it should play an prompt of "Welcome to XXX Company" but i hear a ringtone and then immediately an re-order / Fast Busy tone. Can anyone help me to interpret this output? I can see my cell number as "calling number", Please find below the out of "sh dial-peer voice summary" and "show debugging" Thanks in advance. abc-def-ghi#sh dial-peer voice summary dial-peer hunt 0              AD                                    PRE PASS                OUT TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT 110    pots  up   up             9.T                0                      up 0/2/1 111    pots  up   up                                0                      down 0/2/1 3000   voip  up   up                                0  syst 3001   voip  up   up             1000               0  syst ipv4:10.255.63.254 3002   voip  up   up             1300               0  syst ipv4:10.255.63.254 3003   voip  down down           1000               2  syst ipv4:10.163.242.2 abc-def-ghi# *Oct 16 03:35:00.795: htsp_process_event: [0/2/3, FXOLS_ONHOOK, E_DSP_SIG_0000]f xols_onhook_ringing *Oct 16 03:35:00.799: TGRM: reg_invoke_tgrm_call_update(0, 2, 3, 65535, 1, TGRM_ CALL_PENDING, TGRM_CALL_VOICE, TGRM_DIRECTION_IN) *Oct 16 03:35:00.799: htsp_timer - 125 msec *Oct 16 03:35:00.927: htsp_process_event: [0/2/3, FXOLS_WAIT_RING_MIN, E_HTSP_EV ENT_TIMER]fxols_wait_ring_min_timer *Oct 16 03:35:00.927: htsp_timer - 10000 msec *Oct 16 03:35:00.927: htsp_timer3 - 5600 msec *Oct 16 03:35:00.927: [0/2/3] htsp_start_caller_id_rx:BELLCORE *Oct 16 03:35:00.927: //-1/313233343930/DSM:(0/2/3):-1/dsp_stream_mgr_create: . *Oct 16 03:35:00.927: //-1/313233343930/DSM:(0/2/3):-1/set_echo_canceller_data: echo_cancel: 0 *Oct 16 03:35:00.927: //-1/313233343930/DSM:(0/2/3):-1/set_echo_canceller_data: echo_flags: 38, echo_len: 512 *Oct 16 03:35:00.927: //-1/313233343930/DSM:(0/2/3):-1/dsp_stream_mgr_implement_ feature: *Oct 16 03:35:00.927: dsp_stream_mgr_implement_feature: feature id = 7 *Oct 16 03:35:00.927: //-1/313233343930/DSM:(0/2/3):-1/dsmp_dsmapi_reserve_resou rce_cb: . *Oct 16 03:35:00.927: [0/2/3] htsp_dsm_create_success  returns 1 *Oct 16 03:35:00.927: //-1/313233343930/DSM:(0/2/3):-1/dsmp_dsmapi_reserve_resou rce_cb: . *Oct 16 03:35:00.927: [0/2/3] htsp_dsm_is_sig_done_cb *Oct 16 03:35:01.891: htsp_process_event: [0/2/3, FXOLS_RINGING, E_DSP_SIG_0100] *Oct 16 03:35:01.891: fxols_ringing_not *Oct 16 03:35:01.891: htsp_timer_stop *Oct 16 03:35:01.891: htsp_timer - 10000 msec *Oct 16 03:35:02.931: //-1/313233343930/DSM:(0/2/3):-1/dsmp_dsmapi_detection_cb: . *Oct 16 03:35:02.931: [0/2/3] htsp_dsm_feature_notify_cb  returns 2 id=DSM_FEATU RE_SM_CALLERID_RX *Oct 16 03:35:02.931: htsp_process_event: [0/2/3, FXOLS_RINGING, E_HTSP_CALLERID _RX_DONE] *Oct 16 03:35:02.931: htsp_timer_stop *Oct 16 03:35:02.931: [0/2/3] htsp_stop_caller_id_rx. message length 28 *Oct 16 03:35:02.931: //-1/313233343930/DSM:(0/2/3):-1/dsp_stream_mgr_destroy: . *Oct 16 03:35:02.931: //-1/313233343930/DSM:(0/2/3):-1/dsm_delete_all_streams: . *Oct 16 03:35:02.931: //-1/313233343930/DSM:(0/2/3):-1/dsmapi_accept_modem_passt hrough_session:  : dmgr: 4778EFE0, active sessions 0, max sessions: 16 rejected sessions till now: 0htsp_setup_ind *Oct 16 03:35:02.935: [0/2/3] get_fxo_caller_id:Caller ID received. Message type =128 length=26 checksum=7B *Oct 16 03:35:02.935: [0/2/3] Caller ID String 80 17 01 08 31 30 31 36 30 38 32 31 02 0B 30 33 30 31 38 35 36 39 36 36 39 7B *Oct 16 03:35:02.935: [0/2/3] get_fxo_caller_id calling num=03018569669 calling name= calling time=10/16 08:21 *Oct 16 03:35:02.935: TGRM: reg_invoke_tgrm_accept_call(1, TGRM_CALL_VOICE, TGRM _DIRECTION_IN, 0, 2, 3, 65535) *Oct 16 03:35:02.935: TGRM: Calling vtsp_tsp_call_setup_ind(). *Oct 16 03:35:02.935: fxols_callerid_done: call being answered *Oct 16 03:35:02.939: htsp_process_event: [0/2/3, FXOLS_WAIT_SETUP_ACK, E_HTSP_S ETUP_ACK] *Oct 16 03:35:02.939: fxols_wait_setup_ack: *Oct 16 03:35:02.939: [0/2/3] set signal state = 0xC timestamp = 0 *Oct 16 03:35:02.939: dsp_set_sig_state: [0/2/3] packet_len=12 channel_id=131 pa cket_id=39 state=0xC timestamp=0x0fxols_check_auto_call *Oct 16 03:35:02.947: flex_dsprm_close_cleanup *Oct 16 03:35:02.947: //-1/313233343930/DSM:(0/2/3):-1/dsmp_dsmapi_reserve_resou rce_cb: . *Oct 16 03:35:02.947: [0/2/3] htsp_dsm_close_done *Oct 16 03:35:02.951: htsp_process_event: [0/2/3, FXOLS_PROCEEDING, E_HTSP_PROCE EDING]fxols_offhook_proc *Oct 16 03:35:02.951: htsp_timer - 120000 msec *Oct 16 03:35:02.955: htsp_pre_connect_disconnect, cdb = 46049110 cause = 26 *Oct 16 03:35:02.959: htsp_process_event: [0/2/3, FXOLS_PROCEEDING, E_HTSP_PRE_C ONN_DISC] *Oct 16 03:35:03.243: htsp_process_event: [0/2/3, FXOLS_OFFHOOK, E_DSP_SIG_0110] fxols_offhook_rvs_batteryhtsp_release_req: cause 38, no_onhook 0 *Oct 16 03:35:05.955: htsp_process_event: [0/2/3, FXOLS_CONNECT, E_HTSP_RELEASE_ REQ]fxols_offhook_release *Oct 16 03:35:05.955: htsp_timer_stop *Oct 16 03:35:05.955: htsp_timer_stop2 *Oct 16 03:35:05.955: htsp_timer_stop3 *Oct 16 03:35:05.955: [0/2/3] set signal state = 0x4 timestamp = 0 *Oct 16 03:35:05.955: dsp_set_sig_state: [0/2/3] packet_len=12 channel_id=131 pa cket_id=39 state=0x4 timestamp=0x0 *Oct 16 03:35:05.955: TGRM: reg_invoke_tgrm_call_update(0, 2, 3, 65535, 1, TGRM_ CALL_IDLE, TGRM_CALL_VOICE, TGRM_DIRECTION_IN) *Oct 16 03:35:05.955: htsp_timer - 2000 msec *Oct 16 03:35:05.955: TGRM: reg_invoke_tgrm_call_update(0, 2, 3, 65535, 1, TGRM_ CALL_BUSY, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT) *Oct 16 03:35:05.955: flex_dsprm_close_cleanup *Oct 16 03:35:06.227: htsp_process_event: [0/2/3, FXOLS_GUARD_OUT, E_DSP_SIG_011 0] *Oct 16 03:35:07.955: htsp_process_event: [0/2/3, FXOLS_GUARD_OUT, E_HTSP_EVENT_ TIMER]fxols_guard_out_timeout *Oct 16 03:35:07.955: TGRM: reg_invoke_tgrm_call_update(0, 2, 3, 65535, 1, TGRM_ CALL_IDLE, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT) *Oct 16 03:35:07.955: dsp_req_sig_state: [0/2/3] packet_len=8 channel_id=131 pac ket_id=40 *Oct 16 03:35:07.959: htsp_process_event: [0/2/3, FXOLS_ONHOOK, E_DSP_SIG_0100] # ... 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Dear All, I need to plan a Hosted CUCM VoIP solution for corporate customer as my organization has planned to move towards cloud computing, Can anyone help me find out the way to achieve this? i need to know the hardware requirement (Including switches and server) and also the appropriate way to complete this project. Currently i have Nexus 5K and 2K deployed in data center, Can i achieve it using nexus 5K? I have seen an cisco Hosted VoIP solution in an organization and i think, its very common. looking forward for rapid response.  Regards, Waqas ... View more

HI All, i need to upgrade ASA 5540 from 7.1 to 8.4 for secure connect feature of Cisco Jabber Configuration. Support forum guides that, i need to follow upgrade path from 7.1 --> 7.2  --> 8.0 --> 8.2 -->8.4 and also do a memory upgrade from 1GB to 2GB. https://supportforums.cisco.com/docs/DOC-12690 I need to use this feature for only three or maximum four users in company then would i really need to do  memory upgrade? or can i go with 1GB memory? also how i can get the prices of part number "ASA5540-MEM-2GB=" at cisco.com? Please go through the sh version of ASA and share your suggestion. ASA-ISB-HQ# sh version Cisco Adaptive Security Appliance Software Version 7.1(2) Device Manager Version 5.1(2) Compiled on Tue 14-Mar-06 17:00 by dalecki System image file is "disk0:/asa712-k8.bin" Config file at boot was "startup-config" ASA-ISB-HQ up 1 day 7 hours Hardware:   ASA5540, 1024 MB RAM, CPU Pentium 4 2000 MHz Internal ATA Compact Flash, 256MB BIOS Flash M50FW080 @ 0xffe00000, 1024KB Encryption hardware device : Cisco ASA-55x0 on-board accelerator (revision 0x0)                              Boot microcode   : CNlite-MC-Boot-Cisco-1.2                              SSL/IKE microcode: CNlite-MC-IPSEC-Admin-3.03                              IPSec microcode  : CNlite-MC-IPSECm-MAIN-2.04 0: Ext: GigabitEthernet0/0  : address is 001b.d5fb.67b2, irq 9 1: Ext: GigabitEthernet0/1  : address is 001b.d5fb.67b3, irq 9 2: Ext: GigabitEthernet0/2  : address is 001b.d5fb.67b4, irq 9 3: Ext: GigabitEthernet0/3  : address is 001b.d5fb.67b5, irq 9 4: Ext: Management0/0       : address is 001b.d5fb.67b1, irq 11 5: Int: Not licensed        : irq 11 6: Int: Not licensed        : irq 5 Licensed features for this platform: Maximum Physical Interfaces : Unlimited Maximum VLANs               : 200       Inside Hosts                : Unlimited Failover                    : Active/Active VPN-DES                     : Enabled   VPN-3DES-AES                : Enabled   Security Contexts           : 2         GTP/GPRS                    : Disabled  VPN Peers                   : 5000      WebVPN Peers                : 25        This platform has an ASA 5540 VPN Premium license. ASA-ISB-HQ# ASA-ISB-HQ# sh version | I RAM Hardware:   ASA5540, 1024 MB RAM, CPU Pentium 4 2000 MHz ASA-ISB-HQ# ASA-ISB-HQ# SH memory Free memory:       990265544 bytes (92%) Used memory:        83476280 bytes ( 8%) -------------     ---------------- Total memory:     1073741824 bytes (100%) ASA-ISB-HQ# Looking forward to reply. Thanks & Regards, Waqas ... View more

Hi Everyone, I have Integration of CUCM8.5 with Microsoft Lync 2010 through sip trunk (CUCM_Lync). currently enterprise-voice-service-enabled-users in Lync Server are able to call / dial other lync clients and IP- Phone extensions simaltanously. CUCM is integrated with LDAP and imports the user from Domain Controllers. Lync User are also able to call PSTN network from Lync client through CUCM Sip trunk and its working fine. Now when i dial an user extension from IP Phone, then only His / Her IP phone rings not the Lync Client, Can anyone help me, how to achieve this?? You all will be knowing that , we cannot transfer internal extension to Gateway / Route List, so i cannot forward call from internal_extension to another internal_extension on Gateway / Sip Trunk (CUCM_Lync) I have remote destination profiles for some user but i also dont have any DID's, so i cannot transfer call from cucm to lync-client via Remote Destination-Profile using PSTN trunk.  Please help, if there is any way to ring both IP Phones and Lync-Client simultaneously. Regards, Waqas ... View more