Re: [OSL | CCIE_Voice] Gatekeeper Call Routing
Hi , Gatekeeper will route the call without Zone prefix if all the endpoints are registered in the same zone. Thanks From: datucha123 datucha123 datucha...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Thursday, November 17, 2011 4:30 PM Subject: [OSL | CCIE_Voice] Gatekeeper Call Routing Hello everyone. I have a question about the GK. I could not understand how does the GK knows where to route call based on the following GK configuration: gatekeeper zone local test test.com 177.1.254.1 gw-type-prefix 1#* default-technology no shutdown CUCME and CUCM, both are registered with the Tech prefix: R1#sh gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 1#* (Default gateway-technology) Zone test master gateway list: 177.1.254.3:1720 CME 172.16.4.121:35498 CM_2 172.16.4.120:37425 CM_1 So the CUCME has extensions 3... and CUCM has extensions 2... Basically, the calls are working fine between CUCM and CUCME through this GK. But I cannot get the idea, how does the GK know where to route calls (CUCME and CUCM IP Phones are not registered with GK). For instance, when the CUCM calls number 3003 (this is CUCME IP Phone), the ARQ request is sent to GK, and at this point, how does the GK knows whos IP address to return in ACF message to CUCM? How does the GK get that the the CUCME IP address must be sent to CUCM? There are not zones prefixes configured. Can anybody explain that to me? ___ 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
[OSL | CCIE_Voice] CCIE Voice Passed
Hi All, Thanks to all for the support knowledge sharing. I passed my CCIE Voice Lab . Thanks Again___ 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] CCIE Voice Lab Grading
Hi , Any thoughts or suggestions on this Thanks From: mgscip gpsvoiceexpe...@yahoo.com To: ccie ccie_voice@onlinestudylist.com Sent: Tuesday, October 4, 2011 11:31 PM Subject: CCIE Voice Lab Grading Hi All, I appeared lab exam last week my result was fail.I just want to know that what are the fine tuning we have to do to fulfill CISCO requirement. In Media Resources section i got 3 question i fulfill the requirements . I got 40 % on that section . please advise the best practices recommendations are we have to taken care ? 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
[OSL | CCIE_Voice] SRST Behaviour
Hi , I have some issue in SRST . When the Phones are get into SRST fallback-mode Phones didn't get any DN. I given the SRST mode auto-provision all , but i couldn't see any Ephone configuration in the running configuration. tried with Firmware upgrade , Reload the router but no luck. Thanks, Sriram.P___ 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] CCIE Voice Lab Grading
Hi All, Anyone can guide on this ! Thanks... From: mgscip gpsvoiceexpe...@yahoo.com To: ccie ccie_voice@onlinestudylist.com Sent: Tuesday, October 4, 2011 11:31 PM Subject: [OSL | CCIE_Voice] CCIE Voice Lab Grading Hi All, I appeared lab exam last week my result was fail.I just want to know that what are the fine tuning we have to do to fulfill CISCO requirement. In Media Resources section i got 3 question i fulfill the requirements . I got 40 % on that section . please advise the best practices recommendations are we have to taken care ? 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
[OSL | CCIE_Voice] CCIE Voice Lab Grading
Hi All, I appeared lab exam last week my result was fail.I just want to know that what are the fine tuning we have to do to fulfill CISCO requirement. In Media Resources section i got 3 question i fulfill the requirements . I got 40 % on that section . please advise the best practices recommendations are we have to taken care ? 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] CME Conference mode local only
Hi , Keep conference only with Software conference ( 3 Party Conference -- Max conference ) Conference drop mode for Hardware conference ( conference Hardware).In the hardware conference mode Keep conference command didn't take any effect. Thanks. From: Bill Lake whl...@gmail.com To: Ken Wyan kew...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Thursday, September 1, 2011 10:49 PM Subject: Re: [OSL | CCIE_Voice] CME Conference mode local only keep-conf works in both ephone/ephone-template and voice register pool keep conf has more flexibility Here are some examples Examples In the following example, extension 3555 initiates a three-way conference. After the conference is established, extension 3555 can press the Confrn soft key to disconnect the last party that was connected and remain connected to the first party that was connected. If extension 3555 hangs up from the conference, the other two parties remain connected if one of them is local to the Cisco Unified CME system. ephone-dn 35 number 3555 ephone 24 button 1:35 keep-conference drop-last local-only In the following example, extension 3666 initiates a three-way conference. After the conference is established, extension 3666 can press the Confrn soft key to disconnect the last party that was connected and remain connected to the first party that was connected. Also, extension 3666 can hang up from a three-way conference to terminate the conference and disconnect all parties or can press the EndCall soft key to leave the conference and keep the other two parties connected. ephone-dn 36 number 3666 ephone 25 button 1:36 keep-conference drop-last endcall In the following example, extension 3777 initiates a three-way conference. After the conference is established, extension 3777 can press the Confrn soft key to disconnect the last party that was connected and remain connected to the first party that was connected. Also, extension 3777 can hang up from a three-way conference to terminate the conference and disconnect all parties or press the EndCall soft key to leave the conference and keep the other two parties connected only if one of the two parties is local to the Cisco Unified CME system. ephone-dn 38 number 3777 ephone 27 button 1:38 keep-conference drop-last endcall local-only In the following example, extension 3999 initiates a three-way conference. After the conference is established, extension 3999 can hang up to terminate the conference and disconnect all parties or press the EndCall soft key to leave the conference and keep the other two parties connected only if one of the two parties is local to the Cisco Unified CME system. ephone-dn 39 number 3999 ephone 29 button 1:39 keep-conference endcall local-only conference drop-mode To configure the mode for terminating ad hoc hardware conferences when parties drop out, use the conference drop-mode command in ephone or ephone-template configuration mode. To return to the default, use the no form of this command. conference drop-mode [creator | local] no conference drop-mode [creator | local] Syntax Description creator Specifies that the active conference terminates when the creator hangs up. local Specifies that the active conference terminates when the last local party in the conference hangs up or drops out of the conference. Command Default The conference is not dropped, regardless of whether the creator hangs up, provided three parties remain in the conference. Command Modes Ephone configuration (config-ephone) Ephone-template configuration (config-ephone-template) keep-conference To allow conference initiators to exit from conference calls and to either end or maintain the conference for the remaining parties, use the keep-conference command in ephone or ephone-template configuration mode. To return to the default, use the no form of this command. keep-conference [drop-last] [endcall] [local-only] no keep-conference Syntax Description drop-last (Optional) The action of the Confrn soft key is changed; the conference initiator can press the Confrn soft key (IP phone) or hookflash (analog phone) to drop the last party. Note Analog phones connected to the Cisco Unified CME system through a Cisco VG 224 require Cisco IOS Release 12.3(11)YL1 or a later release to use this feature. endcall (Optional) The action of the EndCall soft key is changed; the conference initiator can hang up or press the EndCall soft key to leave the conference and keep the other two parties connected. Note If this option is not enabled, pressing the EndCall soft key terminates the conference and disconnects all parties. local-only (Optional) The conference initiator can hang up to end the conference and leave the other two parties connected only if one of the remaining parties is local to the Cisco Unified CME
Re: [OSL | CCIE_Voice] MVA - CSS Selection
Hi All, Please clarify on this ? Thanks From: mgscip gpsvoiceexpe...@yahoo.com To: ccie voice ccie_voice@onlinestudylist.com Sent: Saturday, August 27, 2011 11:36 PM Subject: [OSL | CCIE_Voice] MVA - CSS Selection Hi All, I have doubt on MVA calling search selection. My understand on the below service parameters is Calling number matches the Destination number check the service parameter it will choose the CSS ( Trunk or GW , Line + Dest.Profile) If Calling number matches didn't match the destination number , it always choose the GW or Trunk CSS. Inbound Calling Search Space for Remote Destination: Calling Search Space for Remote Destination:This parameter specifies the calling search space (CSS) that Cisco Unified Communications Manager (Unified CM) utilizes to route an incoming call from a configured Remote Destination. Valid values specify Trunk or Gateway Inbound Calling Search Space (Unified CM uses the inbound calling search space of the trunk or gateway from which the call arrived) or Remote Destination Profile + Line Calling Search Space (Unified CM uses the concatenation of the calling search spaces on the line and Remote Destination profile associated with the remote destination that was matched). Calls that do not match a Remote Destination are not affected by this parameter because they always use the trunk or gateway inbound CSS. For calls that come from a Remote Destination (the calling party number matches the Remote Destination number), choose Remote Destination Profile + Line Calling Search Spaces to use those calling search spaces to route the call instead of using the Trunk/Gateway Calling Search Space. The digits that come from the trunk or gateway must be formatted in a way that can be dialed using the Remote Destination Profile + Line Calling Search Spaces. For me in both case it selects the Line + Remote Destination profile CSS .During my test kept the service parameter as Trunk or Gatway CSS. ___ 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
[OSL | CCIE_Voice] Translation Pattern with Device Based CSS
Hi Mates, I have tried with Device based CSS approach during the practice. Line CSS was none. I kept the patterns in both tranaslation pattern Route pattern. from the Device CSS i can't access the translation pattern numbers . if the configure the same way in the route pattern i can able to dial. 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
[OSL | CCIE_Voice] MVA - CSS Selection
Hi All, I have doubt on MVA calling search selection. My understand on the below service parameters is Calling number matches the Destination number check the service parameter it will choose the CSS ( Trunk or GW , Line + Dest.Profile) If Calling number matches didn't match the destination number , it always choose the GW or Trunk CSS. Inbound Calling Search Space for Remote Destination: Calling Search Space for Remote Destination:This parameter specifies the calling search space (CSS) that Cisco Unified Communications Manager (Unified CM) utilizes to route an incoming call from a configured Remote Destination. Valid values specify Trunk or Gateway Inbound Calling Search Space (Unified CM uses the inbound calling search space of the trunk or gateway from which the call arrived) or Remote Destination Profile + Line Calling Search Space (Unified CM uses the concatenation of the calling search spaces on the line and Remote Destination profile associated with the remote destination that was matched). Calls that do not match a Remote Destination are not affected by this parameter because they always use the trunk or gateway inbound CSS. For calls that come from a Remote Destination (the calling party number matches the Remote Destination number), choose Remote Destination Profile + Line Calling Search Spaces to use those calling search spaces to route the call instead of using the Trunk/Gateway Calling Search Space. The digits that come from the trunk or gateway must be formatted in a way that can be dialed using the Remote Destination Profile + Line Calling Search Spaces. For me in both case it selects the Line + Remote Destination profile CSS .During my test kept the service parameter as Trunk or Gatway CSS. ___ 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
[OSL | CCIE_Voice] FRTS Vs Class Based Shapping
Hi All, I'm very much confused about shaping in the frame relay ( Difference between FRTS Class Based Shape) How the shaping decision happening in the interface. Please help on this___ 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] CUE MWI in CME as SRST
Hi All, MWI start working . I did nothing expect restart of my lab. From: Kshitij Singhi martinian.ksin...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Wednesday, August 10, 2011 8:58 PM Subject: Re: [OSL | CCIE_Voice] CUE MWI in CME as SRST Forgot to add: Also make sure that the gateway address is correct in ccn subsystem sip on the CUE. On Wed, Aug 10, 2011 at 8:18 PM, ccie_voice-requ...@onlinestudylist.com wrote: 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 MWI in CME as SRST (mgscip) 2. Re: CUE MWI in CME as SRST (Ashraf Ayyash) 3. RSVP issue between HQ BR1 (Geoghegan, Stuart) 4. Re: RSVP issue between HQ BR1 (Brian Mulgrew) -- Message: 1 Date: Tue, 9 Aug 2011 21:46:25 -0700 (PDT) From: mgscip gpsvoiceexpe...@yahoo.com To: ccie ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUE MWI in CME as SRST Message-ID: 1312951585.23894.yahoomail...@web43512.mail.sp1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 H All, I'm testing with ?CUE with CME SRST Mode facing the issue in MWI. configuration as below telephony-service srst dn template 1 mwi realy sip-ua mwi server ipv4:10.0.0.20 unsolicited ephone-dn template 1 mwi sip When i checked the show ccn sip mwi subscription it says no 0 subscription for MWI. I have ensured that MWI method configured as unsolicited in CUE. Tried with resload of CUE tooo , no go :( Thanks -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110809/ddac40be/attachment-0001.html -- Message: 2 Date: Wed, 10 Aug 2011 11:16:01 +0300 From: Ashraf Ayyash ash.ayy...@gmail.com To: mgscip gpsvoiceexpe...@yahoo.com Cc: ccie ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CUE MWI in CME as SRST Message-ID: CAEW==nsPXVABU50Vt6Pts7c8Sqd4=qnxubwyvksthsgt_0k...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Hi , can you collect deb ccsip message and voip dialpeer from your CME ? show run from CUE/CME Thanks Ash On Wed, Aug 10, 2011 at 7:46 AM, mgscip gpsvoiceexpe...@yahoo.com wrote: H All, I'm testing with ?CUE with CME SRST Mode facing the issue in MWI. configuration as below telephony-service srst dn template 1 mwi realy sip-ua mwi server ipv4:10.0.0.20 unsolicited ephone-dn template 1 mwi sip When i checked the show ccn sip mwi subscription it says no 0 subscription for MWI. I have ensured that MWI method configured as unsolicited in CUE. Tried with resload of CUE tooo , no go :( 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 -- Message: 3 Date: Wed, 10 Aug 2011 15:13:51 +0100 From: Geoghegan, Stuart stuart.geoghe...@ngbailey.co.uk To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] RSVP issue between HQ BR1 Message-ID: bbeca2fd86e3284785e0a95b247cb0480d9cbb1...@thor.bailey.pri Content-Type: text/plain; charset=us-ascii Hi All, I have been experiencing an issue whilst attempting to configure the RSVP question 10.1 in the volume 1 work books. The question requests that you configure RSVP Locations between sites HQ BR1 allowing only 2 calls. I have followed the proctor guide and have configured a SCCP controlled software MTP per site. On both HQ-RTR and BR1-RTR I have configured the dspfarm profile whereby I have specified Codec pass-through, g729r8 and rsvp. BR1-RTR's MTP is registered to the Subscriber and assigned to a BR1 specific MRG, which in turn is assigned to a BR1 specific MRGL. This MRGL is assigned to the BR1 Device Pool (and also the individual phones at BR1-Site). The same is true for the HQ-RTR MTP - it is assigned HQ specific MRG, which in turn is assigned to a HQ specific MRGL. This MRGL is assigned to the HQ Device Pool (and also the individual phones at HQ-Site). I have changed my previously static Locations based CAC for HQ to BR1 to unlimited for audio with the RSVP setting to be Mandatory. Obviously this replicates the config for BR1 to HQ under BR1's Locations configuration. This is assigned to the relevant device pools. My MGCP BR1 gateway has the correct BR1 MRGL
[OSL | CCIE_Voice] CUE MWI in CME as SRST
H All, I'm testing with CUE with CME SRST Mode facing the issue in MWI. configuration as below telephony-service srst dn template 1 mwi realy sip-ua mwi server ipv4:10.0.0.20 unsolicited ephone-dn template 1 mwi sip When i checked the show ccn sip mwi subscription it says no 0 subscription for MWI. I have ensured that MWI method configured as unsolicited in CUE. Tried with resload of CUE tooo , no go :( 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
[OSL | CCIE_Voice] Gatekeeper with CUBE
Hi All, I'm testing with Gatekeeper with CUBE. When i make call from HQ to site C through Gatekeeper , HQ Phone keep ringing even the call answered in the Site C. I have unchecked the Wait for TCS in the Gatekeeper trunk . but the results are same. How to proceed further on this ? 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
[OSL | CCIE_Voice] Difference between sccp local and bind interface
Hi All, What is the difference it going to make by giving the bind interface command in sccp ccm group. ex : sccp local loopback 0 sccp ccm group 1 bind interface loopback 0. 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] srst calling name
HI Experts , Please advice on this. From: donny f f.faraday...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Monday, July 11, 2011 11:27 AM Subject: [OSL | CCIE_Voice] srst calling name hi all, Assuming we been told to send calling name SiteC ph 2 in SRST mode call to PSTN. How do we achieve it it, if we ask to only use srst autoprovision all and srst autoprovision dn cause evertyme after u change Name to SiteC ph 2 from +6132 When you switch to UCM mode and back to SRST, it will overide tks for advice ___ 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
[OSL | CCIE_Voice] Privacy Button in CME SRST
Hi All, Hi , In CME SRST mode , we configured the privacy button in the SRST ephone template. Privacy button configured in the CM mode also. Even though enough of button available in the phone we couldn't get the privacy button ? How to proceed further on this ?___ 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
[OSL | CCIE_Voice] BLF Speed dial in SRST
Hi All, Is there any other way to preserve the BLF Key in SRST mode ? 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
[OSL | CCIE_Voice] CME/CUE - MWI Unsolicited
Hi All, In CUE Unsolicited MWI method mwi not working for sip phone working for sccp phone. Getting the following error message debug ccsip sip 481 Call leg/Transaction does not exit In the same time if we configured the MWI under the Voice Register Dn it's working fine. Is the default behaviour of unsolicited method ? We are using 7942 phones CME is 7.0 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
[OSL | CCIE_Voice] CUE - Voice Mail Access from SIP Phone
Hi All, When i dial the Voicemail pilot number from SIP Phone dtmf is not recognized by the CUE . once enabled the dtmf-relay rtp-nte in the voice register pool it's start working. is the way it's working ? ___ 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] B-ACD Not working
Hi , Thanks a lot. Once i give the param voice-mail command it's working. Could you please tell us what are the mandatory parameter we have to configure for ACD AA application. From: Rogers Ochieng rogersochi...@gmail.com To: mgscip gpsvoiceexpe...@yahoo.com Cc: ccie ccie_voice@onlinestudylist.com Sent: Wed, April 13, 2011 8:30:20 PM Subject: Re: [OSL | CCIE_Voice] B-ACD Not working param voice-mail is mandatory even if you are not sending the call to voice mail, you can configure a hunt pilot number or dn number On 13 April 2011 16:35, mgscip gpsvoiceexpe...@yahoo.com wrote: Hi , We tested with B-ACD in CME . whenever we dial the pilot number call disconnect. Config application service aa flash:app-b-acd-aa-3.0.0.2.tcl paramspace english index 1 param menu-timeout 1 param dial-by-extension-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 7500 paramspace english location flash: param second-greeting-time 1 param welcome-prompt _bacd_welcome.au paramspace english prefix en param service-name ACD service ACD flash:app-b-acd-3.0.0.2.tcl paramspace english language en paramspace english index 0 param aa-hunt1 7001 param aa-hunt2 7002 param number-of-hunt-grps 4 param aa-hunt3 7003 Dial-peer dial-peer voice 7500 voip service aa destination-pattern 7500 session target ipv4:192.168.1.100 incoming called-number 7500 dtmf-relay h245-alphanumeric codec g711ulaw I have verified that all the audio files uploade in the flash. MoH working for IP Phones. When i check show call application session , it tries to establish session but it end session immediately ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] B-ACD Not working
Hi , We tested with B-ACD in CME . whenever we dial the pilot number call disconnect. Config application service aa flash:app-b-acd-aa-3.0.0.2.tcl paramspace english index 1 param menu-timeout 1 param dial-by-extension-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 7500 paramspace english location flash: param second-greeting-time 1 param welcome-prompt _bacd_welcome.au paramspace english prefix en param service-name ACD service ACD flash:app-b-acd-3.0.0.2.tcl paramspace english language en paramspace english index 0 param aa-hunt1 7001 param aa-hunt2 7002 param number-of-hunt-grps 4 param aa-hunt3 7003 Dial-peer dial-peer voice 7500 voip service aa destination-pattern 7500 session target ipv4:192.168.1.100 incoming called-number 7500 dtmf-relay h245-alphanumeric codec g711ulaw I have verified that all the audio files uploade in the flash. MoH working for IP Phones. When i check show call application session , it tries to establish session but it end session immediately___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CME - Presence Call list
Hi All We not able to get the Presence call list for only one phone. We enable the presence call list globally ephone also. It's working in others phone. Tried with reboot of router but no luck. BLF -Speed dial is working fine. Thanks in Advance___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
