Re: [OSL | CCIE_Voice] UCCX agent
When the agent goes to reserved state on the phone, it means that UCCX has identified the agent as a resource and it attempting to route the call to the agent. It is normal for the agent to enter reserved as that is how the system prevents other CSQ or applications from trying to route a call to the same resource. The issue could be with the calling search spaces on your CTI ports or some other type of routing problem that is preventing the call from cutting through to the phone. The call will remain in queue until it is routed. Scott Hardesty | Solutions Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | sharde...@presidio.commailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | www.presidio.comhttp://www.presidio.com/ [http://www.presidio.com/images/presidio_logo.gif] From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder Sent: Monday, October 05, 2009 5:59 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX agent No fancy script just testing IPCC using the standard ICD script. As soon as I dial into the ICD RP my agent goes into the reserve state and stay in reserve as long as the call is in Q. Checked the configuration and everything looks fine, any idea what's wrong ? Thanks 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 UXC 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 UXC Getronics Australia. While we endeavour to protect our network from computer viruses, UXC 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. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Choosing the right ISR?
Another consideration is the calls per second for each gateway. Just because you can terminate 4 PRI circuits on a single ISR that does not mean that it is non-blocking. I installed a call center solution a few years back using 2821 router w/ 3 xT1 connections in each. The inbound call volume crashed the routers because of the CPU interrupts from the ISDN signaling. We ended up moving to 3845 routers to terminate the 3xT1 connections to support the call volume. For standard office environments this should not be an issue but be careful if you are working at a location that has HIGH call volumes. Scott Hardesty | Solutions Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | sharde...@presidio.commailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | www.presidio.comhttp://www.presidio.com/ [http://www.presidio.com/images/presidio_logo.gif] From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of shikamaru Sent: Thursday, September 24, 2009 11:45 AM To: groganhockey Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] Choosing the right ISR? Haha, I can do better than that! I've downloaded it! ;) On Thu, Sep 24, 2009 at 8:19 AM, groganhockey groganhoc...@gmail.commailto:groganhoc...@gmail.com wrote: I'm just glad I can finally contribute *something* to these discussions! :) FYI, cisco has moved the doc in the past, so make sure you remember the title in case it moves again. mike On Thu, Sep 24, 2009 at 9:38 AM, shikamaru shikam...@kagadis.commailto:shikam...@kagadis.com wrote: MUCH respect, Mike. This is the perfect document for this kind of question. Thank you. On Wed, Sep 23, 2009 at 7:29 PM, mike deal groganhoc...@gmail.commailto:groganhoc...@gmail.com wrote: I've used this document in the past for sizing purposes: http://www.cisco.com/en/US/prod/collateral/routers/ps259/product_data_sheet0900aecd8057f2e0.pdf mike On Wed, Sep 23, 2009 at 7:59 PM, Nara Shikamaru shikam...@kagadis.commailto:shikam...@kagadis.com wrote: I had no idea there was a PRI limit. I was thinking, potentially, I may need to terminate 8 PRIs on a 2811 but in truth I'm planning on having 3 2811 for redundancy and spread the span against all three. Plenty of ports between them. I guess my question was also whether the 2811 can handle this kind of scenario, but then if it couldn't I don't think Cisco would allow for 4 PRIs to be terminated to it. I'll ask my AM tomorrow. Thanks, Michael. On Wed, Sep 23, 2009 at 5:26 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: Each ISR router is supposed to only be able to handle X number of PRIs (not physical, more CPU / resource load wise.) I would work with your Cisco AM to have them help you detemine what the limits and loading are. I can't find what documents discussed it. I know I came across a third-party testing report (Mircom maybe.) that had like max 4 PRIs on a 2811. My number might be off, but there was a limit. That's why I would suggest working with your Cisco AM--they should be able to help with those numbers. If you are a partner, the PDI helpdesk should be able to help. If not, then that's what the AM will help you with. Not sure if TAC would assist with these design questions, but you can always try. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nara Shikamaru [shikam...@kagadis.commailto:shikam...@kagadis.com] Sent: Wednesday, September 23, 2009 12:01 PM To: OSL Group Subject: [OSL | CCIE_Voice] Choosing the right ISR? Okay, my question is not really out of the modules, just a question about a real world scenario. I'm preparing to increase the size of our VoIP network and am aware of the principle differences between the ISRs. Our remote sites will have subscribers, so SRST is not really an issue, and the ISRs are only being used to terminate PRIs and will not be used to route data VLAN traffic. This being the case, are there caveats to using 2811 routers with 8 VWIC ports? I don't really know what to expect by way of offnet traffic, but have had success with the 2811 line and am wondering if I can repurpose for the new network and not have too much to worry about. Also, I am planning on configuring some hardware conferencing but I have no idea yet how popular it will be, no transcoding is planned as our sites are currently all on G711. -- -Shikamaru -- -Shikamaru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- -Shikamaru ___ For more information regarding industry leading
Re: [OSL | CCIE_Voice] Translation Problem
Please explain what result you are looking for and I think we can help.. Scott Hardesty | Solutions Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot Sent: Monday, June 08, 2009 7:10 PM To: Saud Azar Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Translation Problem You are all over the place. What is 5900? Are you trying to translate ANI number 5900 to 0845444? What is this translate-outgoing called 1? On Mon, Jun 8, 2009 at 4:29 PM, Saud Azarsauda...@hotmail.com wrote: Hi, I spent countless hours and i am hoping someone can help me. I have gone through documents, videos but it doesnt make sense. Basically i have a created a simple voice translation profile below is an example voice translation-rule 1 rule 1 /^5900/ /0845444/ voice translation-profile outgoingcall translate calling 1 ephone-dn 1 dual-line number 2300 translate calling 1 i tested the rule and it works. Unfortunately when i dial from IP phone it says peer not found. I created the dial peer for 0845 pots dial-peer voice 5900 pots translation-profile outgoing outgoingcall destination-pattern 0800... translate-outgoing called 1 port 0/2/1:15 forward-digits all I am not sure if i should create the dial peer for the extension or the outgoing number. I would be greatful if someone can help me. Thanks. Upgrade to Internet Explorer 8 Optimised for MSN. Download Now
Re: [OSL | CCIE_Voice] Open lab dates in SJ NOW
If anyone is interested in dropping their v2 lab date, please let me know directly. Thanks. Scott Hardesty | Solutions Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | sharde...@presidio.commailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | www.presidio.comhttp://www.presidio.com/ [http://www.presidio.com/images/presidio_logo.gif] From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cliff McGlamry Sent: Thursday, April 16, 2009 10:15 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Open lab dates in SJ NOW There are some V2 lab dates open in San Jose right now.
[OSL | CCIE_Voice] What is the official date for the new lab format
All, do you know what date the new lab format will become active? The last I heard it was going to be July but I don't know if Cisco has published a specific date for the new format. If anyone has this date, let me know. Thanks. Scott Hardesty | Solutions Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/
Re: [OSL | CCIE_Voice] CTI RP
Narinder, I usually get a message stating a system problem when I hit IPCC and there is something wrong with my script / configurations. I have received busy signals when I have had issues with calling search spaces on my CTI ports or CTI route points. If the ports are registered, I would take a look at the CSS on your CTI/RP ports. Scott Hardesty | Solutions Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder Sent: Wednesday, March 04, 2009 5:15 PM To: Vik Malhi; Cliff McGlamry; OSLGroup Subject: Re: [OSL | CCIE_Voice] CTI RP No Need of transcoder all G711. All the points cliff has suggested is already checked and are in place. I removed my script and used aa.aef no change. Port conflict checked. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vik Malhi Sent: Thursday, 5 March 2009 4:35 AM To: Cliff McGlamry; OSL Group Subject: Re: [OSL | CCIE_Voice] CTI RP In addition to what Cliff has said- also ensure the JTAPI and RM-CM Subsystem's are showing as active in the CRS control center. Call from the HQ phone where there is no need for a transcoder to be invoked and you do not need any Location bandwidth available. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Cliff McGlamry cl...@mcglamry.net Date: Wed, 4 Mar 2009 10:52:23 -0500 To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CTI RP In my experience, when this happens the problem is usually that there is something wrong with the script. If you open up the script you are using with the script editor, and use the validate function, it will tell you if there is something it doesn't like. If the script has validation problems, it will NOT answer the phone. You can also plug in one of your canned scripts to see if they work (the aa.aef or icd.aef). If they don't answer, you're likely missing some configuration. Remember that in addition to the CTI Route point, you must also configure the CTI Ports themselves (the call control groupthese are the actual ports that are answering the phone) and the Cisco Media Termination Dialog group (Under Cisco Media on the subsystems menu). The Media Termination group is what provides the ability to interact with the caller (i.e. speak to them get digits from them, etc). If these are not set up, then you can't configure them onto the JTAPI trigger. And if they aren't configured on the JTAPI trigger (under the settings for Call Control Group and Primary Dialog Group), then a script that needs to answer the phone and interact with callers.can't. So, you'd likely get a busy signal. On a more basic setting, you could bounce the CRS Node engine, make sure the services are upand if you're also running extension mobility/IPMA Console/etc. you need to change the port number in Tomcat on IPCC to fix the conflict there. HTH Cliff - Original Message - From: Kumar, Narinder narinder.ku...@uxcg.com.au To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 04, 2009 8:51 AM Subject: Re: [OSL | CCIE_Voice] CTI RP I integrated IPCC with CCM. IPCC and CCM are both collocated on the same box. Created new application with trigger point 1700 ON the CCM PUB and SUB the CTI RP shows me it is registered. But when I call 1700 from the IP Phone or from PSTN it keeps coming busy all time. Rest the CTI RP, reset the IIS server ( Don't think that will play any part), I even rebooted both PUB and SUB, don't know why it is busy all the time. Any help is much appreciated. Thanks 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
Re: [OSL | CCIE_Voice] GDM
Kumar, did you ever find a solution to your GDM configuration? I have found the same as you. Members seem to have the ability to listen / delete messages. Scott Hardesty | Solutions Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder Sent: Friday, January 16, 2009 7:09 PM To: karuna durai Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GDM I know that, but I want only PH 1 can delete the msg from the GDM mailbox , Ph 2 not be allowed to delete the msg From: karuna durai [mailto:karu...@gmail.com] Sent: Friday, 16 January 2009 8:53 PM To: Kumar, Narinder Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GDM To retrieve the GDM mailbox, press the message button after password, please enter 9 and 1 try and let me know... On Thu, Jan 15, 2009 at 5:00 PM, Kumar, Narinder narinder.ku...@uxcg.com.au wrote: All, How can I achieve if a msg is left for GDM. Ph 1 (3001) and ph2 (3002) can listen to the msg but only PH1 (3001) can delete the msg for the mail box. Ph2 is not allowed to delete the msg. Is it possible if yes how, can I achieve this task ? Cheers 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. 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] B-Channel Maintenance not working properly
Agreed! It does not work well at all. I typically will have calls incoming / outgoing the 6608 PRI and make a call 1 hour later to find that the call fails. After a fitting Microsoft salute (reset), the call goies through... Scott Hardesty | Solutions Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vik Malhi Sent: Sunday, February 22, 2009 12:33 PM To: Robert Schuknecht; OSL Group Subject: Re: [OSL | CCIE_Voice] B-Channel Maintenance not working properly I've battled with this service parameter for over 5 years. It doesn't work properly- I would use Top-Down with a fractional PRI. But if you are trying to get it working you must ensure that you Enable Status Poll on the gateway page. Also- you could try including a space after every 4 bits (although I'm told that is not necessary, that is what I have always done since the example show this). So try this : (+ Enable Status Poll). S0/ds...@sda000332333241=0001 -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Robert Schuknecht rschukne...@gmx.de Date: 22 Feb 2009 18:07:17 +0100 To: OSL Group ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] B-Channel Maintenance not working properly Hi List, during my last Remote-Rack Sessions i noticed that the B-Channel Maintenance Status Parameter is not workking properly. Always when i configured it and restartet the CCM Srevice and the Gateway itself, it is working for some calls. And suddenly the Gateway is trying to call out over the not available B-Channels I configured the Maintenance Status the following ways: 1) S0/ds...@sda000332333241=0001 2) S0/ds...@sda000332333241 = 0001 But both of them did not work. What am i doing wrong here? /Robert
[OSL | CCIE_Voice] IPMA Woes.. :(
All, I am having issues with the manager phone. Everything seems to be configured correctly but once the IPMA Standard Manager softkey template get applied to the manager's phone the phone hangs requesting softkey template displayed on the phone. If I remove the IPMA Manager softkey template, the phone will load correctly with the appropriate line assignments. I have tried multiple phones / phone types and same situation so it is obvious that I am missing something. Any help is appreciated. Thx! Scott Hardesty | Solutions Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/
Re: [OSL | CCIE_Voice] MOH Issue
Could you post your sho run on the router that you are sourcing the MOH from? If you are getting dead air, that means your CCM is setup correctly and the issue is pointing to the local MOH configuration. Do you have at least 1 ephone defined? Another note, if you had it working with g711 and just changed the moh / multicast information to reflect g729 you may have to delete the entire call-manager fallback configuration and paste it back in with the g729 information. I have run into this in the past. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jose Gregorio Linero (jlinero) Sent: Wednesday, January 14, 2009 3:07 PM To: Ryan Trauernicht; Antonio McCarver Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Issue Hi Ryan: No it does not, it could be G711. Regards, Jose From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ryan Trauernicht Sent: Miércoles, Enero 14, 2009 1:16 PM To: Antonio McCarver Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Issue If i set my MOH server to G729 for the remote branch and put a G711 file on the flash with the following commands: moh .wav multicast moh 239.1.1.3 port 16384 route X.X.X.X X.X.X.X I get dead air is that b/c the file type loaded on the flash needs to be g729? On Wed, Jan 14, 2009 at 12:11 PM, Antonio McCarver amccar...@cciequest.com wrote: Hello group, I am at the very beginning stages of my lab prep so please forgive me if this is one of those come on newbie, you should've known that questions. I have read and re-read the MOH section in the CallManager Fundamentals book, and in the CUCM 7.x SRND and I don't see where either went into detail about the different mcast addresses 239.1.1.1, .2, or .3. My question is, where can I look to read up on them and this issue? Amp Quoting Vik Malhi vma...@ipexpert.com: The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM ³thinks² that the MOH server back in HQ is active and the stream is g729. But I guess that¹s the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM ³thinks² is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
Re: [OSL | CCIE_Voice] using Change B-Channel Maintenance for IOST1 or not?
You use the B-channel maintenance for all MGCP gateways (IOS and 6608 module). For the lab, outbound calls should work without b-channel maintenance if you are using the top-down since the gateway will select the 1st channel and hunt sequentially down for additional channels. How is the PSTN sending the calls inbound? If the PSTN is sending calls inbound starting at the 23rd channel (bottom up) then the call will likely fail because the 23rd channel is showing available but I is not on the MGCP side. So the always correct engineering answer still applies. IT DEPENDS! Scott. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co Sent: Wednesday, December 24, 2008 1:51 AM To: saralilin2...@yahoo.co.jp Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] using Change B-Channel Maintenance for IOST1 or not? I think we use this option to busy out unused channels since we use fractional T1 not full one. But my doubt is to use it on IOS mgcp GW or not? Jeremy On Wed, Dec 24, 2008 at 5:36 PM, saralilin2...@yahoo.co.jp wrote: we only need to do b-channel maintenance if question ask for bottom up right? if we choose top down this is not needed, am i right? Sara jeremy co jeremy.coo...@gmail.com wrote: Hi, I've seen some workbooks use Change B-Channel Maintenance option to busyout unused channels on T1 of IOS GWs as well as 6500 T1 while some of them only use this option on 6500 T1. In cisco Docs, I can it specified to use thi option for MGCP gateways So which method should be used? btw, I use both and both works for IOS . Jeremy Power up the Internet with Yahoo! Toolbar. http://pr.mail.yahoo.co.jp/toolbar/
Re: [OSL | CCIE_Voice] CAC mechanism to limit the number of calls overthe WAN
While that is true, ccm uses 24k per g729 call. Much like the gatekeeper using 16k per call. This is not supposed to be actual values. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: Robertico Gonzalez robertico.gonzale...@gmail.com Sent: Thursday, December 11, 2008 2:55 PM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CAC mechanism to limit the number of calls overthe WAN Hi, IPexpert's Voice 5-Day Boot Camp, Lab 1, Question 7: 7. Use CallManager's only CAC mechanism to limit the number of calls over the WAN between the HQ and BR1 to four audio calls and one video call using the minimum bandwidth allowable by a VTA camera in H263 mode. Question: The solution sets the BR1 location to 96 Kbps. This assumes that there are 4 calls and each one consumes 24 Kbps. However, I get higher values in Kbps for a call assuming 20 ms packetization time. MLP = 13 bytes IP/UDP/RTP = 40 bytes G729 = 20 bytes 73 bytes * 8 = 584 bits 584 bits * 50 pps = 29.2 kbps, which is higher than 24 Kbps for a single call For FR, I obtain 25.6 Kbps, since the overhead is 4 bytes. For FRF.12, I obtain 27.2 Kbps, since the overhead is 8 bytes. As you can see for all options, the calculated rate is greater than 24 kbps. Regards, -rg
Re: [OSL | CCIE_Voice] CBWFQ without specific PVC bandwidth data
I believe you should assume 1536. When using b8zs / ESF, you loose a few bits for overhead. Showing my age, I recall converting circuits from AMI to b8zs and we needed to adjust the line speed accordingly. As for the other, the question should provide you the port speed of the link. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jose Gregorio Linero (jlinero) Sent: Monday, December 08, 2008 1:47 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CBWFQ without specific PVC bandwidth data Hi all: I was testing this but I am not sure what have I to assume. If some reason we don´t have the PVC bandwidth for a specific connection, for example between HQ and BR1, and we have to configure 33% for voice and 5% for signaling, what sould we have to assume?, that it is a T1?, or left the PVC in the default value 56 kbps?, and if we assume a T1 what would be the bandwidth?, 1536?, 1544? Any ideas? Regards, Jose Gregorio Linero Welcker Systems Engineer - Service Provider - CCIP - CCVP Sales / Channels [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Phone :+(571) 3256052 Mobile :+(57) 310 2634216 Fax :+(571) 3256090 Carrera 7 # 71-21 Torre A Piso 17 Colombia www.cisco.com/global/CO/ http://www.cisco.com/global/CO/ This e-mail may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply e-mail and delete all copies of this message. image004.gifimage005.gifimage006.gif
Re: [OSL | CCIE_Voice] H323 Config ??
You should use voice class to set tcp timeout. By default q931 will timeout before fail over from sub to pub. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: Mike Brooks [EMAIL PROTECTED] Sent: Sunday, December 07, 2008 4:05 PM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] H323 Config ?? Hi everyone, On an H323 gateway with dial-peers pointing back to the SUB and PUB, should a voice class be defined or just hardcode the codec to G711ulaw ? I would think G711 being defined on the dialpeer would work fine. Please let me know if you see any issues with the config below. == voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729br8 ! voice class h323 1 h225 tcp establish timeout 3 ! dial-peer voice 100 voip destination-pattern 2... session target ipv4:10.1.200.20 dtmf h245-alpha codec g711ulaw -- or --- voice class codec 1 voice class h323 1 no vad ! dial-peer voice 100 voip destination-pattern 2... session target ipv4:10.1.200.21 dtmf h245-alpha codec g711ulaw - or --- voice class codec 1 voice class h323 1 preference 1 no vad ! dial-peer voice 1 voip incoming called-number . voice class codec 1 dtmf h245-alpha no vad ! Thx, Mike Brooks CCIE# 16027 (RS)
Re: [OSL | CCIE_Voice] UCE failed to connect to UNITY cluster
Looks like DNS has resolved your address as your default gateway. Should point to .2 correct? Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erwan Erwan Sent: Tuesday, November 25, 2008 10:11 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCE failed to connect to UNITY cluster hi , I try to connect UCE to Cisco Unity cluster , but when I verify server , it failed here is the log it is ping able from Unity and UCE, any idea ? - interface Service-Engine0/0 ip unnumbered FastEthernet0/0.51 service-module ip address 177.3.51.2 255.255.255.0 service-module ip default-gateway 177.3.51.1 !--- Testing Device 1 - CiscoUM2-VI1 Device SecurityMode is Non-Secure. *** Device Test Failed. *** CCM Server 1 [177.3.51.1, TCP Port 2000] - Register Test Failed. Resolved IP Address as 177.3.51.1. Failure Reason: ErrorCannotEstablishWinsockConnection Troubleshooting Tip: Make sure that the CCM Server is accessible via the network and is otherwise operating correctly.
Re: [OSL | CCIE_Voice] VM with SRST and AAR (lab 3)
Use should use the full voicemail pilot number under call-manager fallback without wildcards. Voicemail 912122251600. No translation is needed with the voicemail button as it is a direct call to voicemail and you should be prompted for password. Hth. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of marwa Sent: Sunday, November 23, 2008 9:35 AM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] VM with SRST and AAR (lab 3) hello, i am working in lab 3 , i just needed to test the vm with SRST and also in case of AAR in case of SRST i was able to forward the call to vm but if i press the voice mail buttom it fails Also if i went to AAR mode when the hq dials the any ip phone br1 and after the cfw no anwer timesout, it says enter your password , but if i made callforward all to vm in br1 i reaches the vm interface Serial0/3/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable call-manager-fallback max-conferences 8 gain -6 ip source-address 172.25.101.1 port 2000 max-ephones 4 max-dn 4 voicemail 9121222512.. call-forward pattern .T call-forward busy 9121222512.. call-forward noan 9121222512.. timeout 9 cor incoming all 1 2002 any advise plz Marwa
Re: [OSL | CCIE_Voice] Announciator messages to PSTN
You can not use annunciator for pstn. You need to route the call to unity and use call handler... Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: Michael Shavrov [EMAIL PROTECTED] Sent: Tuesday, November 18, 2008 11:56 AM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Announciator messages to PSTN Hi, How to play messages with announciator to PSTN? For example, if PSTN phone calls number, which belongs to location but has no configured DN, user should hear message Number is not in service. I tried to configure both, Route Pattern and Translation Pattern with Block pattern - it works internally, but does not work from PSTN. Also, there is no configurable option for Number not in service - call manager just rejects the call. Mike
Re: [OSL | CCIE_Voice] CUE Wired problem: call to voice pilot numberworks, but message button not working.
Define voicemail pilot number under telephony service? Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: jeremy co [EMAIL PROTECTED] Sent: Friday, November 14, 2008 11:12 PM To: CCIE Voice Maillist ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUE Wired problem: call to voice pilot numberworks, but message button not working. Hi, When I call to voice pilot number, it works, but when hit message button it wouldn't work. Can someone shed light on what may cause this issue? I restarted cue and phones but it's not working. Jeremy
Re: [OSL | CCIE_Voice] CUE Wired problem: call to voice pilotnumberworks, but message button not working.
I do not see your XCODER defined in CME. Try adding sdspfarm tag 1 XCODER under Telephony-service. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jeremy co Sent: Saturday, November 15, 2008 12:00 AM To: Greg Miglucci (gmiglucc) Cc: CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] CUE Wired problem: call to voice pilotnumberworks, but message button not working. Hi, I forgot to mention that when I press message button , call connects to voice pilot number but I cannot hear anything Here is the config: ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 ! ! ! ! ! mgcp ! sccp local FastEthernet0/0.200 sccp ccm 200.0.0.254 identifier 1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register XCODER ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 4 associate application SCCP ! ! dial-peer voice 5000 pots destination-pattern 098215... direct-inward-dial forward-digits 8 ! dial-peer voice 4000 pots destination-pattern 06134... direct-inward-dial forward-digits 4 ! dial-peer voice 30 pots destination-pattern 01124344... port 1/2:0 forward-digits 8 ! dial-peer voice 200 pots incoming called-number . direct-inward-dial port 1/2:0 ! dial-peer voice 600 voip destination-pattern session target ras dtmf-relay h245-alphanumeric ! dial-peer voice 999 voip destination-pattern session protocol sipv2 session target ipv4:200.0.0.100 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 1001 voip destination-pattern 7000 session protocol sipv2 session target ipv4:200.0.0.100 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 7878 voip destination-pattern 800[01] session protocol sipv2 session target ipv4:114.0.0.0.254 dtmf-relay sip-notify codec g711ulaw no vad ! gateway timer receive-rtp 1200 ! sip-ua ! ! ! ! gatekeeper zone local FXSZONE abc.com 7.7.7.7 invia IPIPGWZONE outvia IPIPGWZONE enable-intrazone zone local CCMZONE abc.com zone local HQZONE abc.com zone local IPIPGWCCMZONE abc.com zone local IPIPGWZONE abc.com zone prefix HQZONE 011* zone prefix IPIPGWCCMZONE 3... gw-type-prefix 1#* default-technology no use-proxy FXSZONE default inbound-to terminal no use-proxy FXSZONE default outbound-from terminal bandwidth interzone zone CCMZONE 144 no shutdown ! ! telephony-service max-ephones 10 max-dn 200 ip source-address 200.0.0.254 port 2000 sdspfarm units 1 sdspfarm transcode sessions 2 create cnf-files version-stamp 7960 Mar 02 2002 04:09:38 voicemail mwi relay max-conferences 4 gain -6 web admin system name admin secret 5 $1$fS6t$7Pfc17AMenIo5ISKL3w2N0 ! ! ephone-dn 38 number 1000 ! ! ephone-dn 70 number 9911 call-forward noan timeout 10 ! ! ephone-dn 80 number 8000 mwi on ! ! ephone-dn 81 number 8001 mwi off ! ! ephone 1 username pstn password cisco mac-address 0030.94C2.BFB1 button 1:70 ! Jeremy On Sat, Nov 15, 2008 at 3:40 PM, Greg Miglucci (gmiglucc) [EMAIL PROTECTED] wrote: Post your telephony-service config and ephone-dn and dial-peer config. Also are you using dial-plan pattern under teleph-service? If so you need to make sure you have appropriate translation-rules. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jeremy co Sent: Friday, November 14, 2008 11:28 PM To: CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] CUE Wired problem: call to voice pilotnumber works, but message button not working. I forgot to mention that when I press message button , call even connected to voice pilot number but I cannot hear anything Jeremy On Sat, Nov 15, 2008 at 3:12 PM, jeremy co [EMAIL PROTECTED] wrote: Hi, When I call to voice pilot number, it works, but when hit message button it wouldn't work. Can someone shed light on what may cause this issue? I restarted cue and phones but it's not working. Jeremy
Re: [OSL | CCIE_Voice] Invoking a transcoder for WAN calls intoB-ACD
If you are able to get calls into CUE across the WAN it sounds like the xcoder is working. I would suggest looking at your dial-peer pointing to BACD. PSTN calls and CME phones will match on your incoming called-number definition in your dial-peer. Depending on how you are handling calls from the WAN, you may need destination-pattern 3500 defined in your dial-peer. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Brooks Sent: Saturday, November 08, 2008 10:35 AM To: rob Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Invoking a transcoder for WAN calls intoB-ACD Hi Rob, What incoming dial-peer are you hitting when you call from BR1 or HQ into the BACD on CME ? If the codec is not hardcoded to G729 then the transcoder will not be invoked. This occurs if the dial-peer you are hitting has flexibility within the codec such dial-peer 0 or a dial-peer with a voice-class codec assigned. Regards, Mike Brooks CCIE# 16027 (RS) On Fri, Nov 7, 2008 at 4:53 AM, rob [EMAIL PROTECTED] wrote: Hi, I'm testing Workbook 3, lab 2 task 42 and although I have completed the requirements for this task and can call into the B-ACD via the PSTN and from CME phones. Although it is not a requirement I am unable to establish a call across the WAN from the HQ or BR1 sites. My CME transcoder is working for 729 to 711 calls into CUE but doesnt invoke for WAN calls into B-ACD. Is this even possible? Thanks, Rob
Re: [OSL | CCIE_Voice] 4d from CME to CCM
Are you using Regions and locations based CAC between hq and br1? Since the call is hitting HQ as g711 you need to make sure that HQ can talk to BR1 using g711 or ensure that a xcoder is available to transcode to g729. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olson, Pete Sent: Thursday, November 06, 2008 4:17 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] 4d from CME to CCM When making 4 digit calls from CME to CCM. Calls to HQ (1003, 1001) work, but calls to BR1 (2001, 2003) fail. What extra is needed for calls to BR1? Since they are registered with CCM, I wouldn't think anything else would be needed. The flow is CME (G729, SIP, RTP-NSE) to HQ-RTR to CCM (h323, G711, h245-a) Debug voip dialpeer on HQ-RTR shows both calls using the same dialpeers. Pete Olson [EMAIL PROTECTED] 425-965-2577
Re: [OSL | CCIE_Voice] VPN connection to vrack
I had a problem with po18 on friday night. I could not get my vpn router to connect. I used pod19 last night without an issue.' Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [EMAIL PROTECTED] Sent: Sunday, November 02, 2008 9:47 AM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] VPN connection to vrack hi all, I had a problem with the vpn connection to the pod18. initially i was able to be authenticated and the tunnel is up. but once the ip phone register to the callmanager, the ezvpn connection got disconnected. i have tried to bring up the tunnel via the web link. however every time the tunnel is up, my router gets reboot at the same time. anyone has similiar experience? i am using 870 router. thanks in advance Sara Power up the Internet with Yahoo! Toolbar. http://pr.mail.yahoo.co.jp/toolbar/
Re: [OSL | CCIE_Voice] Pod 26 - BR1 NM-16ESW port config
I just logged into pod 26 for my afternoon session. I will let you know if i have problems. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: Trevor Peddle [EMAIL PROTECTED] Sent: Friday, October 24, 2008 4:16 PM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Pod 26 - BR1 NM-16ESW port config I have had many replies and verification of my configuration, thank you all for that. Yes the VLANs were ceated and verified I could also ping the VLAN interfaces. So the scenario is this the interfaces and vlans are configured correctly as is the DHCP scope. The phones pick up an address from DHCP and then drop it, if I shut/no shut the interfaces they will pick up the DHCP ip again and then drop it. I cleared CDP else the IP would stay there. This has happened to me 2 days running I wonder if the module could be checked ? I definatley went over the 10 minute troubleshooting rule on this one From: Trevor Peddle [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Sent: Friday, 24 October, 2008 18:55:24 Subject: [OSL | CCIE_Voice] Pod 26 - BR1 NM-16ESW port config Hi all, I have configured the phone ports on BR1 as follows, they get an IP from the DHCP scope on CUCM server which I can see also with CDP on the router. However they are not contactable via IP, if I clear the CDP table the addresses disapear. I can ping the default gateway, it is correct in DHCP as is the net mask. I sometimes get a response via ping but probably a couple out of the blue now and again. The VLAN interfaces's are set correctly because I can see them ok. interface FastEthernet1/0 switchport trunk native vlan 360 switchport mode trunk switchport voice vlan 460 interface FastEthernet1/8 switchport trunk native vlan 360 switchport mode trunk switchport voice vlan 460 I configured switchport trunk encapsulation dot1q but it does not show in the config I also tried just as an access port with access vlan and voice vlan, also without success. This seems such an easy issue as I have never had such issues before when configuring similar ? I know my brain is swimming at the moment, first lab attempt on 6th Nov, but I do not think I have missed anything? I had the same problem on the same pod yesterday.
[OSL | CCIE_Voice] POD26 Subscriber not available
Proctor lab folks. POD26 subscriber is not accessible. I tried to open a ticket for after hours support but both links sends you to the support forum. I have posted the issue there as well. Scott. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/
Re: [OSL | CCIE_Voice] Problem with PVDM-2 Conference registration
I have experienced the same situation as Jacob. In fact, when the conference bridge registers, you can see an ephone type registration. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Owen Sent: Wednesday, October 22, 2008 2:39 PM To: Michael Shavrov Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with PVDM-2 Conference registration Michael, It may seem weird but i have noticed until I set the Max-ephones X command my MTP/CONF resources won't register to CCME. It's worth a shot, i know it usually is only an issue when setting a transcoder on HQ since it inherintly doesn't have ephones registered to it and therefore that command is usually missing. Try it and see what happens On Wed, Oct 22, 2008 at 12:57 PM, Michael Shavrov [EMAIL PROTECTED] wrote: Hi All, I have an ongoing problem with creating a media resources on CME router. I create 2 dspfarm profiles, configure SCCP, register with CME... Transcoding registeres fine, but Conferencing does not. Here is part of the config, and the SCCP status: voice-card 0 dspfarm dsp services dspfarm ! sccp local GigabitEthernet0/0.123 sccp ccm 10.10.66.254 identifier 1 sccp ip precedence 3 sccp ! sccp ccm group 1 bind interface GigabitEthernet0/0.123 associate ccm 1 priority 1 associate profile 1 register cfb001da10466d8 associate profile 2 register mtp001da10466d8 ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP ! telephony-service sdspfarm units 2 sdspfarm tag 1 cfb001da10466d8 sdspfarm tag 2 mtp001da10466d8 ip source-address 10.10.66.254 port 2000 max-conferences 2 gain -6 ! - R7-BR2#sh sccp SCCP Admin State: UP Gateway IP Address: 142.103.66.254, Port Number: 2000 IP Precedence: 3 User Masked Codec list: None Call Manager: 142.103.66.254, Port Number: 2000 Priority: N/A, Version: 3.1, Identifier: 1 Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 142.103.66.254, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 Conferencing Oper State: ACTIVE_IN_PROGRESS - Cause Code: CCM_REGISTER_FAILED Active Call Manager: 142.103.66.254, Port Number: 2000 TCP Link Status: NOT_CONNECTED, Profile Identifier: 1 Reported Max Streams: 16, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 -- Jacob Owen CCIE #14063 (RS, Service Provider), CCDP, CCVP
Re: [OSL | CCIE_Voice] Problem with PVDM-2 Conferenceregistration
Michael, I am running CME version 3.3 and I can register a hardware conference Transcoder. I would make sure you have enough ephones. SEE BELOW, EPHONE-3 has registed using the gi0/0 interface BR2#sho ephone registered ephone-1 Mac:0011.BB53.7636 TCP socket:[6] activeLine:0 REGISTERED in SCCP ver 6 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.107.66.11 50031 Telecaster 7940 keepalive 54 max_line 2 button 1: dn 1 number 4001 CH1 IDLE CH2 IDLE Username: phnOne Password: null ephone-2 Mac:0014.A963.9EDF TCP socket:[5] activeLine:0 REGISTERED in SCCP ver 6 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.107.66.12 50035 Telecaster 7940 keepalive 54 max_line 2 button 1: dn 2 number 4002 CH1 IDLE CH2 IDLE Username: phoneTwo Password: null ephone-3 Mac:0011.9348.0940 TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 0 + Authentication mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.107.66.254 39825 Unknown 51 keepalive 6 max_line 0 SEE that the SCCP Conference bridge is ACTIVE BR2#sho Oct 22 20:23:10.488: %SYS-5-CONFIG_I: Configured from console by consolesccp SCCP Admin State: UP Gateway IP Address: 142.107.66.254, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 142.107.71.254, Port Number: 2000 Priority: N/A, Version: 3.1, Identifier: 1 Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 142.107.71.254, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 16, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 BR2#sho ver Cisco IOS Software, 3800 Software (C3845-ADVIPSERVICESK9-M), Version 12.4(3j), R ELEASE SOFTWARE (fc1) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2007 by Cisco Systems, Inc. Compiled Fri 14-Dec-07 03:18 by stshen ROM: System Bootstrap, Version 12.3(11r)T1, RELEASE SOFTWARE (fc1) BR2 uptime is 1 day, 16 hours, 39 minutes System returned to ROM by reload at 15:23:51 CEST Mon Oct 20 2008 System restarted at 04:44:21 CEST Tue Oct 21 2008 System image file is flash:c3845-advipservicesk9-mz.124-3j.bin Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Shavrov Sent: Wednesday, October 22, 2008 3:01 PM To: Robert Schuknecht; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with PVDM-2 Conferenceregistration Looks like you right - I've tried it with both, PVDM and PVDM2. In both cases transcoder registers instantly, but conferencing does not. I will try to upgrade IOS on one of the router to CME 4.0 and see if it will solve the problem with the same config. - Original Message - From: Robert Schuknecht [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com; [EMAIL PROTECTED] Sent: Wednesday, October 22, 2008 2:26 PM Subject: Antw: [OSL | CCIE_Voice] Problem with PVDM-2 Conferenceregistration Hi Michael, Hardware Conferencing is not supported with CCME 3.3. When you want to use a Hardware Conferencebridge with CCME you have to use CCME Version 4.0 or maybe higher, i don´t remember it at the moment. HTH /Robert Michael Shavrov[EMAIL PROTECTED] schrieb am Mittwoch, 22. Oktober 2008 um 18:57 in Nachricht 93f2ba42c46071d4f82376100596cbc2: Hi All, I have an ongoing problem with creating a media resources on CME router. I create 2 dspfarm profiles, configure SCCP, register with CME... Transcoding registeres fine, but Conferencing does not. Here is part of the config, and the SCCP status: voice-card 0 dspfarm dsp services dspfarm ! sccp local GigabitEthernet0/0.123 sccp ccm 10.10.66.254 identifier 1 sccp ip precedence 3 sccp ! sccp ccm group 1 bind interface GigabitEthernet0/0.123 associate ccm 1 priority 1 associate profile 1 register cfb001da10466d8 associate profile 2 register mtp001da10466d8 ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP ! telephony-service sdspfarm units 2 sdspfarm tag 1
Re: [OSL | CCIE_Voice] IP connecttivity to CUE module
Make sure you have a static route to the CUE service module. IP route 142.107.66.2 255.255.255.255 service-module 0/1 Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Suresh Solomon Sent: Monday, October 20, 2008 6:10 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] IP connecttivity to CUE module Hi All, 1. I have CUE installed on Branch 2. From the BR2 I am able to ping the service engine ip address. From the HQ I am unable to ping the IP of the service engine module. Is this the way it works or have I made a mistake? 2. CATOS. Presently the CAt6503 has ios installed on this. Can I boot to a flash disk to boot CATOS. And then remove the disk and it boots as it currently does? e.g. Like on a pc when you boot to a USB with linux and without the USB it boots to Windows . Thank you Suresh __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [OSL | CCIE_Voice] IP connecttivity to CUE module
You are correct Jacob. IT should be Service-Engine 0/1 not Service-Module. Sorry for the typo! Thanks Jacob.. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: Jacob Owen [mailto:[EMAIL PROTECTED] Sent: Monday, October 20, 2008 7:47 AM To: Hardesty, Scott Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IP connecttivity to CUE module Scott, Are you sure the service-module is the destination interface for that route? For some reason I thought had used Service-engine 0/0 in all of my routes. Maybe they are interchangeable. On Mon, Oct 20, 2008 at 7:43 AM, Hardesty, Scott [EMAIL PROTECTED] wrote: Make sure you have a static route to the CUE service module. IP route 142.107.66.2 255.255.255.255 service-module 0/1 Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | [EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | www.presidio.com http://www.presidio.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Suresh Solomon Sent: Monday, October 20, 2008 6:10 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] IP connecttivity to CUE module Hi All, 1. I have CUE installed on Branch 2. From the BR2 I am able to ping the service engine ip address. From the HQ I am unable to ping the IP of the service engine module. Is this the way it works or have I made a mistake? 2. CATOS. Presently the CAt6503 has ios installed on this. Can I boot to a flash disk to boot CATOS. And then remove the disk and it boots as it currently does? e.g. Like on a pc when you boot to a USB with linux and without the USB it boots to Windows . Thank you Suresh __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -- Jacob Owen CCIE #14063 (RS, Service Provider), CCDP, CCVP
Re: [OSL | CCIE_Voice] call disconnected after couple of seconds, cannot hear any thing when picked up.
Make sure that you do NOT have wait for h245 capabilities exchange enabled on your CCM trunk. Please send full configuration on your Gatekeeper / IPIPGW. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jeremy co Sent: Friday, October 17, 2008 11:26 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] call disconnected after couple of seconds,cannot hear any thing when picked up. I forgot to say I got bearer capability not implemented (65) for disconnected cause code. On Sat, Oct 18, 2008 at 2:21 PM, jeremy co [EMAIL PROTECTED] wrote: Hi, I add some commands as u guys suggested: Interesting things happened. from CCM to MC3810 when I make a call ,it would ring ( not a full ring ,just hear ringing for 0.5 sec) and then disconnected. from MC3810 FXS port to CCM ,I can call and ip phone ringing then when I pick up the ipphone ,nothing happens and from analogue phone prespective it seems no body picks up the ip phone on other side and keep hearing dial tone wait for other side to pick up , after couple of seconds I hear fast busy signal and both ends disconnected. So wired! ANy idea what's going on!!!? By the way I use 711 on ccm and 711 ulaw hardcoded to dial peer on IPIPGW/GK as u can see. Here are some changes I made AS U GUYS SUGGESTED: --- unchecked wait for media capabilities on MC3810: dial-peer voice 300 voip destination-pattern 3... session target ras dtmf-relay h245-alphanumeric ///this command added on IPIPGW/GK: telephony-service max-ephones 10 max-dn 10 sdspfarm units 1 sdspfarm transcode sessions 2 Cheers, On Sat, Oct 18, 2008 at 6:3 AM, James Key [EMAIL PROTECTED] wrote: Jeremy, On your Gatekeeper controlled trunk, make sure that Wait for Far End H.245 Terminal Capabilities Set is NOT checked. When an IPIPGW is involved, everything has to be hardcoded and H.245 negotiation doesn't place nice with IPIPGW in 12.4 mainline IOS. James From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jeremy co Sent: Friday, October 17, 2008 11:16 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] call disconnected after couple of seconds,cannot hear any thing when picked up. Hi, (3001) ipphone--CCM--HQ---IPIPGW/GK-MC3810--FXS () CCM in IPIPGWCCMZONE zone HQ in HQZONE IPIPGW/GK in IPIPGWZONE MC3810 in FXSZONE I get no resource (47) disconnected cause code . When I remove zone local FXSZONE abc.com invia IPIPGWZONE outvia IPIPGWZONE enable-intrazone from GK and replace it with zone local FXSZONE abc.com , basically removing ipipgw , every thing is fine and I can call from both sides ,ccm and FXS port, but as soon as I add invia IPIPGWZONE outvia IPIPGWZONE enable-intrazone, I can still call fraom both sides but when i pick up the phone I cannot hear any thing and call will disconnected in couple of seconds. even if I let phones ringing after couple of seconds it will disconnect. It sunds like codec problem but I believe some thing is wrong with ipipgw config. Any idea? this is my scenario: PIPGW/GK config : hostname C2600 ! frame-relay switching isdn switch-type primary-ni voice-card 1 dsp services dspfarm ! ! ! ! voice service voip allow-connections h323 to h323 ! ! ! ! ! controller T1 1/0 framing esf clock source internal linecode b8zs pri-group timeslots 1-10,24 ! controller T1 1/1 framing esf clock source internal linecode b8zs pri-group timeslots 1-10,24 ! controller E1 1/2 ds0-group 0 timeslots 1-2 type r2-digital r2-semi-compelled ani ! ! interface Loopback0 ip address 7.7.7.7 255.255.255.255 ! interface Loopback2 ip address 2.2.2.2 255.255.255.0 ip route-cache same-interface h323-gateway voip interface h323-gateway voip id IPIPGWZONE ipaddr 7.7.7.7 1719 h323-gateway voip h323-id IPIPGW ! interface FastEthernet0/0.200 encapsulation dot1Q 200 ip address 200.0.0.254 255.255.255.0 ! ! mgcp ! sccp local FastEthernet0/0.200 sccp ccm 142.4.64.11 identifier 1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register XCODER ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 4 associate application SCCP ! ! dial-peer voice 30 pots destination-pattern 01122443... port 1/2:0 forward-digits 8 ! dial-peer voice 200 pots incoming called-number . direct-inward-dial port 1/2:0 ! dial-peer voice 600 voip destination-pattern session target ras dtmf-relay h245-alphanumeric codec g711ulaw ! dial-peer voice 601 voip destination-pattern 3... session target ras dtmf-relay h245-alphanumeric codec g711ulaw ! gateway timer receive-rtp 1200 ! ! ! !
Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt
Can you send your dial-peer for the BACD application? Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kapil Atrish Sent: Saturday, October 04, 2008 7:58 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD issue - No welcome prompt HI, Attached is my config. I get fast busy tone and Unknown number on display when I dial the pilot number from any CME phone. I can dial hunt-pilot directly and call get routed correctly or give the aa-pilot to hunt-pilot and ring the phones fine. Call in between phones are setup using G711ulaw. I've tried single voip dial-peer with incoming called-address and destination-pattern, reload of router, re-configure script. Below is the snapshot of bacd config and debug voice application seesion.. application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 15 param aa-hunt5 3701 param queue-manager-debugs 1 param number-of-hunt-grps 2 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 2 param menu-timeout 6 param handoff-string aa param dial-by-extension-option 4 paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3700 paramspace english location flash: param second-greeting-time 30 param welcome-prompt _bacd_welcome.au param queue-manager-debugs 1 param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 3005 paramspace english prefix en param service-name queue ! ! ! BR2#dir flash: Directory of flash:/ 1 -rw- 24679no date app-b-acd-2.1.0.0.tcl 2 -rw- 33870no date app-b-acd-aa-2.1.0.0.tcl 3 -rw- 75650no date en_bacd_allagentsbusy.au 4 -rw- 83291no date en_bacd_disconnect.au 5 -rw- 63055no date en_bacd_enter_dest.au 6 -rw- 37952no date en_bacd_invalidoption.au 7 -rw- 496521no date en_bacd_music_on_hold.au 8 -rw- 123446no date en_bacd_options_menu.au 9 -rw- 42978no date en_bacd_welcome.au 10 -rw- 496521 Mar 01 2002 01:13:09 +00:01 music-on-hold_3db.au 11 -rw- 496521 Mar 01 2002 02:47:07 +00:01 music-on-hold.au 536870908 bytes total (534895700 bytes free) BR2# BR2# There is no output when I do debug voice application script OUTPUT OF debug voice application session is as below. Calling no: 3002, called no: 3700 BR2#debug voice app BR2#debug voice application sess voip application session debugging is on BR2# Mar 1 03:17:40: //37//AFW_:/Closing_AnyEvent: Mar 1 03:17:40: //37//AFW_:/Session_Cleaner: Mar 1 03:17:40: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:40: //37/8A066DCF802F/AFW_:/C_ServiceSession_Event_Handler: Received event CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop Mar 1 03:17:40: //37//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:40: //-1//AFW_:HN000FF5F4:/AFW_M_Session_Free: MOD[Session_65BFE164_0_1046004]( ) Mar 1 03:17:42: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:42: //-1//AFW_:/AFW_Session_New: Mar 1 03:17:42: //40//AFW_:/C_PackageSession_NewCall: Session module listened by TclModule_65BE268C_0_1066356 Mar 1 03:17:42: //40//AFW_:/Open_SetupIndication: Calling #(3002), Called #(), peer_tag(20002) Mar 1 03:17:44: //40//AFW_:/GettingDest_DigitCollectDone: status(4) discCause(0) ovrlp(TRUE) Mar 1 03:17:44: //-1//AFW_:/C_PackageSession_GetSigPeer: Mar 1 03:17:44: //40//AFW_:/ContactingDest_SetupDone: Mar 1 03:17:44: //40//AFW_:/Session_Close: lastFailureCause 34 Mar 1 03:17:44: //40//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:44: //40//AFW_:/AFW_M_Session_Terminate: lastFailureCause 34 Mar 1 03:17:44: //40//AFW_:/Session_Cleaner: Mar 1 03:17:47: //40//AFW_:/Closing_AnyEvent: Mar 1 03:17:47: //40//AFW_:/Session_Cleaner: Mar 1 03:17:47: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:47: //40/96270EF58032/AFW_:/C_ServiceSession_Event_Handler: Received event CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop Mar 1 03:17:47: //40//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:47: //-1//AFW_:HN00104580:/AFW_M_Session_Free: MOD[Session_65BFE214_0_1066368]( ) BR2# BR2# BR2# BR2# Any inputs are very welcome... MSN Technology brings you the latest on gadgets, gizmos and the new hits in the gaming market. Try it now! http://computing.in.msn.com/
[OSL | CCIE_Voice] TEST EMAIL
Have not seen a post in a while just checking my account. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/
[OSL | CCIE_Voice] Unity Question
All, I am testing to voicemail looping scenario and having troubles finding the advanced setting for unity to play a tone on answer. This is supposed to prevent looping when ANI is lost. Unity plays the tone and that is the signal that the call was looped and it drops. I think I have the logic right but I cant find the parameter to change.. :) Any help or direction would be appreciated! Thanks. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, September 13, 2008 12:00 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 31, Issue 81 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. NTP Question (Kumar, Narinder) -- Message: 1 Date: Sun, 14 Sep 2008 01:03:19 +1000 From: Kumar, Narinder [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] NTP Question To: ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII In the real lab , if NTP is required to configure on the CCM , do we configure the NTP ( auto or manual method whichever is asked) on both pub and sub or pub only... Cheers. -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 31, Issue 81 **
[OSL | CCIE_Voice] WAN QOS Question
I am trying to mark all of my rtp and control traffic at the WAN edge router and not trust the LAN. The issue I am having is that when I try to apply the service-policy to the ingress Ethernet Interface I get the following error. I have tried applying this to the sub-interface as well as the physical interface with the same result.I am assuming I am using the wrong technique to make this happen. Pertinent parts of the configuration are listed below. Any help would be GREATLY appreciated. thx. CBWFQ : Can be enabled as an output feature only class-map match-all CONTROL match access-group name CONTROL class-map match-all RTP match access-group name RTP ! ! policy-map VOICE class RTP set dscp ef class CONTROL set dscp cs3 class class-default fair-queue = ACTUAL ERROR FROM COMMAND= BR2(config-if)#service-policy input VOICE CBWFQ : Can be enabled as an output feature only Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Thu 9/11/2008 7:40 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 31, Issue 72 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Unity error question (Jeff BCI) -- Message: 1 Date: Thu, 11 Sep 2008 19:47:35 -0500 From: Jeff BCI [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Unity error question To: 'Kumar, Narinder' [EMAIL PROTECTED], OSL Group ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain Just noticed something else interesting. When Unity doesn't answer the call after 4 rings, then that Unity port unregisters in CUCM and the call continues to ring over to the next Unity port. For instance, Unity port 1 is being hit first. After 4 rings, the call goes to Unity port 2 and Unity port 1 unregisters. Now it rings off of port 2 and goes to port 3, now Unity port 2 unregisters in CUCM. The only way to recover this is to reboot Unity to get all Unity voice ports to re-register, but still Unity refuses to answer any call. -Jeff Lost in Unity-Land -Original Message- From: Jeff BCI [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2008 8:28 PM To: 'Kumar, Narinder'; OSL Group Subject: Re: [OSL | CCIE_Voice] Unity error question Well, after completely rebuilding Unity from scratch, I have the exact same problem. The rebuild was necessary to move it to a new domain, changing partner server alone didn't work. Anyhow, the event log says Cisco Unity-TSP; TSP device 5 (Cisco Unity port 2) disconnected from Call Manager x.x.x.x. If there are many of these in sequence from the same device 5, this port may not be functioning anymore. Check to see that it is answering calls, and the server may need to be restarted to activate the port again. UTIM shows proper integration, all unity ports in CUCM are registered, licensing is correct and validated, hunt list is good, everything seems to be proper except that the VM ports just simply won't answer the call. Call Viewer doesn't even show the call hitting Unity. After 4 rings from the phone after the call goes to unity, I get fast busy. Never seen this before at all, and I have to get this fixed quickly if I am to be ready for my Oct 8th lab date. Any ideas? All Unity services are running, no errors in event log indicating that a service croaked. Thanks, Jeff -Original Message- From: Kumar, Narinder [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 10, 2008 11:52 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unity error question As Chris suggested check the event log see if any clue or service is failing. Also go through the service may be some critical services are not running... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, 11 September 2008 1:28 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 31, Issue 66 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 [EMAIL
Re: [OSL | CCIE_Voice] WAN QOS Question
Thanks Everyone! That was it. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: Devildoc [mailto:[EMAIL PROTECTED] Sent: Fri 9/12/2008 11:45 AM To: Hardesty, Scott; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] WAN QOS Question Scott, CBWFQ only works for the output queue on the router. It does not work for the input queue. Your statements class class-default and fair-queue in the policy-map voice configured the CBWFQ. For marking purposes only, you mustn't put those 2 statements in your policy-map. You must remove them for the marking policy to work properly. You use those 2 statements when you try to configure LLQ. JD Date: Fri, 12 Sep 2008 10:39:59 -0400 From: [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com; ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] WAN QOS Question I am trying to mark all of my rtp and control traffic at the WAN edge router and not trust the LAN. The issue I am having is that when I try to apply the service-policy to the ingress Ethernet Interface I get the following error. I have tried applying this to the sub-interface as well as the physical interface with the same result. I am assuming I am using the wrong technique to make this happen. Pertinent parts of the configuration are listed below. Any help would be GREATLY appreciated. thx. CBWFQ : Can be enabled as an output feature only class-map match-all CONTROL match access-group name CONTROL class-map match-all RTP match access-group name RTP ! ! policy-map VOICE class RTP set dscp ef class CONTROL set dscp cs3 class class-default fair-queue = ACTUAL ERROR FROM COMMAND= BR2(config-if)#service-policy input VOICE CBWFQ : Can be enabled as an output feature only Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Thu 9/11/2008 7:40 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 31, Issue 72 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Unity error question (Jeff BCI) -- Message: 1 Date: Thu, 11 Sep 2008 19:47:35 -0500 From: Jeff BCI [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Unity error question To: 'Kumar, Narinder' [EMAIL PROTECTED], OSL Group ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain Just noticed something else interesting. When Unity doesn't answer the call after 4 rings, then that Unity port unregisters in CUCM and the call continues to ring over to the next Unity port. For instance, Unity port 1 is being hit first. After 4 rings, the call goes to Unity port 2 and Unity port 1 unregisters. Now it rings off of port 2 and goes to port 3, now Unity port 2 unregisters in CUCM. The only way to recover this is to reboot Unity to get all Unity voice ports to re-register, but still Unity refuses to answer any call. -Jeff Lost in Unity-Land -Original Message- From: Jeff BCI [mailto:[EMAIL PROTECTED] Sent: Thursday, September 11, 2008 8:28 PM To: 'Kumar, Narinder'; OSL Group Subject: Re: [OSL | CCIE_Voice] Unity error question Well, after completely rebuilding Unity from scratch, I have the exact same problem. The rebuild was necessary to move it to a new domain, changing partner server alone didn't work. Anyhow, the event log says Cisco Unity-TSP; TSP device 5 (Cisco Unity port 2) disconnected from Call Manager x.x.x.x. If there are many of these in sequence from the same device 5, this port may not be functioning anymore. Check to see that it is answering calls, and the server may need to be restarted to activate the port again. UTIM shows proper integration, all unity ports in CUCM are registered, licensing is correct and validated, hunt list is good, everything seems to be proper except that the VM ports just simply won't answer the call. Call Viewer doesn't even show the call hitting Unity. After 4 rings from the phone after the call goes to unity, I get
[OSL | CCIE_Voice] IPCC - Agent in Reserved State
Silvia, the times that I have seen this in the past it was a network issue. IPCC places the agent into reserved when it attempts to transfer the call to the agent. The agent will become active / talking once the call is connected to the agent. The agent state resevered means that that the call signaling from IPCC to the phone is working correctly but the RTP stream is not getting to the phone. Identify the types of calls that are not getting to the agent. For instance, call from a phone on the same IP subnet and see if that works and work you way backwards to the gateway. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, September 09, 2008 2:27 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 31, Issue 53 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Having issues with mailing list - test message (Jacob Owen) 2. MGCP Gateway: ccm-manager config (Kumar, Narinder) 3. Block incoming International call in h.323gateway SRST mode? (Balamurugan Singaram) 4. Block incoming International call in h.323gateway SRST mode? (Balamurugan Singaram) 5. Re: Easy way to find module numbers (Jonathan Charles) 6. block Incoming international call h.323 (Balamurugan Singaram) -- Message: 1 Date: Mon, 8 Sep 2008 19:13:54 -0700 (PDT) From: Jacob Owen [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] Having issues with mailing list - test message To: CCIE Voice ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Please disregard, testing email to mailing list Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP -- Message: 2 Date: Tue, 9 Sep 2008 12:13:48 +1000 From: Kumar, Narinder [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] MGCP Gateway: ccm-manager config To: ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII It create issues with partial PRI's... Use ccm-manager config on BR1 , put BR1 under SRST, keep in SRST and reload ur router, you will see what happens.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 9 September 2008 11:43 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 31, Issue 52 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: IPCC - Agent in Reserved State (Ricardo Arevalo) 2. MGCP Gateway: ccm-manager config (Robertico Gonzalez) 3. Re: MGCP Gateway: ccm-manager config (Jonathan Charles) 4. Re: IPCC - Agent in Reserved State (Jonathan Charles) 5. Test / Verification plan (Michael Shavrov) 6. Easy way to find module numbers (Paul and Bobs) 7. Re: Easy way to find module numbers (Michael Shavrov) -- Message: 1 Date: Mon, 8 Sep 2008 12:38:57 -0400 From: Ricardo Arevalo [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] IPCC - Agent in Reserved State To: o Ninja [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Did you check the CSS applied to CTI ports? Sometimes, when the agent goes to reserved state and the goes back to queue, its a CSS problem. //r.a. On Mon, Sep 8, 2008 at 12:33 PM, o Ninja [EMAIL PROTECTED] wrote: The timers are the same, I did not change any value. The phone rings showing that the agent is reserved but for some reason it does not connect after that the call goes to the queue. When the phone is ringing I do not see the Calling ID, I just see reserved in the phone?s display. -- Receba
[OSL | CCIE_Voice] ICD extension Not Showing up
You need to associate the ICD extension to the rmjtapi user. Once associated, you should se the number appear in ICD. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, September 07, 2008 10:53 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 31, Issue 34 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. ICD Extension Not Showing Up (Devildoc) 2. Re: Fast busy on unallocated number (Jonathan Charles) 3. Re: Failed GK Calls to IPCC Services (Jonathan Charles) -- Message: 1 Date: Sun, 7 Sep 2008 07:37:10 -0700 From: Devildoc [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] ICD Extension Not Showing Up To: CCIE Voice Online Study List ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I know there was a post on here a while back that has the solution to restart a certain IPCC Express service to have the ICD extension showing up when you try to associate a user to an ICD resource. Does anyone know what's the service name? I can't find it anywhere. It's not the CRS Engine service because i restarted that service and the ICD extension still didn't show up. Thanks. JD _ See how Windows connects the people, information, and fun that are part of your life. http://clk.atdmt.com/MRT/go/msnnkwxp1020093175mrt/direct/01/ -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080907/b6a 7f65a/attachment-0001.html -- Message: 2 Date: Sun, 7 Sep 2008 07:52:55 -0500 From: Jonathan Charles [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Fast busy on unallocated number To: Paul and Bobs [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Create a translation pattern that translates 18XX to 2000 and have 2000 as DNIS in Unity call routing to go to a CH that says not allocated... Jonathan On Sun, Sep 7, 2008 at 3:27 AM, Paul and Bobs [EMAIL PROTECTED] wrote: If my system is setup with teh following DID number range 617 302 1XXX and I only have DN setup for 617 302 10XX when someone tries to call a number with my range but that has not been allocated they get fast busy 617 302 1800 What I would like to try and do is create perhaps a CTI RP with DN 18XX and put a Call-Forward-All on this to voicemail, and try to get Unity to say somethings like this numebr is not available at this time .I am not sure what standard messages unity has. Does anyone have any ideas on this. -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080907/07a a1569/attachment-0001.html -- Message: 3 Date: Sun, 7 Sep 2008 08:31:08 -0500 From: Jonathan Charles [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Failed GK Calls to IPCC Services To: Devildoc [EMAIL PROTECTED] Cc: CCIE Voice Online Study List ccie_voice@onlinestudylist.com, Christian Hennrich [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 So, basically what we said initially CSS on translation pattern... Jonathan On Sun, Sep 7, 2008 at 8:16 AM, Devildoc [EMAIL PROTECTED] wrote: The reason why it worked when i changed to 4 digits was because i had my GK translation patterns in their own partition called pt-gk and the internal DNs in their own partition called pt-internal. So when I configured the GK trunk to accept the 10 digits, I only gave its CSS access to pt-gk and not the pt-internal. But when i configured it to accept 4 digits, the trunk didn't need to access any GK translation patterns, and therefore, I gave it access to pt-internal directly. JD -- Date: Fri, 5 Sep 2008 15:27:25 -0500 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Failed GK Calls to IPCC Services CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Wait... why would
[OSL | CCIE_Voice] fastbusy on unallocated number
Johnathan, I worked through this scenario last night. I don't think Unity has an un-allocated number prompt/greeting. I created a translation pattern of 18XX and a CTIport x1995(dummy phone)that was forwarded to voicemail. Created a call routing rule in Unity to send calls forwaded from 1995 to a call handler named unknownNumber. The call handler would then play the prompt that you recorded. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, September 07, 2008 10:53 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 31, Issue 34 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. ICD Extension Not Showing Up (Devildoc) 2. Re: Fast busy on unallocated number (Jonathan Charles) 3. Re: Failed GK Calls to IPCC Services (Jonathan Charles) -- Message: 1 Date: Sun, 7 Sep 2008 07:37:10 -0700 From: Devildoc [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] ICD Extension Not Showing Up To: CCIE Voice Online Study List ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I know there was a post on here a while back that has the solution to restart a certain IPCC Express service to have the ICD extension showing up when you try to associate a user to an ICD resource. Does anyone know what's the service name? I can't find it anywhere. It's not the CRS Engine service because i restarted that service and the ICD extension still didn't show up. Thanks. JD _ See how Windows connects the people, information, and fun that are part of your life. http://clk.atdmt.com/MRT/go/msnnkwxp1020093175mrt/direct/01/ -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080907/b6a 7f65a/attachment-0001.html -- Message: 2 Date: Sun, 7 Sep 2008 07:52:55 -0500 From: Jonathan Charles [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Fast busy on unallocated number To: Paul and Bobs [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Create a translation pattern that translates 18XX to 2000 and have 2000 as DNIS in Unity call routing to go to a CH that says not allocated... Jonathan On Sun, Sep 7, 2008 at 3:27 AM, Paul and Bobs [EMAIL PROTECTED] wrote: If my system is setup with teh following DID number range 617 302 1XXX and I only have DN setup for 617 302 10XX when someone tries to call a number with my range but that has not been allocated they get fast busy 617 302 1800 What I would like to try and do is create perhaps a CTI RP with DN 18XX and put a Call-Forward-All on this to voicemail, and try to get Unity to say somethings like this numebr is not available at this time .I am not sure what standard messages unity has. Does anyone have any ideas on this. -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080907/07a a1569/attachment-0001.html -- Message: 3 Date: Sun, 7 Sep 2008 08:31:08 -0500 From: Jonathan Charles [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Failed GK Calls to IPCC Services To: Devildoc [EMAIL PROTECTED] Cc: CCIE Voice Online Study List ccie_voice@onlinestudylist.com, Christian Hennrich [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 So, basically what we said initially CSS on translation pattern... Jonathan On Sun, Sep 7, 2008 at 8:16 AM, Devildoc [EMAIL PROTECTED] wrote: The reason why it worked when i changed to 4 digits was because i had my GK translation patterns in their own partition called pt-gk and the internal DNs in their own partition called pt-internal. So when I configured the GK trunk to accept the 10 digits, I only gave its CSS access to pt-gk and not the pt-internal. But when i configured it to accept 4 digits, the trunk didn't need to access any GK translation patterns, and therefore, I gave it access to
Re: [OSL | CCIE_Voice] fastbusy on unallocated number
Nice.. how did you do that? Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Sunday, September 07, 2008 6:14 PM To: Cardwell, Mark Cc: Hardesty, Scott; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] fastbusy on unallocated number I actually wanted to do it using the Annunciator and I figured it out... basically hacked the Annunciator to do it, but got it to play a custom prompt... Jonathan On Sun, Sep 7, 2008 at 4:40 PM, Cardwell, Mark [EMAIL PROTECTED] wrote: One otherway to do this is to create a CTI RoutePoint of 18XX forward all to VM. On Unity create a call routing rule (Forwarding) of 18** and route to a Call handler that plays what ever message you want it to play. Mark Cardwell | Systems Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 571.225.0132 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hardesty, Scott Sent: Sunday, September 07, 2008 11:20 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] fastbusy on unallocated number Johnathan, I worked through this scenario last night. I don't think Unity has an un-allocated number prompt/greeting. I created a translation pattern of 18XX and a CTIport x1995(dummy phone)that was forwarded to voicemail. Created a call routing rule in Unity to send calls forwaded from 1995 to a call handler named unknownNumber. The call handler would then play the prompt that you recorded. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, September 07, 2008 10:53 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 31, Issue 34 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. ICD Extension Not Showing Up (Devildoc) 2. Re: Fast busy on unallocated number (Jonathan Charles) 3. Re: Failed GK Calls to IPCC Services (Jonathan Charles) -- Message: 1 Date: Sun, 7 Sep 2008 07:37:10 -0700 From: Devildoc [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] ICD Extension Not Showing Up To: CCIE Voice Online Study List ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I know there was a post on here a while back that has the solution to restart a certain IPCC Express service to have the ICD extension showing up when you try to associate a user to an ICD resource. Does anyone know what's the service name? I can't find it anywhere. It's not the CRS Engine service because i restarted that service and the ICD extension still didn't show up. Thanks. JD _ See how Windows connects the people, information, and fun that are part of your life. http://clk.atdmt.com/MRT/go/msnnkwxp1020093175mrt/direct/01/ -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080907/b6a 7f65a/attachment-0001.html http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080907/b6 a7f65a/attachment-0001.html -- Message: 2 Date: Sun, 7 Sep 2008 07:52:55 -0500 From: Jonathan Charles [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Fast busy on unallocated number To: Paul and Bobs [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Create a translation pattern that translates 18XX to 2000 and have 2000 as DNIS in Unity call routing to go to a CH that says not allocated... Jonathan On Sun, Sep 7, 2008 at 3:27 AM, Paul and Bobs [EMAIL PROTECTED] wrote: If my system is setup with teh following DID number range 617 302 1XXX and I only have DN setup for 617 302 10XX when someone tries to call a number with my range but that has not been allocated they get fast busy 617 302 1800 What I
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 30, Issue 77
Johnathan, were you ever able to get your BACD welcome prompts to work? Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Sun 8/24/2008 6:56 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 30, Issue 77 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: B-ACD just dead air... (Jonathan Charles) 2. UniverCD (Michael Gross) 3. Re: UniverCD (Gary Kuhl) -- Message: 1 Date: Sun, 24 Aug 2008 11:24:33 -0500 From: Jonathan Charles [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] B-ACD just dead air... To: Chad Stachowicz [EMAIL PROTECTED] Cc: OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Current config: ! application service queue flash:app-b-acd-2.1.2.2.tcl param queue-len 15 param aa-hunt1 2000 param aa-hunt2 2001 param number-of-hunt-grps 2 param queue-manager-debugs 1 ! service aa flash:app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 paramspace english language en paramspace english location flash: param handoff-string aa param dial-by-extension-option 3 param voice-mail 4500 param welcome-prompt _bacd_welcome.au param service-name queue param aa-pilot 5000 Still works... but no prompts playing... also the hunt groups (pilot 2000 and 2001) work fine when you dial them directly (phones ring in sequence), but when you select option 1 or 2, both phones in hunt group ring at the same time... any idea why? Jonathan On Sat, Aug 23, 2008 at 5:47 PM, Chad Stachowicz [EMAIL PROTECTED]wrote: param welcome-prompt flash:en_bacd_welcome.au it should be param welcome-prompt flash:_bacd_welcome.au because it prepends the en with paramspace english language en HTH Chad On Sat, Aug 23, 2008 at 1:26 PM, Jonathan Charles [EMAIL PROTECTED]wrote: This is an H.323 gateway, source phone is CCM, destination is CCME... same behavior when calling from an IP phone on the ccme... Jonathan On Sat, Aug 23, 2008 at 2:12 PM, Stephen Collinson [EMAIL PROTECTED] wrote: How are you calling it? PSTN or VOIP g729 or g711u? Going out on a limb here, to perhaps save a few emails. If you are calling in remotely via the GK the incoming call is perhaps g729, depending on what you set on your trunk. This voip call needs an inbound g729 voip dp to match on. When you get the dead air. Do show call active voice comp to see what the call legs are doing. Also do debug voip appl script and see what you get. HTH S -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Jonathan Charles *Sent:* 23 August 2008 19:11 *To:* OSL CCIE Voice Lab Exam *Subject:* [OSL | CCIE_Voice] B-ACD just dead air... So, I configured B-ACD (from the config on Cisco's site...) and when I call it I get dead air... ! ! interface FastEthernet0/0 ip address 10.0.0.131 255.255.255.0 speed auto no cdp log mismatch duplex h323-gateway voip interface h323-gateway voip id home ipaddr 10.0.0.63 1719 h323-gateway voip h323-id CCME h323-gateway voip tech-prefix 2# h323-gateway voip bind srcaddr 10.0.0.131 ! ! application service queue flash:app-b-acd-2.1.2.2.tcl param queue-len 15 param aa-hunt3 2001 param queue-manager-debugs 1 param aa-hunt2 2000 param number-of-hunt-grps 2 ! service aa flash:app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 param number-of-hunt-grps 2 param handoff-string aa param dial-by-extension-option 1 paramspace english language en param max-time-vm-retry 2 param aa-pilot 5000 paramspace english location flash: param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param voice-mail 4500 param max-time-call-retry 700 param service-name queue ! global service alternate Default ! dial-peer voice 3983 voip service aa destination-pattern 5000 session target ipv4:10.0.0.131 incoming called-number 5000 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! ephone-hunt 1 sequential pilot 2000 list 3003, 3002 statistics collect ! ! ! !
[OSL | CCIE_Voice] BR1 MGCP T1 connection problem
All, I am having troubles getting the BR1 MGCP PRI operational. The controller is up and the serial interface is up but when running a debug q921 I do not see any traffic. L2 is TEI_Assigned. I expected to at lease see q921 on the link regardless of my configurations (assuming at a minimal PRI-group) but I am not seeing a thing. I must be missing something. Here is my config and if someone has a chance to look at it I would appreciate it. P5-BR1-RTR#sho run Building configuration... Current configuration : 3686 bytes ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname P5-BR1-RTR ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs-mz.124-3g.bin.bin boot-end-marker ! ! no aaa new-model ! resource policy ! network-clock-participate wic 0 network-clock-select 1 T1 0/0/0 ip subnet-zero ! ! ip cef no ip dhcp use vrf connected ! ip dhcp pool br1 import all network 10.5.201.0 255.255.255.0 default-router 10.5.201.1 option 150 ip 10.5.200.21 dns-server 10.5.200.22 ! ! ! isdn switch-type primary-ni ! voice-card 0 no dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,24 service mgcp ! ! ! ! ! interface Loopback0 ip address 172.5.101.1 255.255.255.255 ip ospf network point-to-point ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/1/0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/1/0.1 point-to-point ip address 162.5.101.2 255.255.255.0 ip ospf mtu-ignore frame-relay interface-dlci 101 ! interface FastEthernet1/0 switchport access vlan 150 switchport trunk native vlan 150 switchport mode trunk switchport voice vlan 250 mls qos trust dscp spanning-tree portfast ! interface FastEthernet1/1 shutdown ! interface FastEthernet1/2 switchport access vlan 150 switchport voice vlan 250 mls qos trust dscp spanning-tree portfast ! interface FastEthernet1/3 switchport access vlan 150 switchport voice vlan 250 mls qos trust dscp spanning-tree portfast ! interface FastEthernet1/4 switchport access vlan 150 switchport voice vlan 250 mls qos trust dscp spanning-tree portfast ! interface FastEthernet1/5 switchport access vlan 150 switchport voice vlan 250 mls qos trust dscp spanning-tree portfast ! interface FastEthernet1/6 switchport access vlan 150 switchport voice vlan 250 mls qos trust dscp spanning-tree portfast ! interface FastEthernet1/7 switchport access vlan 150 switchport voice vlan 250 mls qos trust dscp spanning-tree portfast ! interface FastEthernet1/8 switchport access vlan 150 switchport trunk native vlan 150 switchport mode trunk switchport voice vlan 250 mls qos trust dscp spanning-tree portfast ! interface FastEthernet1/9 shutdown ! interface FastEthernet1/10 shutdown ! interface FastEthernet1/11 shutdown ! interface FastEthernet1/12 shutdown ! interface FastEthernet1/13 shutdown ! interface FastEthernet1/14 shutdown ! interface FastEthernet1/15 switchport access vlan 250 no keepalive ! interface Vlan1 no ip address shutdown ! interface Vlan250 ip address 10.5.201.1 255.255.255.0 ! router ospf 1 log-adjacency-changes network 10.5.101.0 0.0.0.255 area 0 network 10.5.201.0 0.0.0.255 area 0 network 162.5.101.0 0.0.0.255 area 0 network 172.5.101.0 0.0.0.255 area 0 ! ip classless ! ! ip http server no ip http secure-server ! ! ! ! ! control-plane ! ! ! voice-port 0/0/0:23 ! ccm-manager mgcp ccm-manager config server 10.5.200.21 ! mgcp mgcp call-agent 10.5.200.21 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode cisco ! mgcp profile default ! ! ! ! ! ! gatekeeper shutdown ! ! line con 0 stopbits 1 line aux 0 stopbits 1 line vty 0 4 privilege level 15 no login transport input telnet line vty 5 15 no login transport input telnet ! warm-reboot scheduler allocate 2 1000 ! end P5-BR1-RTR# Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/
[OSL | CCIE_Voice] E1 configuration on POD25
All, I was working on Pod25 last night but was unable to get the E1 circuit to come up. I used the following configuration Controller e1 0/0/0 Ds0-group 1 timeslots 1-3 type r2-digital r2-semi-compelled ani I set framing =crc4 and line code = HDB3 but they do not show up in the config. Must be default setting. The controller would not come up and I saw the following error: Far End block error detected / Receive loss of frame With that said, when I initially setup the controller, I used compelled, not semi-compelled as the configuration so the framing was incorrect at 1st. It looks as is the PSTN E1 port went into an err disable type state. Can someone let me know if you have seen this in the past and lastly, how could I have cleared this issue? Thanks. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/
[OSL | CCIE_Voice] Proctor Labs phone support
I am checking to see if I can use 7911 / 7961 / 7940 IP Phones on the proctor labs call mangers / CME. I am going to use the lab equipment today and want to be sure that the phones I have will be supported. Thanks. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/
[OSL | CCIE_Voice] Meet-me conference and xcoders
Scenario: x2001-- BR1rtr-WAN-HQrtr--CCM(Meet-m e) The issue is that x2007 at BR1 is running g729 across the WAN and attempting to join a meet-me conference. I had the xcoder resource allocated to both the HQ-MRG and BR1-MRG but was unable to join the conference call. I ended up placing my xcoder in the HQ device pool (previously in BR1) and it my xcoding sessions started to work correctly. I had expected the MRGL to provide access to the xcoder when transcoding was required and did not expect the device pool assignment to come into play. Does this scenario sound accurate? It is counter-intuitive to me. Any additional feedback would be appreciated. Thanks.
[OSL | CCIE_Voice] IPIPGW Question
I am testing various scenarios between BR2 and HQ and have a few questions. While running deb voip ipipgw I do NOT see activity when using running SIP on both the inbound and outbound call legs. When using H323 on one leg and SIP on the other I see activity. Is this normal activity? Running deb voip dialpeer, I see the incoming and outgoing dial peers match and calls are processed correctly. Since I have SIP to SIP configured under voice services voip I expected to see IPIPGW activity. I welcome your thoughts! Thanks. . Scott Hardesty | Presidio Networked Solutions | E-mail [EMAIL PROTECTED] | Voice: 443-789-1219
[OSL | CCIE_Voice] Help on IPExpert Lab5
I am working on lab 5 of the IPExpert workbook and I can not seem to get audio to cut through when initiating a call from an HQ IP Phone to the ATA. The ATA is registered as a terminal to the GK. When calling from the HQ phone, the ATA rings and when answered, ringback stops on the IP phone so signaling seems to be working properly but the audio path never gets established. Any guidance on this would be appreciated. Thanks. . Scott Hardesty | Presidio Networked Solutions | E-mail [EMAIL PROTECTED] | Voice: 443-789-1219
Re: [OSL | CCIE_Voice] help on ipexpert 5
Thanks Christian. I checked those settings and all seemed correct. I ended up rebooting the HQ-RTR and restarting the CCM service as a shot in the dark and that seemed to clear the issue. Audio is cutting through correctly. I am not sure which reset actually fixed the problem because I did them at the same time. Thanks for your response! . Scott Hardesty | Presidio Networked Solutions | E-mail [EMAIL PROTECTED] | Voice: 443-789-1219 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, May 28, 2008 8:33 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 27, Issue 105 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: icd extension (Vik Malhi) 2. Re: B-ACD call fails ( Ovais Iqbal ) 3. Help on IPExpert Lab5 (Hardesty, Scott) 4. Re: Help on IPExpert Lab5 (Christian Narvaez) -- Message: 1 Date: Tue, 27 May 2008 20:58:58 -0700 From: Vik Malhi [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] icd extension To: 'OSL CCIE Voice Lab Exam' ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Jane- you most likely had problems with the www publishing service. When you restart IIS it also restarts this service too- maybe you the www publishing service never restarted and you could have manually started it. I know I run into this all the time. Greg- I would restart the IIS service taking note of the point raised here. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jane Ryer (jryer) Sent: Monday, May 26, 2008 11:18 AM To: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] icd extension Greg, During one of my study sessions, I totally screwed up my Pub server by restarting the IIS service, and the Proctor Labs guys could not fix it and had to revert my Pub. I couldn't bring up any Internet Explorer windows, including the CCM administration after I restarted IIS. What I have found is that if you restart the CRS engine twice - once after creating the jtapi service and then again after creating the rmjtapi service (as suggested by the GUI), then the ICD buttons always seem to show up. If you try to save time by not restarting the CRS engine after creating jtapi but before creating rmjtapi, that's when I seem to have trouble with the buttons not being there. I do not understand enough of the technical details of the various CRS services and how they interact with CCM to give you an explanation of why that is, I just know what seems to work. Jane _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Jost (grjost) Sent: Monday, May 26, 2008 11:37 AM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] icd extension What's the quick fix if the ICD extension doesn't appear? A reboot did it, but I'm sure a service restart will do it. I tried restarting CCM service. Is this the IIS service? I didn't think to try that before rebooting. Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922 -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080527/a10 a7dd5/attachment-0001.html -- Message: 2 Date: Wed, 28 May 2008 03:55:14 + From: Ovais Iqbal [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] B-ACD call fails To: Ovais Iqbal [EMAIL PROTECTED], Christian Narvaez [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Message-ID: 1814020281-1211946894-cardhu_decombobulator_blackberry.rim.net-84422251 [EMAIL PROTECTED] Content-Type: text/plain; charset=Windows-1252 Excellent, yes it works now. Thanks
