Re: [OSL | CCIE_Voice] Lab 12A - UCCX custom script
Hi Vik, Thanks for the answer. Cristal clear as always :) Roger Källberg Unified Communication Consultant Cygate AB From: Vik Malhi [mailto:vma...@ipexpert.com] Sent: den 25 januari 2010 01:33 To: Roger Källberg; OSL Group Subject: Re: [OSL | CCIE_Voice] Lab 12A - UCCX custom script Roger, I think the wording of sending the second caller in queue to AA is a little misleading. Better wording would tell you whilst there is 1 contact in the queue all subsequent callers should be sent to AA. In other words the second caller should not even invoke the select resource step- we should find out in advance of that step whether there is a contact in the queue. -- Vik Malhi - CCIE #13890 Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com . From: Roger Källberg roger.kallb...@cygate.se Date: Sun, 24 Jan 2010 20:27:47 +0100 To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab 12A - UCCX custom script No one that has any thought about this? Roger Källberg Unified Communication Consultant Cygate AB From: Roger Källberg [mailto:roger.kallb...@cygate.se] Sent: den 22 januari 2010 09:24 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab 12A - UCCX custom script Hi guys and girls, (if there are any around, I mean except for Amy :))! I need to get some clarification on the last criteria under q12.2, it's phrased this way. There can only be a maximum of one caller in queue. The second caller being placed into the queue should be sent to UC AutoAttendant instead of hearing queue prompt. I have no issues getting this to work, but my question is what EXACTLY the phrasing aims are as an end result? The first sentence says There can only be a maximum of one caller in queue. and the second says The second caller being placed into the queue, this is what I need to get some clarification about. What I mean is this, should the if statement with the check of call in queue = max queue depth that redirect (goto Xfer2AA) be before the select resource step, as the first sentence in the above phrasing would suggest. Or should it be under the queued sub step in the select resource, or whatever it's called, I guess u guys know what I mean? As one might interpret the second sentence in the question if you look at that part and do EXACTLY as it says. Namely that the second caller should be placed into the queue, but then be sent to the AA before the queue prompt begins to play. As I said I have no problem with the intended functionality, I just want the end result clarified. I mean there is a big difference in the two ways of doing it, not in the sense of result for the callers, but in the real lab these little nuances can bite you in the butt if you're not careful with this kind of iffy questions. Brgds, Roger Källberg Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se mailto:roger.kallb...@cygate.se ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP load for 7941/61 and CME
Hi Bill, This site might be helpful for you: Cisco Unified CME 7.0(1) Supported Firmware, Platforms, Memory, and Voice Products http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/c me701spc.htm cheers, sd Steve Denney, CISSP Systems Engineer - Technology Solutions Network Voice and Unified Communications Products Cisco Systems, Inc. 125 High Street, 21st Floor Boston, MA 02110 978-936-4048 (Office) 617-872-5031 (Mobile) stden...@cisco.com mailto:stden...@cisco.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill Hatcher Sent: Monday, January 25, 2010 6:58 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] SIP load for 7941/61 and CME Can someone please tell me what load to use for SIP on the CME for the Cisco 7941/61 phone model? I see 3 types of files to download, the .zip the .cop and the .cop.sgn file on Cisco's site, but they all referance CallManager only. Bill ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calling Number Transformation Behavior
Hi David, It is recommended to create separate pt/css for both cg and cd xform patterns, we have seen this unexpected behavior before in scenarios like yours, so please try to separate pt and css and let us know, BR, On Sat, Jan 23, 2010 at 11:12 PM, David Wagner unifiedd...@gmail.comwrote: I am trying to transform a DN on my egress gateway (MGCP PRI) I have used DP transform CSS unchecked and a hard coded CSS of HQ_Xform which has access to H!_Xform_PT for both calling and called transformation CSS. I have a calling number transformation mask of 1XXX which is set to use external phone number mask and then mask it done to 7 digits using XXX. It does not work unless i hard code the transformation pattern to 1001 (Or another DN) then it works fine. I have rebooted the cluster and deleted and re-added the pattern a few times no go unless hard coded exact match. Anyone else see this? TIA Dave ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol1 Question 5.20 Mobile Voice Access Questions
Hi Viyay, 1.- When calling out from MVA the ucm will use RDP Calling Search Space. In your case that CSS should have access to the RDN 2.- You are right, the 5999 number is obtained from the xml application loaded in the router from ucm HTH, On Sat, Jan 23, 2010 at 2:00 PM, Vijay vsbal...@yahoo.com wrote: 1) Need to know what partition will MVA use when calling the PSTN 916178632683. In my lab, for Inbound H.323 GW CSS, I use css_internal which has partitions pt-internal and ptsupport-hq. It has no access to RP 9.1[2-9]XX[2-9]XXhttps://10.88.154.16:8443/ccmadmin/routePattern2Edit.do?key=ee282b4c-1e88-a31e-75fa-a278db457451in pt-hq-ld. But, Still option1 and call to 916178632683 worked. PG was using css-hq-ld. Can some one explain this? 2) In the Dial-peer, I removed the no-digits strip from Inbound Pots DP and call still worked. By default, pots DP will strip the matching digits. dial-peer voice 2 pots service cmm incoming called-number 2123945999 How does call go from Inbound DP to outbound DP dial-peer voice 5010 voip. Is the number 5999 got from IVR? Thanks, Vijay ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Extension Mobility - XML Error [4] - parse error
Hi Allen, Did you make sure the service URL is: http://10.10.210.10:8080/emapp/EMAppServlet?device=#DEVICENAME# I could reproduce the same behavior configuring just a part of that url, e.g. http://10.10.210.10:8080/emapp/EMAppServlet?device= BR, On Fri, Jan 22, 2010 at 6:56 PM, Allen Su yenlins...@hotmail.com wrote: Hi folks, I ran into a weird problem in Lab Vol1 9.4 – EM and IPMA. When I tried to access the EM service on the IP Phone, I got this message on the phone screen “XML Error [4] – parse error”, and I couldn’t access the EM service. Has anyone also ran into this? I tried to search this online and found this troubleshooting guide “ http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080093e72.shtml”, and this exact error is listed in the guide. But I could not make any sense out of the solution it provided. *XML Error [4] Parse Error is returned when selecting the login service.** * - *Problem:* The form.jsp downloaded includes HTTP header information. - *Solution:* On this page, right-click on the *form.jsp*, then select *Save Link As* or *Save Target As*. Select the location to download the form. Ensure that the first line of the form.jsp page reads: *%@ page import=java.net.InetAddress %* Any suggestions? Thanks for your help! Allen Su ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP load for 7941/61 and CME (UNCLASSIFIED)
Classification: UNCLASSIFIED Caveats: FOUO Wow - One that I can answer. Bill - I had the exact same issue. Download the zip file. The images files are located within. I know that it says CUCM only, that is what threw me also Jeff --- Jeffrey T. Girard (Jeff) COL, 53 Future Forces Integration Directorate (FFID), Deputy - Networks office: (915)568-1240 DSN 978 Mobile: (915)727-4222 reply to: jeffrey.gir...@us.army.mil -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill Hatcher Sent: Monday, January 25, 2010 4:58 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] SIP load for 7941/61 and CME Can someone please tell me what load to use for SIP on the CME for the Cisco 7941/61 phone model? I see 3 types of files to download, the .zip the .cop and the .cop.sgn file on Cisco's site, but they all referance CallManager only. Bill Classification: UNCLASSIFIED Caveats: FOUO ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MULTICAST MOH over SIP TRUNK?
After further testing I have confirmed that is indeed the IOS version that is not allowing for the Multicast MOH over a SIP trunk to a PSTN Termination. Tested versions: c2801-adventerprisek9_ivs-mz.124-20.T4.bin = PSTN MOH DOESNT WORK c2801-adventerprisek9_ivs-mz.124-22.T.bin = PSTN MOH WORKS Typology SIP/SKINNYPHONE - UCM (Multicast-PUB/MGRL) - SIPTRUNK - (HQ-2801 sourced Multicast MOH) PRI - (PSTN-2821) -PSTNPHONE If any one know anything to the contrary to my findings please respond as I am under the assumption that this is the final outcome. Thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MOH from flash to PSTN
Hello: I've two scenarios: 1- BR1 phone calls PSTN phone: Then I press hold on BR1 phone and I hear moh from flash on PSTN phone (expected behaviour) BR1#sh ccm-manager music-on-hold Current active multicast sessions : 1 Multicast RTP port Packets Call CodecIncoming Address number in/outid Interface === 239.1.1.1 16384 476/476 34 g711ulaw Lo0 2- PSTN phone calls BR1 phone: Then I press hold on BR1 phone and I hear 'beep beep beep Is this the normal situation or I shoul hear moh too in this case? BR1#sh ccm-manager music-on-hold Current active multicast sessions : 0 My config: ccm-manager music-on-hold call-manager-fallback ip source-address 10.2.30.254 port 2000 max-ephones 2 max-dn 2 dual-line moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 10.2.30.254 172.2.1.254 ! voice vlan and loo0 Thanks in advance _ http://www.quemovileres.com/ ¡Descubre qué Móvil eres! Hay uno hecho para ti.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CME files for phones
Hi, I want to know what are the four files in CME? I don't work with CME and I am looking at lab 3A .bin .loads .sb2 .sbn Thanks, Randall -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Tuesday, January 26, 2010 9:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 47, Issue 127 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Call Forward not working (Sivakumar Mahalingam) 2. Re: MULTICAST MOH over SIP TRUNK? (Vccie Vccie) -- Message: 1 Date: Tue, 26 Jan 2010 11:22:32 -0500 From: Sivakumar Mahalingam sima...@gmail.com Subject: [OSL | CCIE_Voice] Call Forward not working To: OSL Group ccie_voice@onlinestudylist.com Message-ID: 703b51d01001260822y1df82b73se07cd3463a9f3...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi All, I need some help for the below issue that am facing. I have CUCM 7.1.3(a) running on my VoIP network and the issue is ,when i setup forward all for Extn A to Extn B ,the off campus calls are forwarded to Extn B correctly and the on campus calls are not being forwarded and it rings the Extn A phone directly. If anyone of you have faced a simillar kind of problem,please let me know you thoughts. Thanks, Simah. -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100126/da62b3f3/attachment-0001.htm -- Message: 2 Date: Tue, 26 Jan 2010 10:29:43 -0600 From: Vccie Vccie voiceccie2...@gmail.com Subject: Re: [OSL | CCIE_Voice] MULTICAST MOH over SIP TRUNK? To: ccie_voice@onlinestudylist.com Message-ID: 8adf63bc1001260829x19043ec8kde845eed89336...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 After further testing I have confirmed that is indeed the IOS version that is not allowing for the Multicast MOH over a SIP trunk to a PSTN Termination. Tested versions: c2801-adventerprisek9_ivs-mz.124-20.T4.bin = PSTN MOH DOESNT WORK c2801-adventerprisek9_ivs-mz.124-22.T.bin = PSTN MOH WORKS Typology SIP/SKINNYPHONE - UCM (Multicast-PUB/MGRL) - SIPTRUNK - (HQ-2801 sourced Multicast MOH) PRI - (PSTN-2821) -PSTNPHONE If any one know anything to the contrary to my findings please respond as I am under the assumption that this is the final outcome. Thank you -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100126/97cae30a/attachment-0001.htm -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 47, Issue 127 *** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] fair-queue when configuring FRF.12
Hi Sean, This is the router's expected behavior whenever you configure fragmentation (frame-relay fragmentation) on a frame-relay class map (and haven't assigned any service policy to it yet), please take a look at the command reference: http://www.cisco.com/en/US/docs/ios/wan/command/reference/wan_f1.html#wp1013944 As you noticed, if you already configured a hierarchical service policy into a FR map-class and configure the frame-relay fragment, the command suddenly appears. However, if you do the same with a non-hierarchical policy the frame-relay fair-command will not be there automatically, To sum up, if the FR map-class is directly attached to a hierarchical service policy or hasn't any configured before the introduction of the fragment command, the frame-relay fair-queue will appear, In any case, I think this command won't hurt as the configured service policy (whether it is hierarchical or not) queue configuration will take effect, Hope this makes sense, On Sun, Jan 24, 2010 at 4:25 PM, sean hurricane shurric...@gmail.comwrote: When configuring FRF.12 using class based shaping is it normal for frame relay to auto magically append fair queue, i have been noticing this problem since last week and can't turn it off. i have tried turning it off on the physical interface but it show back up again, see below HQ#sh run Building configuration... Current configuration : 4076 bytes ! ! Last configuration change at 12:57:58 PST Sun Jan 24 2010 ! NVRAM config last updated at 12:58:05 PST Sun Jan 24 2010 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname HQ ! boot-start-marker warm-reboot boot-end-marker ! logging message-counter syslog ! no aaa new-model clock timezone PST -8 clock summer-time PST recurring network-clock-participate wic 1 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 10.10.200.1 10.10.200.49 ip dhcp excluded-address 10.10.200.70 10.10.200.254 ip dhcp excluded-address 10.10.201.1 10.10.201.49 ip dhcp excluded-address 10.10.201.70 10.10.201.254 ip dhcp excluded-address 10.10.202.1 10.10.202.49 ip dhcp excluded-address 10.10.202.70 10.10.202.254 ! ip dhcp pool HQ network 10.10.200.0 255.255.255.0 default-router 10.10.200.3 option 150 ip 10.10.210.11 ! ip dhcp pool BR1 network 10.10.201.0 255.255.255.0 default-router 10.10.201.1 option 150 ip 10.10.210.11 ! ip dhcp pool BR2 network 10.10.202.0 255.255.255.0 default-router 10.10.202.1 option 150 ip 10.10.210.11 ! ! no ip domain lookup ip domain name ipexpert.com no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-ni ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! voice-card 0 ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller T1 0/1/0 pri-group timeslots 1-3,24 service mgcp ! controller T1 0/1/1 ! ! class-map match-any MEDIA match protocol sip match protocol h323 match protocol mgcp match protocol rsvp match protocol skinny class-map match-any RTP match protocol rtp audio match protocol rtcp ! ! policy-map LLQ class RTP priority 23 compress header ip rtp set dscp ef class MEDIA bandwidth 17 set dscp af31 class class-default fair-queue policy-map SHAPE class class-default shape average 364800 3648 service-policy LLQ ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.0 ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/0.10 encapsulation dot1Q 10 native ip address 10.10.100.1 255.255.255.0 ! interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 ! interface FastEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing display-ie no cdp enable ! interface Serial0/2/0 no ip address shutdown no fair-queue clock rate 200 ! interface Serial0/3/0 no ip address encapsulation frame-relay ! interface Serial0/3/0.1 point-to-point ip address 10.10.111.1 255.255.255.0 snmp trap link-status frame-relay interface-dlci 102 class BR1 ! interface Serial0/3/0.2 point-to-point ip address 10.10.112.1 255.255.255.0 snmp trap link-status frame-relay interface-dlci 103 ! router ospf 10 log-adjacency-changes network 0.0.0.0 255.255.255.255 area 0 ! ip forward-protocol nd ip http server no ip http secure-server ! ! ! ! map-class frame-relay BR1 frame-relay fragment 480 frame-relay fair-queue service-policy output SHAPE ! ! ! ! ! ! ! control-plane ! ! !
Re: [OSL | CCIE_Voice] MOH from flash to PSTN
Maybe it is for some reason a codec mismatch. So maybe your MOH Server is using G711 and the Gateway is using G729. Please try to allow G729 as well on the IP Voice Media Streaming App and check if you still hear beep when calling from the PSTN phone. If you hear nothing, then do a sh ccm-manager music-on-hold again to check if the multicast address is 239.1.1.3. Which protocol are you using, MGCP or H.323? HTH, Omar Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Angel Perez Gesendet: Dienstag, 26. Januar 2010 18:13 An: ccie_voice@onlinestudylist.com Betreff: [OSL | CCIE_Voice] MOH from flash to PSTN Hello: I've two scenarios: 1- BR1 phone calls PSTN phone: Then I press hold on BR1 phone and I hear moh from flash on PSTN phone (expected behaviour) BR1#sh ccm-manager music-on-hold Current active multicast sessions : 1 Multicast RTP port Packets Call CodecIncoming Address number in/outid Interface === 239.1.1.1 16384 476/476 34 g711ulaw Lo0 2- PSTN phone calls BR1 phone: Then I press hold on BR1 phone and I hear 'beep beep beep Is this the normal situation or I shoul hear moh too in this case? BR1#sh ccm-manager music-on-hold Current active multicast sessions : 0 My config: ccm-manager music-on-hold call-manager-fallback ip source-address 10.2.30.254 port 2000 max-ephones 2 max-dn 2 dual-line moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 10.2.30.254 172.2.1.254 ! voice vlan and loo0 Thanks in advance _ ¡Nuevo MSN Deportes! Sigue los partidos en directo y encuentra la última http://deportes.es.msn.com/ información de tus equipos favoritos. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 47, Issue 128-Re: MOH from flash to PSTN
You need to force the g711 on your dial-peers. I am assuming you are using H323 so just include codec g711 on voip dialpeers pointing to CUCM. Thanks -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Tuesday, January 26, 2010 1:47 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 47, Issue 128 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. MOH from flash to PSTN (Angel Perez) 2. CME files for phones (Randall Crumm) 3. Re: fair-queue when configuring FRF.12 (Otto Sanchez) 4. Re: MOH from flash to PSTN (Omar Dahmani) -- Message: 1 Date: Tue, 26 Jan 2010 17:12:38 + From: Angel Perez gorr...@hotmail.com Subject: [OSL | CCIE_Voice] MOH from flash to PSTN To: ccie_voice@onlinestudylist.com Message-ID: col110-w65124ac136eea6fcbe80b8a1...@phx.gbl Content-Type: text/plain; charset=iso-8859-1 Hello: I've two scenarios: 1- BR1 phone calls PSTN phone: Then I press hold on BR1 phone and I hear moh from flash on PSTN phone (expected behaviour) BR1#sh ccm-manager music-on-hold Current active multicast sessions : 1 Multicast RTP port Packets Call CodecIncoming Address number in/outid Interface === 239.1.1.1 16384 476/476 34 g711ulaw Lo0 2- PSTN phone calls BR1 phone: Then I press hold on BR1 phone and I hear 'beep beep beep Is this the normal situation or I shoul hear moh too in this case? BR1#sh ccm-manager music-on-hold Current active multicast sessions : 0 My config: ccm-manager music-on-hold call-manager-fallback ip source-address 10.2.30.254 port 2000 max-ephones 2 max-dn 2 dual-line moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 10.2.30.254 172.2.1.254 ! voice vlan and loo0 Thanks in advance _ http://www.quemovileres.com/ ?Descubre qu? M?vil eres! Hay uno hecho para ti. -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100126/8c353027/attachment-0001.htm -- Message: 2 Date: Tue, 26 Jan 2010 09:47:40 -0800 From: Randall Crumm randall.cr...@harmonicinc.com Subject: [OSL | CCIE_Voice] CME files for phones To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: 9473270a65ca67458d287f3da3c9f37d0f008f0...@exch-cms.hlit.local Content-Type: text/plain; charset=us-ascii Hi, I want to know what are the four files in CME? I don't work with CME and I am looking at lab 3A .bin .loads .sb2 .sbn Thanks, Randall -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Tuesday, January 26, 2010 9:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 47, Issue 127 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Call Forward not working (Sivakumar Mahalingam) 2. Re: MULTICAST MOH over SIP TRUNK? (Vccie Vccie) -- Message: 1 Date: Tue, 26 Jan 2010 11:22:32 -0500 From: Sivakumar Mahalingam sima...@gmail.com Subject: [OSL | CCIE_Voice] Call Forward not working To: OSL Group ccie_voice@onlinestudylist.com Message-ID: 703b51d01001260822y1df82b73se07cd3463a9f3...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi All, I need some help for the below issue that am facing. I have CUCM 7.1.3(a) running on my VoIP network and the issue is ,when i setup forward all for Extn A to Extn B ,the off campus calls are forwarded to Extn B correctly and the on campus calls are not being forwarded and it rings the Extn
Re: [OSL | CCIE_Voice] MOH from Flash
Hi, The branch router will always multicast moh in g.711 format from the flash. So, branch phones should be configured to hear the 239.1.1.1 port 16384 and not 239.1.1.3:16384 and codec g.729 (this is a not supported configuration) when they are put on hold. How do you guarantee your branch phones will tune 239.1.1.1:16384 in g.711 format and still be able to communicate with other sites using g.729?. Well that's when the g.711 region/DP show up, configure a region such as the codec relationship with any other region is g.711, assign this region to a new DP and then assign that DP to the MOH server configured for the siteB DP MRGL-MRG, I still trying to find a reason why you will need to stream g.729 from the router flash in this scenario, you might want to do that if the moh streaming is coming from the hq site, There's a very good document about this feature, I reference it just in case you haven't take a look at it, http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cusrst/admin/srst/configuration/guide/srs_moh.html HTH, On Mon, Jan 25, 2010 at 8:03 PM, vccie2010 vccie2...@gmail.com wrote: Thanks Otto for yoru kind reponse, so let me put the question otherway , what should I do on the UCM and SiteB router if I was to use MOH from flash and it shd be G729. Is this scenario possible please as we say on SiteB router the IP address will be 239.1.1.3 and I think in this case all the configs on UCM remain same, right pls? 2010/1/25 Otto Sanchez o...@ipexpert.com Hi, The only format supported by the moh file is g.711, you should also take care of the multicast ip address numbering to make it match to a g.711 moh streaming not g.729, BR, ,2010/1/22 Roger Källberg roger.kallb...@cygate.se As far as I know you need to have the file(s) in the correct format in flash, ie G729 if you want to use that for your MOH or G711U if that's what you want. Vik please correct me if I'm wrong. Brgds, *Roger Källberg* Consultant Cygate AB -- *Från:* vccie2010 [vccie2...@gmail.com] *Skickat:* den 22 januari 2010 04:18 *Till:* Vik Malhi *Kopia:* OSL Group *Ämne:* Re: [OSL | CCIE_Voice] MOH from Flash Also Vik, is it must to have MOH flash file in G729 format if I need to have G729 MOH from flash ??? I am confused here...Could you please help me here. On Wed, Jan 20, 2010 at 9:41 PM, vccie2010 vccie2...@gmail.com wrote: I was trying to test MOH from flash on BR1. G711 MOH works fine butI am having problem when trying config and test G729. You said There should be no reason why you can’t play a g729 file from the flash if you have a music file in that format on the flash ..so does it mean I need to have MOH file shd be in G729 format so that the MOH from flash is G729 MOH, right ? On Wed, Jan 20, 2010 at 8:51 PM, Vik Malhi vma...@ipexpert.com wrote: I’m not fully understanding the scenario. When you press hold from a PSTN phone then you are not testing your MOH settings. Can you confirm? When you are sourcing from the BR1 flash I would guess you would ALWAYS want it to be g711 since it is now local to the BR1 site. That’s one of the main reasons for doing this- we can play g711 music without any bandwidth being tied up. There should be no reason why you can’t play a g729 file from the flash if you have a music file in that format on the flash and you use the multicast moh 239.1.1.3 port 16384 route favoicesubinterface IP Add lo0 IP Addr command and increment on ip address on the UCM, use a single server (the one which has the base multicast ip of 239.1.1.1) and audio source 1. -- Vik Malhi – CCIE #13890 Instructor - IPexpert, Inc. Mailto: *vma...@ipexpert.com *Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat * http://www.ipexpert.com/chat* IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities * http://www.ipexpert.com/communities* and our public website at www.ipexpert.com *http://www.ipexpert.com* . -- *From: *vccie2010 vccie2...@gmail.com *Date: *Wed, 20 Jan 2010 20:22:50 -0800 *To: *OSL Group ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] MOH from Flash Vik, so you mean whether we want G729 MOH or G711MOH to be played on BR1 from flash we should ALWAYS place MOH server in G711only. I did that and with that G711 MOH works perfectly fine when I put either HQ phone or PSTN phone on hold. But I am having problem work G729 MOH heard on BR1 phone when I place it on hold from HQ phone ( there is dead silence) while G729 MOH from works fine when I place BR1 phone on hold from PSTN ( I hear G729 Music on BR1 phone)
[OSL | CCIE_Voice] CUCME SIP Issues
Scenario: I have an X-Lite softphone setup with a dn of 20004. I also setup another dn of 20005 to call forward all to 20004. The dn of 20005 is not assigned to another phone. In this scenario, there is only one phone registered to the CUCME SIP instance. Problem: I go off-hook, dial 2-0-0-0-5 and receive a Call failed: Not Acceptable Media Debug: SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/UDP 10.25.3.210:55446;branch=z9hG4bK-d8754z-262a3d2cba0faf54-1---d8754z-;rport From: CUCME SIPsip:20...@10.25.3.200;tag=4c43d170 To: 20005sip:20...@10.25.3.200;tag=1B1F3408-1F39 Date: Wed, 27 Jan 2010 03:45:22 GMT Call-ID: M2JmNGU0OTZhM2Y4MzJhNTg2YmY5NDc4NmFlZjI2ZmU. Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Reason: Q.850;cause=65 Content-Length: 0 Question 1: Why does it give me the 488 error? Question 2: Do DNs need to be assigned to working phones in order for calls to be directed to them? If so, what happens if a SIP phone with said dn loses network connectivity? Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCME SIP Issues
Matthew - How are the Vol 1 Video Solutions working out? Keep in touch! Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert Mailto: wlaw...@ipexpert.com Telephone: +1.810.326.1444, ext. 101 Live Assistance, Please visit: www.ipexpert.com/chat eFax: +1.810.454.0130 ::Message sent from iPhone:: IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com. On Jan 26, 2010, at 10:49 PM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Scenario: I have an X-Lite softphone setup with a dn of 20004. I also setup another dn of 20005 to call forward all to 20004. The dn of 20005 is not assigned to another phone. In this scenario, there is only one phone registered to the CUCME SIP instance. Problem: I go off-hook, dial 2-0-0-0-5 and receive a Call failed: Not Acceptable Media Debug: SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/UDP 10.25.3.210:55446;branch=z9hG4bK- d8754z-262a3d2cba0faf54-1---d8754z-;rport From: CUCME SIPsip:20...@10.25.3.200;tag=4c43d170 To: 20005sip:20...@10.25.3.200;tag=1B1F3408-1F39 Date: Wed, 27 Jan 2010 03:45:22 GMT Call-ID: M2JmNGU0OTZhM2Y4MzJhNTg2YmY5NDc4NmFlZjI2ZmU. Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Reason: Q.850;cause=65 Content-Length: 0 Question 1: Why does it give me the 488 error? Question 2: Do DNs need to be assigned to working phones in order for calls to be directed to them? If so, what happens if a SIP phone with said dn loses network connectivity? Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 952 221 2814| mjbe...@krollontrack.com www.krollontrack.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCME SIP Issues
It means that the invite SDP did not match with the incoming dial peers sdp. i.e. there is a codec mismatch. So look at the incoming dial peer and check the codec in the invite and match it approprately On Tue, Jan 26, 2010 at 9:52 PM, Wayne Lawson groupst...@ipexpert.comwrote: Matthew - How are the Vol 1 Video Solutions working out? Keep in touch! Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert Mailto: wlaw...@ipexpert.comwlaw...@ipexpert.com Telephone: +1.810.326.1444, ext. 101 Live Assistance, Please visit: http://www.ipexpert.com/chat www.ipexpert.com/chat eFax: +1.810.454.0130 ::Message sent from iPhone:: IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com. On Jan 26, 2010, at 10:49 PM, Berry, Matthew J. mjbe...@krollontrack.com wrote: *Scenario:* I have an X-Lite softphone setup with a dn of 20004. I also setup another dn of 20005 to call forward all to 20004. The dn of 20005 is not assigned to another phone. In this scenario, there is only one phone registered to the CUCME SIP instance. *Problem:* I go off-hook, dial 2-0-0-0-5 and receive a Call failed: Not Acceptable Media *Debug:* SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/UDP 10.25.3.210:55446 ;branch=z9hG4bK-d8754z-262a3d2cba0faf54-1---d8754z-;rport From: CUCME SIPsip:20...@10.25.3.200 sip%3a20...@10.25.3.200 ;tag=4c43d170 To: 20005sip:20...@10.25.3.200 sip%3a20...@10.25.3.200 ;tag=1B1F3408-1F39 Date: Wed, 27 Jan 2010 03:45:22 GMT Call-ID: M2JmNGU0OTZhM2Y4MzJhNTg2YmY5NDc4NmFlZjI2ZmU. Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Reason: Q.850;cause=65 Content-Length: 0 *Question 1:* Why does it give me the 488 error? *Question 2:* Do DNs need to be assigned to working phones in order for calls to be directed to them? If so, what happens if a SIP phone with said dn loses network connectivity? Thanks, * * *Matthew Berry*, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 952 221 2814| mjbe...@krollontrack.com agutz...@krollontrack.com www.krollontrack.com ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] ProctorLAB VRack VOICE V3 - Unable to get access to Servers
Hello Guys, I am unable to get access to all the application servers at VRack v3 from my home pc via VPN . Does any specific settings to config on the HQ switch? I tried to load final configs and this problem still remain. I reported this to lab support but doesn't help much. Cheers, Jerry _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/196390709/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] ProctorLAB VRack VOICE V3 - Unable to get access to Servers
What rack are you on right now? I only see a Christopher Ring on the voice rack besides the bootcamp students. Where you connected to the VPN? Hardware or Software? Can you please provide more information. When replying to me please reply directly to me to remove this conversation from the mailing list. Regards, Tyson Scott - CCIE #13513 RS, Security, and SP Technical Instructor - IPexpert, Inc. Mailto: mailto:tsc...@ipexpert.com tsc...@ipexpert.com Telephone: +1.810.326.1444, ext. 208 Live Assistance, Please visit: http://www.ipexpert.com/chat www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at http://www.ipexpert.com/communities www.ipexpert.com/communities and our public website at http://www.ipexpert.com www.ipexpert.com From: Bai Min jerry...@hotmail.com Date: January 26, 2010 10:58:50 PM EST To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] ProctorLAB VRack VOICE V3 - Unable to get access to Servers Hello Guys, I am unable to get access to all the application servers at VRack v3 from my home pc via VPN . Does any specific settings to config on the HQ switch? I tried to load final configs and this problem still remain. I reported this to lab support but doesn't help much. Cheers, Jerry _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign http://clk.atdmt.com/GBL/go/196390709/direct/01/ up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MOH from Flash
HI Otto, So in which scenario I will be using 239.1.1.3:16384 on SiteB router please ? Thanks // AK On Tue, Jan 26, 2010 at 12:22 PM, Otto Sanchez o...@ipexpert.com wrote: Hi, The branch router will always multicast moh in g.711 format from the flash. So, branch phones should be configured to hear the 239.1.1.1 port 16384 and not 239.1.1.3:16384 and codec g.729 (this is a not supported configuration) when they are put on hold. How do you guarantee your branch phones will tune 239.1.1.1:16384 in g.711 format and still be able to communicate with other sites using g.729?. Well that's when the g.711 region/DP show up, configure a region such as the codec relationship with any other region is g.711, assign this region to a new DP and then assign that DP to the MOH server configured for the siteB DP MRGL-MRG, I still trying to find a reason why you will need to stream g.729 from the router flash in this scenario, you might want to do that if the moh streaming is coming from the hq site, There's a very good document about this feature, I reference it just in case you haven't take a look at it, http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cusrst/admin/srst/configuration/guide/srs_moh.html HTH, On Mon, Jan 25, 2010 at 8:03 PM, vccie2010 vccie2...@gmail.com wrote: Thanks Otto for yoru kind reponse, so let me put the question otherway , what should I do on the UCM and SiteB router if I was to use MOH from flash and it shd be G729. Is this scenario possible please as we say on SiteB router the IP address will be 239.1.1.3 and I think in this case all the configs on UCM remain same, right pls? 2010/1/25 Otto Sanchez o...@ipexpert.com Hi, The only format supported by the moh file is g.711, you should also take care of the multicast ip address numbering to make it match to a g.711 moh streaming not g.729, BR, ,2010/1/22 Roger Källberg roger.kallb...@cygate.se As far as I know you need to have the file(s) in the correct format in flash, ie G729 if you want to use that for your MOH or G711U if that's what you want. Vik please correct me if I'm wrong. Brgds, *Roger Källberg* Consultant Cygate AB -- *Från:* vccie2010 [vccie2...@gmail.com] *Skickat:* den 22 januari 2010 04:18 *Till:* Vik Malhi *Kopia:* OSL Group *Ämne:* Re: [OSL | CCIE_Voice] MOH from Flash Also Vik, is it must to have MOH flash file in G729 format if I need to have G729 MOH from flash ??? I am confused here...Could you please help me here. On Wed, Jan 20, 2010 at 9:41 PM, vccie2010 vccie2...@gmail.com wrote: I was trying to test MOH from flash on BR1. G711 MOH works fine butI am having problem when trying config and test G729. You said There should be no reason why you can’t play a g729 file from the flash if you have a music file in that format on the flash ..so does it mean I need to have MOH file shd be in G729 format so that the MOH from flash is G729 MOH, right ? On Wed, Jan 20, 2010 at 8:51 PM, Vik Malhi vma...@ipexpert.comwrote: I’m not fully understanding the scenario. When you press hold from a PSTN phone then you are not testing your MOH settings. Can you confirm? When you are sourcing from the BR1 flash I would guess you would ALWAYS want it to be g711 since it is now local to the BR1 site. That’s one of the main reasons for doing this- we can play g711 music without any bandwidth being tied up. There should be no reason why you can’t play a g729 file from the flash if you have a music file in that format on the flash and you use the multicast moh 239.1.1.3 port 16384 route favoicesubinterface IP Add lo0 IP Addr command and increment on ip address on the UCM, use a single server (the one which has the base multicast ip of 239.1.1.1) and audio source 1. -- Vik Malhi – CCIE #13890 Instructor - IPexpert, Inc. Mailto: *vma...@ipexpert.com *Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat * http://www.ipexpert.com/chat* IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities * http://www.ipexpert.com/communities* and our public website at www.ipexpert.com *http://www.ipexpert.com* . -- *From: *vccie2010 vccie2...@gmail.com *Date: *Wed, 20 Jan 2010 20:22:50 -0800 *To: *OSL Group ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] MOH from Flash Vik, so you mean whether we want G729 MOH or G711MOH to be played on BR1 from flash we should ALWAYS place MOH server in G711only. I did that and with that G711 MOH works perfectly fine when I put either HQ phone or PSTN phone on hold. But I am having problem work
