Re: [OSL | CCIE_Voice] Thank you
Congrats Mark well deserved Sent from my iPhone On Nov 2, 2010, at 7:17 PM, Mark Holloway wrote: > I want to say thank you to everyone on the OSL who has participated in any of > my discussions or helped resolve issues that I encountered. I went to San > Jose for my second attempt on Friday and received the news yesterday that I > passed. CCIE #27384. > > Thanks, > Mark > > ___ > 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] Call Forward Unregistered
Greetings, Following up on this question. I was able to achieve the desired results from HQ. I'm running VMware and I don't have VM profile associated with the SiteB phone. However, When I tried to dial from SiteC (CME going through the GK), the call always go out the PSTN on SiteC.If I take SiteB off SRST mode then the call goes through the GK. According to the scenario listed below, the call should go through the GK then leave using the HQ router. Forward (5002) For +19723033001 (3...) By+19723033001 (3...) SiteC calls set to go through GK then I have a POTS dialpeer in case GK is down. When Site B is in SRST, calls from Site C always dial international and doesn't go through the GK. If I shutdown my pots dial peer, to force the call to go through the GK, the call fails. Any ideas From: sisiaji To: Mark Holloway Cc: ccie_voice@onlinestudylist.com; Afzal Bhutta Sent: Mon, October 18, 2010 6:06:00 AM Subject: Re: [OSL | CCIE_Voice] Call Forward Unregistered hey guys, i truly have no clue what you are talking about :))) what VM has to do with CFUR? For and By fields are representing what you have configured as redirecting Calling Party number mask (in this case redirecting ip phone) and what you have configured as a destination for such calls (Unregistered). if both are set with +... then both will be shown as +... in For/By fields... it is not a rocket science I would say so... however, for CFUR, you have to be extremely careful, as it doesn't require separate partitions/CSS to work, but if you think about it, it is the only way to fine tune it to what you want. so don't overcomplicate it, set your Unregistered destination to be +19723033001 and assuming your calling party mask is already the same, then you just need to create RP for the same + number inside separate partition which will be the only one present in a separate CSS, which in turn will need to be assigned to Unregistered Destination CSS. nothing else. when you create RP for +..., you just need to do proper digit manipulation depending on which location gateway calls is supposed to go out. so if this is national call, then you have to put inside RG/RL manipulation pre-dot (for +1.972XX), called type National, plan isdn) and don't touch calling party xformations at all as by default they are set on callmanager which means only internal 4 digits will be sent as calling numbers (that is what you see inside brackets). ok? :) On Mon, Oct 18, 2010 at 1:51 AM, Mark Holloway wrote: I think the main thing to understand is that it should work using E164 in For/By under normal circumstances and everything else we are suggesting is a work around to a known bug with CUCM 7.0 and VMWare. > > > > >On Oct 17, 2010, at 3:56 PM, Daniel Berlinski wrote: > >Hello guys >> >>If you want to manipulate this with CUCM the place to change the redirected >>number is the VM profile as indicated by Mark. Alternatively you could >>attach >>an additional rule to the translation-profile plugged inbound to the POTS >>call >>leg in the branch router in SRST mode and configure it to change the >>redirect-called number from to the e164 that you are after. >> >>Cheers >> >> >>On Mon, Oct 18, 2010 at 11:36 AM, Mark Holloway wrote: >> >>I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and >>VMWare. >> If you go to the Device > Phone and click on the Site B phones > Line and >>specifically assign the Voicemail Profile to the Line it might work. I had >>success a couple of times doing this, but then after resetting my rack the >>last >>time and assigning the VM profile to the Line I still had this issue. >>> >>> >>>On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote: >>> >>>Scenario: In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits dialing in SRST.(Wan failure) I use call forward unregistered feature. When I call from HQ Phone-1 call routed through HQ Gateway. When I call from Site-C Phone-1 call routed through the GK first and then HQ Gateway. Below is the display I am getting on my Site-B phone display. Forward HQ Phone 1 (2001) For 3001 By3001 Forward Site-C Phone 1 (4001) For 3001 By3001 My question how can I achieve below display in FOR and BY field it should be E.164 number format and than 4 digits internal ID Forward (2001) For +19723033001 (3...) By+19723033001 (3...) Forward (4001) For +19723033001 (3...) By+19723033001 (3...) Thanking you in anticipation folks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com >>> >>>
Re: [OSL | CCIE_Voice] Calling and Called info
Great. Thanks everyone. Much appreciated. From: C-note To: C-note ; ccieid1ot ; ayman labib Cc: OSL Group Sent: Mon, October 11, 2010 8:18:07 PM Subject: Re: [OSL | CCIE_Voice] Calling and Called info I'm sorry, that pattern would be used in UCM. It appears that your configuration is correct. From: C-note To: ccieid1ot ; ayman labib Cc: OSL Group Sent: Mon, October 11, 2010 7:58:39 PM Subject: Re: [OSL | CCIE_Voice] Calling and Called info voice translation-rule 70 rule 1 /\(^3...\)/ /+1617303\1/ The + sign in +1617 needs to be preceded with \+ to treat it as the literal character '+' and not the wild card character + as such: voice translation-rule 70 rule 1 /\(^3...\)/ /\+1617303\1/ From: ccieid1ot To: ayman labib Cc: OSL Group Sent: Mon, October 11, 2010 4:44:05 PM Subject: Re: [OSL | CCIE_Voice] Calling and Called info I c nothing wrong with it. On Mon, Oct 11, 2010 at 3:42 PM, ayman labib wrote: Greetings. > >Any ideas about my question below. I need to be 100% sure this time. I >booked >for my second attempt in December, and I would like to be well prepared this >time. My 1st attempt was back in June. Any input is greatly appreciated. >Thanks > > > > ____ From: ayman labib >To: OSL Group >Sent: Mon, October 11, 2010 8:57:09 AM >Subject: Calling and Called info > > > >Hello Experts. > >Just wondering, what would be the best way to insert calling and called info >at >remote sites. Assuming that SiteB is an SRST H323. Site C is a CME site. >/ >Here's an example of what I'm using to insert the calling and called info. >Can >anyone please validate and offer suggestions? Thanks in advance > >voice translation-rule 40 > rule 1 /.*/ /\0/ type any international plan any isdn > >voice translation-rule 70 > rule 1 /\(^3...\)/ /+1617303\1/ > >voice translation-profile INT > translate calling 70 > translate called 40 >! > >dial-peer voice 11 pots > translation-profile outgoing INT > destination-pattern 9011T > port 1/0:23 > prefix 011 >! > > > > >___ >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] Calling and Called info
Greetings. Any ideas about my question below. I need to be 100% sure this time. I booked for my second attempt in December, and I would like to be well prepared this time. My 1st attempt was back in June. Any input is greatly appreciated. Thanks From: ayman labib To: OSL Group Sent: Mon, October 11, 2010 8:57:09 AM Subject: Calling and Called info Hello Experts. Just wondering, what would be the best way to insert calling and called info at remote sites. Assuming that SiteB is an SRST H323. Site C is a CME site. / Here's an example of what I'm using to insert the calling and called info. Can anyone please validate and offer suggestions? Thanks in advance voice translation-rule 40 rule 1 /.*/ /\0/ type any international plan any isdn voice translation-rule 70 rule 1 /\(^3...\)/ /+1617303\1/ voice translation-profile INT translate calling 70 translate called 40 ! dial-peer voice 11 pots translation-profile outgoing INT destination-pattern 9011T port 1/0:23 prefix 011 ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Calling and Called info
Hello Experts. Just wondering, what would be the best way to insert calling and called info at remote sites. Assuming that SiteB is an SRST H323. Site C is a CME site. / Here's an example of what I'm using to insert the calling and called info. Can anyone please validate and offer suggestions? Thanks in advance voice translation-rule 40 rule 1 /.*/ /\0/ type any international plan any isdn voice translation-rule 70 rule 1 /\(^3...\)/ /+1617303\1/ voice translation-profile INT translate calling 70 translate called 40 ! dial-peer voice 11 pots translation-profile outgoing INT destination-pattern 9011T port 1/0:23 prefix 011 ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unity Connection - Error when trying to record a customer greeting
The only time I saw this message or similar is when I use my Firefox and not IE. From: Mark Holloway To: CCIE Voice Maillist Sent: Fri, October 8, 2010 2:56:37 PM Subject: [OSL | CCIE_Voice] Unity Connection - Error when trying to record a customer greeting Has anyone ever seen this before? I login to Unity Connection then click on my BR1PH1 user so I can record a custom greeting. When I press the Record button I get the following error. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site
Thanks for the reply. As it turns out. Loopback interface is a required step. Now everything is working. Thanks Next challenge is to get Site HQ and SRST to use MoH with CME using the Gatekeeper. Thanks From: ayman labib To: amr thabt Cc: ccie_voice@onlinestudylist.com Sent: Thu, October 7, 2010 3:49:41 PM Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site Thanks for the reply. I do have the max ephone etc.. I removed my config to keep it short. I tried it with bind command and without. Same Issue. I don't have Lo0 configured. Everything is configured using the fa0/1 interface. Please have a look at the screen shots of my config. I really appreciate everyone's help. 2 days and it's driving me crazy. call-manager-fallback secondary-dialtone 9 max-conferences 8 gain -6 transfer-system full-consult ip source-address 192.168.31.10 port 2000 strict-match max-ephones 10 max-dn 10 transfer-pattern .T voicemail 912123945020 call-forward pattern .T call-forward busy 12123945020 call-forward noan 12123945020 timeout 20 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 192.168.31.10 time-zone 8 ! From: amr thabt To: ayman labib Cc: ccie_voice@onlinestudylist.com Sent: Thu, October 7, 2010 3:07:59 PM Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site Hi Ayman, I have three comments that may help 1 Do you add max-dn and max-ephone under call-manager-fallback 2-in "ccm-manager music-on-hold bind fa0/1 " remove the bind use only ccm-manager music-on-hold 3- in multicast command add both loopback and VLan SVI ip address. HTH AMR On Thu, Oct 7, 2010 at 9:56 PM, ayman labib wrote: Just wondering if anyone encountered this problem. > >I still can't get MOH when calling the PSTN phone and the site is not in SRST >mode. According to the sh command below. The call manager has done its job >but the GWY is not responding. Any ideas? MOH local and between HQ works >fine. Just need a sanity check. Thanks for all your help > >SRST-Site#sh ccm-manager music-on-hold >Current active multicast sessions : 1 > Multicast RTP port Packets Call CodecIncoming > Address number in/outidInterface >=== >239.1.1.1 16384 0/0 12 g711ulaw > >ccm-manager music-on-hold bind fa0/1 > >call-manager-fallback > ip source-address 192.168.31.10 port 2000 strict-match > moh music-on-hold.au > multicast moh 239.1.1.1 port 16384 route 192.168.31.10 > > >http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1046789 > > > > > From: ayman labib >To: ccie_voice@onlinestudylist.com >Cc: ccie_voice@onlinestudylist.com >Sent: Wed, October 6, 2010 9:45:12 AM >Subject: MoH to PSTN from SRST site > > > > > >Hello Experts, > >Follow up to Mark's email about Moh to PSTN. I don't hear the Piano music as >well. Inter-site and Intra-site with HQ works. > > >I see the Muticast on the gateway is invoked and on the server, but don't hear >anything. Any idea? Thanks in advance > >admin:show perf query class "Cisco MOH Device" >==>query class : > > - Perf class (Cisco MOH Device) has instances and values: >MOH_2 -> MOHHighestActiveResources = 1 >MOH_2 -> MOHMulticastResourceActive = 0 >MOH_2 -> MOHMulticastResourceAvailable = 25 >MOH_2 -> MOHOutOfResources = 0 >MOH_2 -> MOHTotalMulticastResources = 25 >MOH_2 -> MOHTotalUnicastResources = 250 >MOH_2 -> MOHUnicastResourceActive = 0 >MOH_2 -> MOHUnicastResourceAvailable= 250 >MOH_3 -> MOHHighestActiveResources = 1 >MOH_3 -> MOHMulticastResourceActive = 1 >MOH_3 -> MOHMulticastResourceAvailable = 24 >MOH_3 -> MOHOutOfResources = 0 >MOH_3 -> MOHTotalMulticastResources = 25 >MOH_3 -> MOHTotalUnicastResources = 250 >MOH_3 -> MOHUnicastResourceActive = 0 >MOH_3 -> MOHUnicastResourceAvailable= 250 > > > > > > >___ >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 to PSTN from SRST site
Just wondering if anyone encountered this problem. I still can't get MOH when calling the PSTN phone and the site is not in SRST mode. According to the sh command below. The call manager has done its job but the GWY is not responding. Any ideas? MOH local and between HQ works fine. Just need a sanity check. Thanks for all your help SRST-Site#sh ccm-manager music-on-hold Current active multicast sessions : 1 Multicast RTP port Packets Call CodecIncoming Address number in/outidInterface === 239.1.1.1 16384 0/0 12 g711ulaw ccm-manager music-on-hold bind fa0/1 call-manager-fallback ip source-address 192.168.31.10 port 2000 strict-match moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 192.168.31.10 http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1046789 From: ayman labib To: ccie_voice@onlinestudylist.com Cc: ccie_voice@onlinestudylist.com Sent: Wed, October 6, 2010 9:45:12 AM Subject: MoH to PSTN from SRST site Hello Experts, Follow up to Mark's email about Moh to PSTN. I don't hear the Piano music as well. Inter-site and Intra-site with HQ works. I see the Muticast on the gateway is invoked and on the server, but don't hear anything. Any idea? Thanks in advance admin:show perf query class "Cisco MOH Device" ==>query class : - Perf class (Cisco MOH Device) has instances and values: MOH_2 -> MOHHighestActiveResources = 1 MOH_2 -> MOHMulticastResourceActive = 0 MOH_2 -> MOHMulticastResourceAvailable = 25 MOH_2 -> MOHOutOfResources = 0 MOH_2 -> MOHTotalMulticastResources = 25 MOH_2 -> MOHTotalUnicastResources = 250 MOH_2 -> MOHUnicastResourceActive = 0 MOH_2 -> MOHUnicastResourceAvailable= 250 MOH_3 -> MOHHighestActiveResources = 1 MOH_3 -> MOHMulticastResourceActive = 1 MOH_3 -> MOHMulticastResourceAvailable = 24 MOH_3 -> MOHOutOfResources = 0 MOH_3 -> MOHTotalMulticastResources = 25 MOH_3 -> MOHTotalUnicastResources = 250 MOH_3 -> MOHUnicastResourceActive = 0 MOH_3 -> MOHUnicastResourceAvailable= 250 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX challenges
Check your CSQ field if it's spelled properly and have quotations. That's what got me the other time. From: "cciefo...@hotmail.com" To: Pithog Oil ; ccie_voice-boun...@onlinestudylist.com; ccie_voice@onlinestudylist.com Sent: Wed, October 6, 2010 5:27:27 PM Subject: Re: [OSL | CCIE_Voice] UCCX challenges There is something wrong with the script. It could be a misspelled variable; or if you are referencing a holiday script that coukd be wrong too. Try doing. Reactive debug to see where the problem is in the script. Sent from my Verizon Wireless BlackBerry -Original Message- From: Pithog Oil Sender: ccie_voice-boun...@onlinestudylist.com Date: Wed, 6 Oct 2010 14:12:27 To: Subject: [OSL | CCIE_Voice] UCCX challenges ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MoH to PSTN from SRST site
Hello Experts, Follow up to Mark's email about Moh to PSTN. I don't hear the Piano music as well. Inter-site and Intra-site with HQ works. I see the Muticast on the gateway is invoked and on the server, but don't hear anything. Any idea? Thanks in advance admin:show perf query class "Cisco MOH Device" ==>query class : - Perf class (Cisco MOH Device) has instances and values: MOH_2 -> MOHHighestActiveResources = 1 MOH_2 -> MOHMulticastResourceActive = 0 MOH_2 -> MOHMulticastResourceAvailable = 25 MOH_2 -> MOHOutOfResources = 0 MOH_2 -> MOHTotalMulticastResources = 25 MOH_2 -> MOHTotalUnicastResources = 250 MOH_2 -> MOHUnicastResourceActive = 0 MOH_2 -> MOHUnicastResourceAvailable= 250 MOH_3 -> MOHHighestActiveResources = 1 MOH_3 -> MOHMulticastResourceActive = 1 MOH_3 -> MOHMulticastResourceAvailable = 24 MOH_3 -> MOHOutOfResources = 0 MOH_3 -> MOHTotalMulticastResources = 25 MOH_3 -> MOHTotalUnicastResources = 250 MOH_3 -> MOHUnicastResourceActive = 0 MOH_3 -> MOHUnicastResourceAvailable= 250 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] +Dialing to Site C without Gatekeeper
Hello Experts, I'm working on TEHO to Site C with Gatekeeper. Everything works except for the + sign. On the debug ISDN Q931 on both sites, I see the + being sent and received, but on the phone it only shows my 11 digits (12123945003), but on the bottom of the phone it shows my plus sign. Any ideas??? Thanks in advance Cheers ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SRST to UC VoiceMail
Thank you all, got it working. On my outgoing dial-peer I add Translation profile byu accident to translate my number from 4 digits to 10 digits. Would that be an acceptable solution, Real-life scenario and LAB? Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA9838C Exclusive, Channel 12 Progress Ind i = 0x8183 - Origination address is non-ISDN Display i = '+16178631002' Calling Party Number i = 0x0080, '1002' Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '12123945020' Plan:ISDN, Type:National Sep 16 15:37:11.629: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x8097 Channel ID i = 0xA9838C Exclusive, Channel 12 Sep 16 15:37:11.661: ISDN Se0/1/0:23 Q931: RX <- ALERTING pd = 8 callref = 0x8097 Progress Ind i = 0x8088 - In-band info or appropriate now available Sep 16 15:37:11.729: ISDN Se0/1/0:23 Q931: RX <- CONNECT pd = 8 callref = 0x8097 Display i = 'VoiceMail' Sep 16 15:37:11.733: %ISDN-6-CONNECT: Interface Serial0/1/0:11 is now connected to 12123945020 N/A SRST-Site(config-dial-peer)# Sep 16 15:37:11.733: ISDN Se0/1/0:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x0097 SRST-Site(config-dial-peer)# Sep 16 15:37:13.417: %ISDN-6-DISCONNECT: Interface Serial0/1/0:11 disconnected from 12123945020 , call lasted 1 seconds SRST-Site(config-dial-peer)# Sep 16 15:37:13.417: ISDN Se0/1/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x0097 Cause i = 0x8090 - Normal call clearing Sep 16 15:37:13.425: ISDN Se0/1/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x8097 Sep 16 15:37:13.429: ISDN Se0/1/0:23 Q931: TX -> RE ____ From: Paul Kruger To: ayman labib Cc: ccie_voice@onlinestudylist.com Sent: Thu, September 16, 2010 11:19:15 AM Subject: Re: [OSL | CCIE_Voice] SRST to UC VoiceMail What is the extension configured in CUC? 4 digits? If so, then it won't work. Your number you have configured here in SRST is the full 10-dig extension. You would have to add a redirecting voice translation-pattern to match 4-digits to go out and hit the correct extension in CUC. On Thu, Sep 16, 2010 at 5:10 PM, ayman labib wrote: Thanks for the reply, > >I have already done that. I reset the gateway(HQ) and still no luck. Below >is >my debug from the HQ router (CUC is located in HQ) > >6178631002 --> User phone number in SRST >2123945020 --> CUC Pilot number > >Sep 16 15:04:14.631: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8 callref = 0x00A3 >Bearer Capability i = 0x9090A2 >Standard = CCITT >Transfer Capability = 3.1kHz Audio >Transfer Mode = Circuit >Transfer Rate = 64 kbit/s >Channel ID i = 0xA98381 >Exclusive, Channel 1 >Progress Ind i = 0x8583 - Origination address is non-ISDN >Display i = '+16178631002' >Calling Party Number i = 0x2180, '6178631002' >Plan:ISDN, Type:National >Called Party Number i = 0xA1, '2123945020' >Plan:ISDN, Type:National >Sep 16 15:04:14.643: ISDN Se2/0:23 Q931: TX -> CALL_PROC pd = 8 callref = >0x80A3 >Channel ID i = 0xA98381 >Exclusive, Channel 1 >Central-GWY# >Sep 16 15:04:14.647: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x80A3 >Progress Ind i = 0x8088 - In-band info or appropriate now available >Sep 16 15:04:14.779: ISDN Se2/0:23 Q931: TX -> CONNECT pd = 8 callref = 0x80A3 >Display i = 'VoiceMail' >Sep 16 15:04:14.787: ISDN Se2/0:23 Q931: RX <- CONNECT_ACK pd = 8 callref = >0x00A3 >Central-GWY# >Sep 16 15:04:17.151: ISDN Se2/0:23 Q931: RX <- DISCONNECT pd = 8 callref = >0x00A3 >Cause i = 0x8290 - Normal call clearing >Sep 16 15:04:17.179: ISDN Se2/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x80A3 >Sep 16 15:04:17.183: ISDN Se2/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = >0x00A3 > > > > > > > From: Paul Kruger >To: ayman labib > >Cc: ccie_voice@onlinestudylist.com >Sent: Thu, September 16, 2010 11:00:25 AM >Subject: Re: [OSL | CCIE_Voice] SRST to UC VoiceMail > > >On your HQ GW (or whatever GW is used to reach the CUC) you need to check: >Redirecting Number IE Delivery - Inbound >Redirecting Number IE Delivery - Outbound > > >HTH, >Paul > > >On Thu, Sep 16, 2010 at 4:53 PM, ayman labib wrote: > >Greetings, >> >>SRST to Unity Con
Re: [OSL | CCIE_Voice] SRST to UC VoiceMail
Thanks Amy, My bad, this method always worked. I have it working right now with Alternate Extension. I remember reading in previous posts few months ago that there's another method to achieve this. From: Amy Ryan To: ayman labib ; ccie_voice@onlinestudylist.com Sent: Thu, September 16, 2010 11:09:11 AM Subject: Re: [OSL | CCIE_Voice] SRST to UC VoiceMail Re: [OSL | CCIE_Voice] SRST to UC VoiceMail You will need to add an alternate extension for the voicemail user in Unity Connection of the expanded number as it comes in from the PSTN. If you are globalizing numbers, you will need to use the full e164 number as your alternate extension. --- Amy Ryan – CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat <http://www.ipexpert.com/chat> eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (R&S, Voice, Wireless, Security & Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit 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: ayman labib Date: Thu, 16 Sep 2010 07:53:18 -0700 (PDT) To: Cc: Subject: [OSL | CCIE_Voice] SRST to UC VoiceMail Greetings, SRST to Unity Connection VM question. I put my site in SRST, Everything works as expected except when I hit the message button to go to my VM box in UC. It goes to the general UC message. Under my HQ gateway, I have Redirecting number IE inbound and outbound checked. Am I missing any step? Any ideas or suggestion is greatly appreciated My config: interface Serial0/1/0:23 no ip address isdn switch-type primary-ni isdn incoming-voice voice isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! call-manager-fallback secondary-dialtone 9 max-conferences 8 gain -6 transfer-system full-consult ip source-address 192.168.31.10 port 2000 strict-match max-ephones 10 max-dn 10 transfer-pattern .T voicemail 912123945020 call-forward pattern .T call-forward busy 912123945020 call-forward noan 912123945020 timeout 20 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 192.168.31.10 192.168.31.10 time-zone 8 ! ___ 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] SRST to UC VoiceMail
Thanks for the reply, I have already done that. I reset the gateway(HQ) and still no luck. Below is my debug from the HQ router (CUC is located in HQ) 6178631002 --> User phone number in SRST 2123945020 --> CUC Pilot number Sep 16 15:04:14.631: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8 callref = 0x00A3 Bearer Capability i = 0x9090A2 Standard = CCITT Transfer Capability = 3.1kHz Audio Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Display i = '+16178631002' Calling Party Number i = 0x2180, '6178631002' Plan:ISDN, Type:National Called Party Number i = 0xA1, '2123945020' Plan:ISDN, Type:National Sep 16 15:04:14.643: ISDN Se2/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x80A3 Channel ID i = 0xA98381 Exclusive, Channel 1 Central-GWY# Sep 16 15:04:14.647: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x80A3 Progress Ind i = 0x8088 - In-band info or appropriate now available Sep 16 15:04:14.779: ISDN Se2/0:23 Q931: TX -> CONNECT pd = 8 callref = 0x80A3 Display i = 'VoiceMail' Sep 16 15:04:14.787: ISDN Se2/0:23 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x00A3 Central-GWY# Sep 16 15:04:17.151: ISDN Se2/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x00A3 Cause i = 0x8290 - Normal call clearing Sep 16 15:04:17.179: ISDN Se2/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x80A3 Sep 16 15:04:17.183: ISDN Se2/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x00A3 From: Paul Kruger To: ayman labib Cc: ccie_voice@onlinestudylist.com Sent: Thu, September 16, 2010 11:00:25 AM Subject: Re: [OSL | CCIE_Voice] SRST to UC VoiceMail On your HQ GW (or whatever GW is used to reach the CUC) you need to check: Redirecting Number IE Delivery - Inbound Redirecting Number IE Delivery - Outbound HTH, Paul On Thu, Sep 16, 2010 at 4:53 PM, ayman labib wrote: Greetings, > >SRST to Unity Connection VM question. I put my site in SRST, Everything works >as expected except when I hit the message button to go to my VM box in UC. It >goes to the general UC message. Under my HQ gateway, I have Redirecting >number >IE inbound and outbound checked. Am I missing any step? Any ideas or >suggestion is greatly appreciated > >My config: > >interface Serial0/1/0:23 > no ip address > isdn switch-type primary-ni > isdn incoming-voice voice > isdn outgoing display-ie > isdn outgoing ie redirecting-number > no cdp enable >! >call-manager-fallback > secondary-dialtone 9 > max-conferences 8 gain -6 > transfer-system full-consult > ip source-address 192.168.31.10 port 2000 strict-match > max-ephones 10 > max-dn 10 > transfer-pattern .T > voicemail 912123945020 > call-forward pattern .T > call-forward busy 912123945020 > call-forward noan 912123945020 timeout 20 > moh music-on-hold.au > multicast moh 239.1.1.1 port 16384 route 192.168.31.10 192.168.31.10 > time-zone 8 >! > > >___ >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] SRST to UC VoiceMail
Greetings, SRST to Unity Connection VM question. I put my site in SRST, Everything works as expected except when I hit the message button to go to my VM box in UC. It goes to the general UC message. Under my HQ gateway, I have Redirecting number IE inbound and outbound checked. Am I missing any step? Any ideas or suggestion is greatly appreciated My config: interface Serial0/1/0:23 no ip address isdn switch-type primary-ni isdn incoming-voice voice isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! call-manager-fallback secondary-dialtone 9 max-conferences 8 gain -6 transfer-system full-consult ip source-address 192.168.31.10 port 2000 strict-match max-ephones 10 max-dn 10 transfer-pattern .T voicemail 912123945020 call-forward pattern .T call-forward busy 912123945020 call-forward noan 912123945020 timeout 20 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 192.168.31.10 192.168.31.10 time-zone 8 ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE #26721 - I PASSED!
Congrats, I'm new to the list. I got to admit, 2 CCIE's in one week on the first try is very inspiring. Excellent Job guys. I'm going for my lab exam in December. From: Thomas Koch To: "Wodarski, Tim" Cc: CCIE Voice OSL Sent: Wed, August 18, 2010 7:53:22 AM Subject: Re: [OSL | CCIE_Voice] CCIE #26721 - I PASSED! Outstanding! Nice job Matthew! Sent from my iPad On Aug 17, 2010, at 5:46 PM, "Wodarski, Tim" wrote: Great job!! Congrats > > >On 8/17/10 5:05 PM, "Matthew Berry" wrote: > > >I just got my score report. I passed guys. >> >>More follow-up to come later. Right now I'm now on cloud nine. :) >> >>CCIE #26271 >> >>Thanks, >> >>Matthew Berry >>ciscovoiceg...@gmail.com >>http://ciscovoiceguru.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 > ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
