Re: [OSL | CCIE_Voice] Two confused questions - QoS bw CME/Gatekeeper
Much appreciated! For different sources, all refer to use 'after-compressed' to caculate cRTP. For CME, I will use dedicate dial-peer for incoming. James From: abay...@secura.com.tr To: mo...@hotmail.com CC: ccie_voice@onlinestudylist.com Date: Mon, 2 Feb 2009 16:32:52 +0200 Subject: Two confused questions - QoS bw CME/Gatekeeper Hi James, Here are my comments: Q1: when assign priority bw to voice traffic, should caculate based on before-compressed or after-comressed? Following, important sentences from Cisco Unified Communicaton SRND based on 7.X about this question http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/netstruc.html#wp1044222 Note that cRTP compression occurs as the final step before a packet leaves the egress interface; that is, after LLQ class-based queueing has occurred. Beginning in Cisco IOS Release 12.(2)2T and later, cRTP provides a feedback mechanism to the LLQ class-based queueing mechanism that allows the bandwidth in the voice class to be configured based on the compressed packet value. With Cisco IOS releases prior to 12.(2)2T, this mechanism is not in place, so the LLQ is unaware of the compressed bandwidth and, therefore, the voice class bandwidth has to be provisioned as if no compression is taking place. Q2: In CME, when received call from gatekeeper, I need incoming translation to strip 'tech-prefix'. Do I need dedicate incoming-called dial-peer for this translation, or I can reuse CME outgoing ras dial-peer? In fact, I think that is all about your choice whether to use a dedicated dial-peer or reuse another one. But I always prefer to create new dial-peers for both each incoming and outgoing call legs on every voice gateway/CME/etc. configuration. So, by this way the configuration becomes easily readable and more modular. Best Regards _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us
[OSL | CCIE_Voice] VM-integration required??
During SRST mode, do I really need 'vm-integration' to support vm for srst phones??? I think under ccm-manager-fallback already have 'voicemail ' and related dial-peer should be sufficient. And I think 'vm-integration' is additional feature for SRST to support other analog vm system which cannot take dtmf digits properly. Can any one clarify when/why need 'vm-integration'? James _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us
[OSL | CCIE_Voice] vtgo lite
hi guys ,can any 1 tell me how to use 2 instances of vtgo lite in the same time ,what is the procedure need before run on the machine _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Hi Kamal, I created additional AC Pilot with queuing disabled and pointed first one to the new AC as Alwasy Route Member. I keep on getting the MOH from first AC even though queuing timer is over. Can you pl comment if you achieved it differently? I am able to route the call to a CTI_RP as Alwasy Route Member and point this RP to a route-pattern which further points it to the Gateway. The RP string is invalid and non-routable by the GW. Using this method, the caller simply gets dropped after queuing timer is over. No Fast-busy to caller but MOH gets played. When I try pointing AC Always Route Member filed to any Route-pattern directly, I get the following message: The Directory Number you entered in the selected Partition is associated with a device that can not be a member of a Hunt Group. Pl let me know how you achieved Anthony's method? Date: Thu, 29 Jan 2009 18:05:31 +1100Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?From: lovingprin...@gmail.comto: anthony.ye...@gmail.comcc: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.comi tested it and it works great.Thanks Anthony for kind help. On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung anthony.ye...@gmail.com wrote: What you can try is assign a dummy AC pilot point to the original ACPilot Point as the 'Always Route Member' creating a linked hunt group.Then for this second dummy AC pilot point assign a dummy AC user likeyou did w/ the first. But instead, disable queuing for this seconddummy Pilot Point. After the hold time expires for the first AC pilot,the call will be forwarded to the second AC pilot. Since queuing isdisabled, the call should drop BUT w/ a disconnect cause of 'userbusy'. On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish kapilatr...@hotmail.com wrote: I did not put the TP directly inside the Hunt-Group. I put a CTIRP as Always Route Member and on CTIRP I did a forward all to the TP. I am yet to try the solution given by Christian. I'll put the call to a gateway via RP and see if I can get fast-busy to the caller after initial queuing prompt. Date: Tue, 27 Jan 2009 21:30:58 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: kapilatr...@hotmail.com CC: christian.hennr...@intact-is.com; gree...@googlemail.com; ccie_voice@onlinestudylist.com; anil...@yahoo.com I tried same way.It plays greeting only once.I also changed service parameter for Cisco TCD Allow Routing with Unknown Line State to True ,and retried.Call still doesn't end. Kapil, how did you add TP as member in HuntGroup.In my case, it gives error saying that member should be a valid DN on system.I was able to add phone/CTIRP DNs though. On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish kapilatr...@hotmail.com wrote: I tried with RP/TP Block this pattern and in that case call stays in queue. AC takes the call out of the queue only when it is routed to a registered end-point that's what I've observed. I'll try to route it to some unallocated number pointing it to the GW and see if it works. Thanks for the input. Date: Tue, 27 Jan 2009 10:39:31 +0100 From: christian.hennr...@intact-is.com To: kapilatr...@hotmail.com CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? what about routing to a number CUCM, which does not exist, or even to a PSTN number, which is unallocated? Christian Kapil Atrish schrieb: The requirement is to drop the call within CCM itself. I don't want to use Unity/IPCCX/TCL for this purpose. Date: Tue, 27 Jan 2009 09:16:49 + Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: gree...@googlemail.com To: anil...@yahoo.com CC: christian.hennr...@intact-is.com; cpar...@cparker.us; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Folks, To get the call to disconnect you can use do the following: Create a CTI RP cfwd all to voicemail. In VM create a CH with the extension number of the CTI RP and configure the greeting to be blank and then after greeting send the caller to hang up. In the ac hunt group config add the CTI RP as the always route member. In acconfig.bat for the annunicator ac pilot set the hold time to be something other than 0 seconds After this time has passed the call will be forwarded to unity and disconnected - you get a little bit of ringing as the call gets to unity which I cant get rid of. 2009/1/27 anil batra anil...@yahoo.com I too tried the way Kapil mentioned and faced same issue as he did. The call from PSTN does it the announcement but the call never gets disonncted, it seems the queue is
[OSL | CCIE_Voice] Two confused questions - QoS bw CME/Gatekeeper
Hi James, Here are my comments: Q1: when assign priority bw to voice traffic, should caculate based on before-compressed or after-comressed? Following, important sentences from Cisco Unified Communicaton SRND based on 7.X about this question http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/netstruc.html#wp1044222 Note that cRTP compression occurs as the final step before a packet leaves the egress interface; that is, after LLQ class-based queueing has occurred. Beginning in Cisco IOS Release 12.(2)2T and later, cRTP provides a feedback mechanism to the LLQ class-based queueing mechanism that allows the bandwidth in the voice class to be configured based on the compressed packet value. With Cisco IOS releases prior to 12.(2)2T, this mechanism is not in place, so the LLQ is unaware of the compressed bandwidth and, therefore, the voice class bandwidth has to be provisioned as if no compression is taking place. Q2: In CME, when received call from gatekeeper, I need incoming translation to strip 'tech-prefix'. Do I need dedicate incoming-called dial-peer for this translation, or I can reuse CME outgoing ras dial-peer? In fact, I think that is all about your choice whether to use a dedicated dial-peer or reuse another one. But I always prefer to create new dial-peers for both each incoming and outgoing call legs on every voice gateway/CME/etc. configuration. So, by this way the configuration becomes easily readable and more modular. Best Regards
[OSL | CCIE_Voice] CME DSP error
I am using PVDM2's, though I know they are PVDM1's on the lab. I create my PRI and bring up 4 channels. Configure CME completely with CUE and then add in a transcoder. When I hit the max sessions ? I have an option to do up to 6. So I want to max out my router and set it to 6. When I make an inbound call I get fast busy and outbound I get 1/2 ring and then busy. The router throws the following error: *Feb 3 02:35:42.419: %FLEXDSPRM-5-OUT_OF_RESOURCES: No dsps found either locally or globally. I am just wondering if what you can do to get around with. If I drop my max sessions down to 4 I am all set, but I want to max out my router. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Alias Static command
All, If I am using alias static command in the GK, can I use 2 alias static command one for PUB and one for SUB and how can I force the call to go to the SUB first and than via PUB. alias static 10.10.10.10 1720 gkid CCM gateway voip ras 10.10.10.10 1719 e164 2001 e164 2002 Regards Narinder CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system.
Re: [OSL | CCIE_Voice] vtgo lite
Its very easy. Go here: http://corner-il.blogspot.com/2007/07/using-multiple-ipblue-phones-on-one-pc.html Cheerz On Tue, Feb 3, 2009 at 9:58 AM, omar itani ram...@live.com wrote: hi guys ,can any 1 tell me how to use 2 instances of vtgo lite in the same time ,what is the procedure need before run on the machine -- Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it!http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Kapil, If you dial your first AC pilot # , you should hear greeting .If you dial second(dummy) AC pilot # , you should hear user busy.Now, When you link 2nd to first,i.e add 2nd AC pilot # as Always Route Member , after hold time expires, call will be routed to dummy pilot point and you will get 'user busy'.I did also change Service Parameter for TCD so that AC can route calls to directory numbers with unknown state. Regds On Mon, Feb 2, 2009 at 9:25 PM, Kapil Atrish kapilatr...@hotmail.comwrote: Hi Kamal, I created additional AC Pilot with queuing disabled and pointed first one to the new AC as Alwasy Route Member. I keep on getting the MOH from first AC even though queuing timer is over. Can you pl comment if you achieved it differently? I am able to route the call to a CTI_RP as Alwasy Route Member and point this RP to a route-pattern which further points it to the Gateway. The RP string is invalid and non-routable by the GW. Using this method, the caller simply gets dropped after queuing timer is over. No Fast-busy to caller but MOH gets played. When I try pointing AC Always Route Member filed to any Route-pattern directly, I get the following message: The Directory Number you entered in the selected Partition is associated with a device that can not be a member of a Hunt Group. Pl let me know how you achieved Anthony's method? -- Date: Thu, 29 Jan 2009 18:05:31 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: anthony.ye...@gmail.com CC: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com I tested it and it works great.Thanks Anthony for kind help. On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung anthony.ye...@gmail.comwrote: What you can try is assign a dummy AC pilot point to the original AC Pilot Point as the 'Always Route Member' creating a linked hunt group. Then for this second dummy AC pilot point assign a dummy AC user like you did w/ the first. But instead, disable queuing for this second dummy Pilot Point. After the hold time expires for the first AC pilot, the call will be forwarded to the second AC pilot. Since queuing is disabled, the call should drop BUT w/ a disconnect cause of 'user busy'. On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish kapilatr...@hotmail.com wrote: I did not put the TP directly inside the Hunt-Group. I put a CTIRP as Always Route Member and on CTIRP I did a forward all to the TP. I am yet to try the solution given by Christian. I'll put the call to a gateway via RP and see if I can get fast-busy to the caller after initial queuing prompt. Date: Tue, 27 Jan 2009 21:30:58 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: kapilatr...@hotmail.com CC: christian.hennr...@intact-is.com; gree...@googlemail.com; ccie_voice@onlinestudylist.com; anil...@yahoo.com I tried same way.It plays greeting only once.I also changed service parameter for Cisco TCD Allow Routing with Unknown Line State to True ,and retried.Call still doesn't end. Kapil, how did you add TP as member in HuntGroup.In my case, it gives error saying that member should be a valid DN on system.I was able to add phone/CTIRP DNs though. On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish kapilatr...@hotmail.com wrote: I tried with RP/TP Block this pattern and in that case call stays in queue. AC takes the call out of the queue only when it is routed to a registered end-point that's what I've observed. I'll try to route it to some unallocated number pointing it to the GW and see if it works. Thanks for the input. Date: Tue, 27 Jan 2009 10:39:31 +0100 From: christian.hennr...@intact-is.com To: kapilatr...@hotmail.com CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? what about routing to a number CUCM, which does not exist, or even to a PSTN number, which is unallocated? Christian Kapil Atrish schrieb: The requirement is to drop the call within CCM itself. I don't want to use Unity/IPCCX/TCL for this purpose. Date: Tue, 27 Jan 2009 09:16:49 + Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: gree...@googlemail.com To: anil...@yahoo.com CC: christian.hennr...@intact-is.com; cpar...@cparker.us; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Folks, To get the call to disconnect you can use do the following: Create a CTI RP cfwd all to voicemail. In VM create a CH with the extension number of the CTI
[OSL | CCIE_Voice] DTMF digits from PSTN callers are not recognized in the Unity Greeting
Hi, When dialing from BR1 into Unity , calls and digit input work .During AAR/SRST , callers dialing via PSTN are able to leave voice-mail messages.However, when subscriber dials directly into unity during AAR/SRST, unity plays subscriber greeting but it doesn't recognize any input dtmf digit inputs.As per support Wiki, call-handlers are working but Unity Subscriber Sign-In fails.Any one else had same problem ? h ttp://supportwiki.cisco.com/ViewWiki/index.php/DTMF_digits_from_PSTN_callers_are_not_recognized_in_the_greeting,_and_Call_Handler_and_callers_are_not_authenticated_on_Cisco_Unityhttp://supportwiki.cisco.com/ViewWiki/index.php/DTMF_digits_from_PSTN_callers_are_not_recognized_in_the_greeting,_and_Call_Handler_and_callers_are_not_authenticated_on_Cisco_Unity
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Yeah..I did that but putting DN didn't work.You would need Secondary AC pilot #. Besides, I prefer to use Unity rather than going through this method.At least for lab, it won't be advisable unless strictly asked to do so. On Tue, Feb 3, 2009 at 5:02 PM, Kapil Atrish kapilatr...@hotmail.comwrote: Cool...I did not check for the TCD Service Parameter. I think if I set this parameter the second AC would not be required. I may simply put a DN as Always route member to extend fast busy to caller after initial MOH. Otherwise I'll also follow your solution. Vik/Mark: Do you think it is an acceptable solution? Question is to customize annunciator and we are using MOH to acheive the results? -- Date: Tue, 3 Feb 2009 15:39:43 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: kapilatr...@hotmail.com CC: anthony.ye...@gmail.com; ccie_voice@onlinestudylist.com; gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com Kapil, If you dial your first AC pilot # , you should hear greeting .If you dial second(dummy) AC pilot # , you should hear user busy.Now, When you link 2nd to first,i.e add 2nd AC pilot # as Always Route Member , after hold time expires, call will be routed to dummy pilot point and you will get 'user busy'.I did also change Service Parameter for TCD so that AC can route calls to directory numbers with unknown state. Regds On Mon, Feb 2, 2009 at 9:25 PM, Kapil Atrish kapilatr...@hotmail.comwrote: Hi Kamal, I created additional AC Pilot with queuing disabled and pointed first one to the new AC as Alwasy Route Member. I keep on getting the MOH from first AC even though queuing timer is over. Can you pl comment if you achieved it differently? I am able to route the call to a CTI_RP as Alwasy Route Member and point this RP to a route-pattern which further points it to the Gateway. The RP string is invalid and non-routable by the GW. Using this method, the caller simply gets dropped after queuing timer is over. No Fast-busy to caller but MOH gets played. When I try pointing AC Always Route Member filed to any Route-pattern directly, I get the following message: The Directory Number you entered in the selected Partition is associated with a device that can not be a member of a Hunt Group. Pl let me know how you achieved Anthony's method? -- Date: Thu, 29 Jan 2009 18:05:31 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: anthony.ye...@gmail.com CC: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com I tested it and it works great.Thanks Anthony for kind help. On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung anthony.ye...@gmail.comwrote: What you can try is assign a dummy AC pilot point to the original AC Pilot Point as the 'Always Route Member' creating a linked hunt group. Then for this second dummy AC pilot point assign a dummy AC user like you did w/ the first. But instead, disable queuing for this second dummy Pilot Point. After the hold time expires for the first AC pilot, the call will be forwarded to the second AC pilot. Since queuing is disabled, the call should drop BUT w/ a disconnect cause of 'user busy'. On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish kapilatr...@hotmail.com wrote: I did not put the TP directly inside the Hunt-Group. I put a CTIRP as Always Route Member and on CTIRP I did a forward all to the TP. I am yet to try the solution given by Christian. I'll put the call to a gateway via RP and see if I can get fast-busy to the caller after initial queuing prompt. Date: Tue, 27 Jan 2009 21:30:58 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: kapilatr...@hotmail.com CC: christian.hennr...@intact-is.com; gree...@googlemail.com; ccie_voice@onlinestudylist.com; anil...@yahoo.com I tried same way.It plays greeting only once.I also changed service parameter for Cisco TCD Allow Routing with Unknown Line State to True ,and retried.Call still doesn't end. Kapil, how did you add TP as member in HuntGroup.In my case, it gives error saying that member should be a valid DN on system.I was able to add phone/CTIRP DNs though. On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish kapilatr...@hotmail.com wrote: I tried with RP/TP Block this pattern and in that case call stays in queue. AC takes the call out of the queue only when it is routed to a registered end-point that's what I've observed. I'll try to route it to some unallocated number pointing it to the GW and see if it works. Thanks for the input. Date: Tue, 27 Jan 2009 10:39:31 +0100 From:
