Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)
Hi, You have to register the br2 with the UCME zone not the VIA zone. Remove h323-gateway voip id VIA ipaddr 172.1.254.1 1719 and replace it with h323-gateway voip id UCME ipaddr 172.1.254.1 1719 Thanks Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Goh Sent: Saturday, April 09, 2011 9:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Hi Guys, I'm trying to get the solutions for question 4.2 to work, but apparently I'm missing something and hope someone can help. I've search thru the list but doesn't really found a solution work for my case. The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 ring, but when i tried to answered, the call drop. I know it might be related to codec issue, but I've my HQ-RTR configured with Xcoder which it is up and active but the call still failing. I also did have the trunk in cucm Wait for Far End H.245 Terminal Capability Set unchecked. once things I notice is that, my call doesn't seems get re-originated on the cube router to BR2 router, what I see during ringing state my show gatekeeper endpoint show the call is directly from the CUCM to BR2 It is only 2 call legs instead of 4 (see below). hm, what have I missed? Some Info: HQ Router (R1) interface Loopback0 ip address 172.1.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R1 h323-gateway voip bind srcaddr 172.1.254.1 gatekeeper zone local UCM 172.1.254.1 zone local UCME outvia VIA zone local VIA zone prefix UCME 3... gw-type-prefix 1#* default-technology no shutdown dial-peer voice 30 voip destination-pattern 3... session target ras codec g711ulaw ! dial-peer voice 31 voip incoming called-number 3... Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 511-32797 6 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk_trunk_25001 CallSignalAddr Port RASSignalAddr Port 172.1.10.20 38233 172.1.10.20 32795 Endpt(s): Alias E.164Addr dst EP: R33003 CallSignalAddr Port RASSignalAddr Port 172.3.254.1 1720 172.3.254.1 49395 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 172.1.10.10 47142 172.1.10.10 32838 UCM VOIP-GW H323-ID: gk_trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.1.10.20 38233 172.1.10.20 32795 UCM VOIP-GW H323-ID: gk_trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.1.254.1 1720 172.1.254.2 56974 VIA H323-GW H323-ID: R1 Voice Capacity Max.= Avail.= Current.= 0 172.3.254.1 1720 172.3.254.1 49395 VIA H323-GW H323-ID: R3 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 R1(config-if)#do sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 1#*(Default gateway-technology) Zone UCM master gateway list: 172.1.10.20:38233 gk_trunk_2 172.1.10.10:47142 gk_trunk_1 Zone VIA master gateway list: 172.3.254.1:1720 R3 172.1.254.2:1720 R1 BR2 Router (R2) interface Loopback0 ip address 172.3.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R3 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.3.254.1 dial-peer voice 10 voip incoming called-number 3... dtmf-relay rtp-nte codec g711ulaw ! CUCM Trunk the trunk was assign a separate DP with a region that using G729 when calling HQ and BR2. Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part
[OSL | CCIE_Voice] vRack Vouchers for Sale!
I have few voice vrack vouchers left for sale. any1 interested , unicast me. cheerz ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] time-zone in telephony-serv
I changed time-zone to 8 and then did no create cnf / create cnf and restart all. But still the time-zone is incorrect. Both are skinny phones. Please advice. T I A Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] vRack Vouchers for Sale
Dear all, I have 45 vrack vouchers to sale.. i'm not going to use them.. so email me if interested.. Thanks Duncan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CME SRST: calling name configuration
I have ran into the same thing, see the note below from one of my colleagues, I believe it is fixed with firmware 9.1.x, but I cannot access the Cisco bug tool at the moment. I am not sure about your first issue with Calling name in SRST mode. But the phones not coming back up is what I am addressing. FYI, If you have any phones that are running SCCP v8.4.x ( was distributed with UCM 7.1.x ) they will most likely have issues registering with SRST, they either do not register at all or will register the ephone but not any DNs. Larry Stern Senior Systems Engineer Black Box Network Services Long Island Voice/Data 6000 New Horizons Blvd. Amittyville, NY, 11701 Direct: +1631.841.5225 www.blackbox.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, April 10, 2011 7:28 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 62, Issue 54 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. Re: CME SRST: calling name configuration (adam compton) 2. Changing Network Settings on SIP Firmware 7961 (Mann Chaddha) 3. Re: Gatekeeper + CUBE (WB2 Lab1 4.2) (Naoufal Kerboute) 4. vRack Vouchers for Sale! (must ccie) 5. time-zone in telephony-serv (Shrini) 6. vRack Vouchers for Sale (Duncan Hamilton-Walker) 7. UCCX Script (Redirect call based on the calling number) (Naoufal Kerboute) -- Message: 1 Date: Sat, 9 Apr 2011 23:41:40 -0400 From: adam compton com...@gmail.com To: Miron Kobelski findko...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SRST: calling name configuration Message-ID: BANLkTi=on9m68hqixawb+3v2jgrgavb...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 I've had the same problem. I never knew what to think of it. I assumed that's just how it works. On Sat, Apr 9, 2011 at 2:57 PM, Miron Kobelski findko...@gmail.com wrote: Hello, I was playing with CME SRST today and I encountered the same issue again. I configured CME SRST with srst mode auto-provision all. Phones reregistered to SRST correctly, ephone and ephone-dn configuration appeared in the config. By default, each ephone-dn is configured with CUCM external phone mask as a calling name. Is it possible to change ephone's calling name to something other in CME SRST? When I changed the name under ephone-dn and restarted the phone. It reregistered, but DN didn't appear on the button (couldn't make any calls). Is it normal/expected behaviour or I missed something? best regards kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110409/a4678fac/attachment-0001.html -- Message: 2 Date: Sun, 10 Apr 2011 10:43:46 +0530 From: Mann Chaddha mann.chad...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Changing Network Settings on SIP Firmware 7961 Message-ID: BANLkTin+sWfthwoDrun2w0d8Uh=dywj...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi All I am running Lab 9 am facing this issue. My home phone 7961 has a SIP firmware on it is trying to register to the Br1-CME (Voice Register Global). I need to edit the TFTP Server settings on the phone but itas prompting for a username password. What are the defaults for these? Thanks Mann -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110410/d926b5bd/attachment-0001.html -- Message: 3 Date: Sat, 9 Apr 2011 18:19:39 + From: Naoufal Kerboute naou...@mhdinfotech.com To: Alex Goh ncsalex@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Message-ID: a1e0b7fadebf714f9a15622f35b1234f53869...@mhditmbx.mhdinfotech.com Content-Type: text/plain; charset=us-ascii Hi, You have to register the br2 with the UCME zone not the VIA zone. Remove h323-gateway voip id VIA ipaddr 172.1.254.1 1719 and replace it with h323-gateway voip id UCME ipaddr 172.1.254.1 1719 Thanks Naoufal -Original Message- From: ccie_voice
Re: [OSL | CCIE_Voice] CME SRST: calling name configuration
Register Global). I need to edit the TFTP Server settings on the phone but itas prompting for a username password. What are the defaults for these? Thanks Mann -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110410/d926b5bd/attachment-0001.html -- Message: 3 Date: Sat, 9 Apr 2011 18:19:39 + From: Naoufal Kerboute naou...@mhdinfotech.com To: Alex Goh ncsalex@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Message-ID: a1e0b7fadebf714f9a15622f35b1234f53869...@mhditmbx.mhdinfotech.com Content-Type: text/plain; charset=us-ascii Hi, You have to register the br2 with the UCME zone not the VIA zone. Remove h323-gateway voip id VIA ipaddr 172.1.254.1 1719 and replace it with h323-gateway voip id UCME ipaddr 172.1.254.1 1719 Thanks Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Goh Sent: Saturday, April 09, 2011 9:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Hi Guys, I'm trying to get the solutions for question 4.2 to work, but apparently I'm missing something and hope someone can help. I've search thru the list but doesn't really found a solution work for my case. The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 ring, but when i tried to answered, the call drop. I know it might be related to codec issue, but I've my HQ-RTR configured with Xcoder which it is up and active but the call still failing. I also did have the trunk in cucm Wait for Far End H.245 Terminal Capability Set unchecked. once things I notice is that, my call doesn't seems get re-originated on the cube router to BR2 router, what I see during ringing state my show gatekeeper endpoint show the call is directly from the CUCM to BR2 It is only 2 call legs instead of 4 (see below). hm, what have I missed? Some Info: HQ Router (R1) interface Loopback0 ip address 172.1.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R1 h323-gateway voip bind srcaddr 172.1.254.1 gatekeeper zone local UCM 172.1.254.1 zone local UCME outvia VIA zone local VIA zone prefix UCME 3... gw-type-prefix 1#* default-technology no shutdown dial-peer voice 30 voip destination-pattern 3... session target ras codec g711ulaw ! dial-peer voice 31 voip incoming called-number 3... Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 511-32797 6 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk_trunk_25001 CallSignalAddr Port RASSignalAddr Port 172.1.10.20 38233 172.1.10.20 32795 Endpt(s): Alias E.164Addr dst EP: R33003 CallSignalAddr Port RASSignalAddr Port 172.3.254.1 1720 172.3.254.1 49395 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 172.1.10.10 47142 172.1.10.10 32838 UCM VOIP-GW H323-ID: gk_trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.1.10.20 38233 172.1.10.20 32795 UCM VOIP-GW H323-ID: gk_trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.1.254.1 1720 172.1.254.2 56974 VIA H323-GW H323-ID: R1 Voice Capacity Max.= Avail.= Current.= 0 172.3.254.1 1720 172.3.254.1 49395 VIA H323-GW H323-ID: R3 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 R1(config-if)#do sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 1#*(Default gateway-technology) Zone UCM master gateway list: 172.1.10.20:38233 gk_trunk_2 172.1.10.10:47142 gk_trunk_1 Zone VIA master gateway list: 172.3.254.1:1720 R3 172.1.254.2:1720 R1 BR2 Router (R2) interface Loopback0 ip address 172.3.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R3 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.3.254.1 dial-peer voice 10 voip incoming called-number 3... dtmf-relay rtp-nte codec g711ulaw ! CUCM Trunk the trunk was assign a separate DP with a region that using G729 when calling HQ and BR2. Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please
Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number)
I believe uccx does not understand the + symbol. - Reply message - From: Naoufal Kerboute naou...@mhdinfotech.com Date: Sun, Apr 10, 2011 7:21 am Subject: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Dear gents, I'm working on UCCX section and I'm trying to reroute some calls coming from Spain (+34) to a specific extension. I've setup the script and it's working only if I set the calling number variable to full Spain PSTN number, but let take the case for many number from Spain. How can I reroute calls coming from spain to a specific extension (I don't want to much the full muber, I want to much only calling number start with +34) Any ideas? Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number)
The + symbol is a string so it can be match. My script is working if I set the condition like If Calling Number == “+3434141891” then redirect call to 5001 But I’m looking for a way to reroute all calls coming from area +34 to 5001 Naoufal From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] Sent: Sunday, April 10, 2011 4:40 PM To: Naoufal Kerboute; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) I believe uccx does not understand the + symbol. - Reply message - From: Naoufal Kerboute naou...@mhdinfotech.com Date: Sun, Apr 10, 2011 7:21 am Subject: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Dear gents, I'm working on UCCX section and I'm trying to reroute some calls coming from Spain (+34) to a specific extension. I've setup the script and it's working only if I set the calling number variable to full Spain PSTN number, but let take the case for many number from Spain. How can I reroute calls coming from spain to a specific extension (I don't want to much the full muber, I want to much only calling number start with +34) Any ideas? Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)
Hi All, Thanks very much for the reply. The issue is due to my mistake that registering BR2 to wrong zone. Now the CUCM Call to BR2 is working fine except the supplementary service e.g hold, Moh doesn't work, do I need MTP for this? also, calling from BR2 Sip phone to CUCM is failling, phone ring, but when answered, it dropped. my Sip phone is using G729 codec, do I still need MTP on BR2 in this case? Thanks Regards, Alex On Sun, Apr 10, 2011 at 2:19 AM, Naoufal Kerboute naou...@mhdinfotech.com wrote: Hi, You have to register the br2 with the UCME zone not the VIA zone. Remove h323-gateway voip id VIA ipaddr 172.1.254.1 1719 and replace it with h323-gateway voip id UCME ipaddr 172.1.254.1 1719 Thanks Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Goh Sent: Saturday, April 09, 2011 9:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Hi Guys, I'm trying to get the solutions for question 4.2 to work, but apparently I'm missing something and hope someone can help. I've search thru the list but doesn't really found a solution work for my case. The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 ring, but when i tried to answered, the call drop. I know it might be related to codec issue, but I've my HQ-RTR configured with Xcoder which it is up and active but the call still failing. I also did have the trunk in cucm Wait for Far End H.245 Terminal Capability Set unchecked. once things I notice is that, my call doesn't seems get re-originated on the cube router to BR2 router, what I see during ringing state my show gatekeeper endpoint show the call is directly from the CUCM to BR2 It is only 2 call legs instead of 4 (see below). hm, what have I missed? Some Info: HQ Router (R1) interface Loopback0 ip address 172.1.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R1 h323-gateway voip bind srcaddr 172.1.254.1 gatekeeper zone local UCM 172.1.254.1 zone local UCME outvia VIA zone local VIA zone prefix UCME 3... gw-type-prefix 1#* default-technology no shutdown dial-peer voice 30 voip destination-pattern 3... session target ras codec g711ulaw ! dial-peer voice 31 voip incoming called-number 3... Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallID Age(secs) BW 511-32797 6 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk_trunk_2 5001 CallSignalAddr Port RASSignalAddr Port 172.1.10.20 38233 172.1.10.20 32795 Endpt(s): Alias E.164Addr dst EP: R3 3003 CallSignalAddr Port RASSignalAddr Port 172.3.254.1 1720 172.3.254.1 49395 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 172.1.10.10 47142 172.1.10.10 32838 UCM VOIP-GW H323-ID: gk_trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.1.10.20 38233 172.1.10.20 32795 UCM VOIP-GW H323-ID: gk_trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.1.254.1 1720 172.1.254.2 56974 VIA H323-GW H323-ID: R1 Voice Capacity Max.= Avail.= Current.= 0 172.3.254.1 1720 172.3.254.1 49395 VIA H323-GW H323-ID: R3 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 R1(config-if)#do sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 1#* (Default gateway-technology) Zone UCM master gateway list: 172.1.10.20:38233 gk_trunk_2 172.1.10.10:47142 gk_trunk_1 Zone VIA master gateway list: 172.3.254.1:1720 R3 172.1.254.2:1720 R1 BR2 Router (R2) interface Loopback0 ip address 172.3.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R3 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.3.254.1 dial-peer voice 10 voip incoming called-number 3... dtmf-relay rtp-nte codec g711ulaw ! CUCM Trunk the trunk was assign a separate DP with a region that using G729 when calling HQ and BR2. Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)
Hi, For supplementary you have to setup an MTP, For the call drop try to enable Inbound Fast Start Naoufal -Original Message- From: Alex Goh [mailto:ncsalex@gmail.com] Sent: Sunday, April 10, 2011 6:29 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Hi All, Thanks very much for the reply. The issue is due to my mistake that registering BR2 to wrong zone. Now the CUCM Call to BR2 is working fine except the supplementary service e.g hold, Moh doesn't work, do I need MTP for this? also, calling from BR2 Sip phone to CUCM is failling, phone ring, but when answered, it dropped. my Sip phone is using G729 codec, do I still need MTP on BR2 in this case? Thanks Regards, Alex On Sun, Apr 10, 2011 at 2:19 AM, Naoufal Kerboute naou...@mhdinfotech.com wrote: Hi, You have to register the br2 with the UCME zone not the VIA zone. Remove h323-gateway voip id VIA ipaddr 172.1.254.1 1719 and replace it with h323-gateway voip id UCME ipaddr 172.1.254.1 1719 Thanks Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Goh Sent: Saturday, April 09, 2011 9:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Hi Guys, I'm trying to get the solutions for question 4.2 to work, but apparently I'm missing something and hope someone can help. I've search thru the list but doesn't really found a solution work for my case. The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 ring, but when i tried to answered, the call drop. I know it might be related to codec issue, but I've my HQ-RTR configured with Xcoder which it is up and active but the call still failing. I also did have the trunk in cucm Wait for Far End H.245 Terminal Capability Set unchecked. once things I notice is that, my call doesn't seems get re-originated on the cube router to BR2 router, what I see during ringing state my show gatekeeper endpoint show the call is directly from the CUCM to BR2 It is only 2 call legs instead of 4 (see below). hm, what have I missed? Some Info: HQ Router (R1) interface Loopback0 ip address 172.1.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R1 h323-gateway voip bind srcaddr 172.1.254.1 gatekeeper zone local UCM 172.1.254.1 zone local UCME outvia VIA zone local VIA zone prefix UCME 3... gw-type-prefix 1#* default-technology no shutdown dial-peer voice 30 voip destination-pattern 3... session target ras codec g711ulaw ! dial-peer voice 31 voip incoming called-number 3... Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallID Age(secs) BW 511-32797 6 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk_trunk_2 5001 CallSignalAddr Port RASSignalAddr Port 172.1.10.20 38233 172.1.10.20 32795 Endpt(s): Alias E.164Addr dst EP: R3 3003 CallSignalAddr Port RASSignalAddr Port 172.3.254.1 1720 172.3.254.1 49395 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 172.1.10.10 47142 172.1.10.10 32838 UCM VOIP-GW H323-ID: gk_trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.1.10.20 38233 172.1.10.20 32795 UCM VOIP-GW H323-ID: gk_trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.1.254.1 1720 172.1.254.2 56974 VIA H323-GW H323-ID: R1 Voice Capacity Max.= Avail.= Current.= 0 172.3.254.1 1720 172.3.254.1 49395 VIA H323-GW H323-ID: R3 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 R1(config-if)#do sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 1#* (Default gateway-technology) Zone UCM master gateway list: 172.1.10.20:38233 gk_trunk_2 172.1.10.10:47142 gk_trunk_1 Zone VIA master gateway list: 172.3.254.1:1720 R3 172.1.254.2:1720 R1 BR2 Router (R2) interface Loopback0 ip address 172.3.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R3 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.3.254.1 dial-peer voice 10 voip incoming called-number 3... dtmf-relay rtp-nte codec g711ulaw ! CUCM Trunk the trunk
Re: [OSL | CCIE_Voice] CME SRST: calling name configuration
: /archives/ccie_voice/attachments/20110409/a4678fac/attachment-0001.html -- Message: 2 Date: Sun, 10 Apr 2011 10:43:46 +0530 From: Mann Chaddha mann.chad...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Changing Network Settings on SIP Firmware 7961 Message-ID: BANLkTin+sWfthwoDrun2w0d8Uh=dywj...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi All I am running Lab 9 am facing this issue. My home phone 7961 has a SIP firmware on it is trying to register to the Br1-CME (Voice Register Global). I need to edit the TFTP Server settings on the phone but itas prompting for a username password. What are the defaults for these? Thanks Mann -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110410/d926b5bd/attachment-0001.html -- Message: 3 Date: Sat, 9 Apr 2011 18:19:39 + From: Naoufal Kerboute naou...@mhdinfotech.com To: Alex Goh ncsalex@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Message-ID: a1e0b7fadebf714f9a15622f35b1234f53869...@mhditmbx.mhdinfotech.com Content-Type: text/plain; charset=us-ascii Hi, You have to register the br2 with the UCME zone not the VIA zone. Remove h323-gateway voip id VIA ipaddr 172.1.254.1 1719 and replace it with h323-gateway voip id UCME ipaddr 172.1.254.1 1719 Thanks Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Goh Sent: Saturday, April 09, 2011 9:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Hi Guys, I'm trying to get the solutions for question 4.2 to work, but apparently I'm missing something and hope someone can help. I've search thru the list but doesn't really found a solution work for my case. The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 ring, but when i tried to answered, the call drop. I know it might be related to codec issue, but I've my HQ-RTR configured with Xcoder which it is up and active but the call still failing. I also did have the trunk in cucm Wait for Far End H.245 Terminal Capability Set unchecked. once things I notice is that, my call doesn't seems get re-originated on the cube router to BR2 router, what I see during ringing state my show gatekeeper endpoint show the call is directly from the CUCM to BR2 It is only 2 call legs instead of 4 (see below). hm, what have I missed? Some Info: HQ Router (R1) interface Loopback0 ip address 172.1.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R1 h323-gateway voip bind srcaddr 172.1.254.1 gatekeeper zone local UCM 172.1.254.1 zone local UCME outvia VIA zone local VIA zone prefix UCME 3... gw-type-prefix 1#* default-technology no shutdown dial-peer voice 30 voip destination-pattern 3... session target ras codec g711ulaw ! dial-peer voice 31 voip incoming called-number 3... Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 511-32797 6 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk_trunk_25001 CallSignalAddr Port RASSignalAddr Port 172.1.10.20 38233 172.1.10.20 32795 Endpt(s): Alias E.164Addr dst EP: R33003 CallSignalAddr Port RASSignalAddr Port 172.3.254.1 1720 172.3.254.1 49395 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 172.1.10.10 47142 172.1.10.10 32838 UCM VOIP-GW H323-ID: gk_trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.1.10.20 38233 172.1.10.20 32795 UCM VOIP-GW H323-ID: gk_trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.1.254.1 1720 172.1.254.2 56974 VIA H323-GW H323-ID: R1 Voice Capacity Max.= Avail.= Current.= 0 172.3.254.1 1720 172.3.254.1 49395 VIA H323-GW H323-ID: R3 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 R1(config-if)#do sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 1#*(Default gateway-technology) Zone UCM master gateway list: 172.1.10.20:38233 gk_trunk_2 172.1.10.10:47142 gk_trunk_1 Zone VIA master gateway list: 172.3.254.1:1720 R3 172.1.254.2:1720 R1 BR2 Router (R2) interface Loopback0 ip
Re: [OSL | CCIE_Voice] CME SRST: calling name configuration
CME 7 Admin Guide Talks abiut such restriction and mentions prebuilding configuration to provide service similar to that during normal operation, on page 1234 On 10 April 2011 15:28, Stern, Larry larry.st...@nuvt.com wrote: I have ran into the same thing, see the note below from one of my colleagues, I believe it is fixed with firmware 9.1.x, but I cannot access the Cisco bug tool at the moment. I am not sure about your first issue with Calling name in SRST mode. But the phones not coming back up is what I am addressing. FYI, If you have any phones that are running SCCP v8.4.x ( was distributed with UCM 7.1.x ) they will most likely have issues registering with SRST, they either do not register at all or will register the ephone but not any DNs. Larry Stern Senior Systems Engineer Black Box Network Services Long Island Voice/Data 6000 New Horizons Blvd. Amittyville, NY, 11701 Direct: +1631.841.5225 www.blackbox.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, April 10, 2011 7:28 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 62, Issue 54 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. Re: CME SRST: calling name configuration (adam compton) 2. Changing Network Settings on SIP Firmware 7961 (Mann Chaddha) 3. Re: Gatekeeper + CUBE (WB2 Lab1 4.2) (Naoufal Kerboute) 4. vRack Vouchers for Sale! (must ccie) 5. time-zone in telephony-serv (Shrini) 6. vRack Vouchers for Sale (Duncan Hamilton-Walker) 7. UCCX Script (Redirect call based on the calling number) (Naoufal Kerboute) -- Message: 1 Date: Sat, 9 Apr 2011 23:41:40 -0400 From: adam compton com...@gmail.com To: Miron Kobelski findko...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SRST: calling name configuration Message-ID: BANLkTi=on9m68hqixawb+3v2jgrgavb...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 I've had the same problem. I never knew what to think of it. I assumed that's just how it works. On Sat, Apr 9, 2011 at 2:57 PM, Miron Kobelski findko...@gmail.com wrote: Hello, I was playing with CME SRST today and I encountered the same issue again. I configured CME SRST with srst mode auto-provision all. Phones reregistered to SRST correctly, ephone and ephone-dn configuration appeared in the config. By default, each ephone-dn is configured with CUCM external phone mask as a calling name. Is it possible to change ephone's calling name to something other in CME SRST? When I changed the name under ephone-dn and restarted the phone. It reregistered, but DN didn't appear on the button (couldn't make any calls). Is it normal/expected behaviour or I missed something? best regards kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110409/a4678fac/attachment-0001.html -- Message: 2 Date: Sun, 10 Apr 2011 10:43:46 +0530 From: Mann Chaddha mann.chad...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Changing Network Settings on SIP Firmware 7961 Message-ID: BANLkTin+sWfthwoDrun2w0d8Uh=dywj...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi All I am running Lab 9 am facing this issue. My home phone 7961 has a SIP firmware on it is trying to register to the Br1-CME (Voice Register Global). I need to edit the TFTP Server settings on the phone but itas prompting for a username password. What are the defaults for these? Thanks Mann -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110410/d926b5bd/attachment-0001.html -- Message: 3 Date: Sat, 9 Apr 2011 18:19:39 + From: Naoufal Kerboute naou...@mhdinfotech.com To: Alex Goh ncsalex@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Message-ID: a1e0b7fadebf714f9a15622f35b1234f53869
Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number)
Looks similar to requirement of IPExpert Workbook 1 Lab 12A - 12.2 On 10 April 2011 15:40, Naoufal Kerboute naou...@mhdinfotech.com wrote: The + symbol is a string so it can be match. My script is working if I set the condition like If Calling Number == “+3434141891” then redirect call to 5001 But I’m looking for a way to reroute all calls coming from area +34 to 5001 Naoufal *From:* bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] *Sent:* Sunday, April 10, 2011 4:40 PM *To:* Naoufal Kerboute; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) I believe uccx does not understand the + symbol. - Reply message - From: Naoufal Kerboute naou...@mhdinfotech.com Date: Sun, Apr 10, 2011 7:21 am Subject: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Dear gents, I'm working on UCCX section and I'm trying to reroute some calls coming from Spain (+34) to a specific extension. I've setup the script and it's working only if I set the calling number variable to full Spain PSTN number, but let take the case for many number from Spain. How can I reroute calls coming from spain to a specific extension (I don't want to much the full muber, I want to much only calling number start with +34) Any ideas? Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] UCCX Script (Redirect call based on the calling number)
Thanks Roger. I was looking for the function StartsWith(+34). You are a great man :D From: Rogers Ochieng [mailto:rogersochi...@gmail.com] Sent: Sunday, April 10, 2011 7:57 PM To: Naoufal Kerboute Cc: bkvalent...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) Looks similar to requirement of IPExpert Workbook 1 Lab 12A - 12.2 On 10 April 2011 15:40, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: The + symbol is a string so it can be match. My script is working if I set the condition like If Calling Number == +3434141891 then redirect call to 5001 But I'm looking for a way to reroute all calls coming from area +34 to 5001 Naoufal From: bkvalent...@gmail.commailto:bkvalent...@gmail.com [mailto:bkvalent...@gmail.commailto:bkvalent...@gmail.com] Sent: Sunday, April 10, 2011 4:40 PM To: Naoufal Kerboute; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) I believe uccx does not understand the + symbol. - Reply message - From: Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Date: Sun, Apr 10, 2011 7:21 am Subject: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Dear gents, I'm working on UCCX section and I'm trying to reroute some calls coming from Spain (+34) to a specific extension. I've setup the script and it's working only if I set the calling number variable to full Spain PSTN number, but let take the case for many number from Spain. How can I reroute calls coming from spain to a specific extension (I don't want to much the full muber, I want to much only calling number start with +34) Any ideas? Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have
Re: [OSL | CCIE_Voice] CME SRST: calling name configuration
CME 7 Admin Guide Talks about such restriction and mentions prebuilding configuration to provide service similar to that during normal operation, on page 1234. If i want names as in CUCM then I use mode autoprovision none and prebuild my ephone dn's with the names as needed On 9 April 2011 21:57, Miron Kobelski findko...@gmail.com wrote: Hello, I was playing with CME SRST today and I encountered the same issue again. I configured CME SRST with srst mode auto-provision all. Phones reregistered to SRST correctly, ephone and ephone-dn configuration appeared in the config. By default, each ephone-dn is configured with CUCM external phone mask as a calling name. Is it possible to change ephone's calling name to something other in CME SRST? When I changed the name under ephone-dn and restarted the phone. It reregistered, but DN didn't appear on the button (couldn't make any calls). Is it normal/expected behaviour or I missed something? best regards kobel ___ 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] Study Partner
HELLO EVERYONE, I am interested to work on labs solutions if anyone want to be a partner or share his lab information please let me know so that we can work out togther :) I have attempted 2 times and got lab 2 with DND and then lab 4 with CUBE so.. i have little information tricks and i am sure we can have a good team work to pass :) Please PM me if anyone is really interested in the same Appreciated.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUCM Calling CME -- CUE
Call Flow ### HQ- Phone CUCM -- SIP CME Br3-Phone-1 CFA - CUE Now in this call flow, the problem is that, CUE has no idea that the Call was forwarded. So it plays the Login Prompt. Am I missing something here? Any leads? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM Calling CME -- CUE
You're not passing the redirecting number somewhere along the path? Something that would keep the HQ CID intact? On Sun, Apr 10, 2011 at 7:10 PM, Divin Mathew John divinj...@gmail.comwrote: Call Flow ### HQ- Phone CUCM -- SIP CME Br3-Phone-1 CFA - CUE Now in this call flow, the problem is that, CUE has no idea that the Call was forwarded. So it plays the Login Prompt. Am I missing something here? Any leads? ___ 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] CUCM Calling CME -- CUE
Enable sip diversion header. -- On Apr 10, 2011, at 17:10, Divin Mathew John divinj...@gmail.com wrote: Call Flow ### HQ- Phone CUCM -- SIP CME Br3-Phone-1 CFA - CUE Now in this call flow, the problem is that, CUE has no idea that the Call was forwarded. So it plays the Login Prompt. Am I missing something here? Any leads? ___ 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] Vol2 Lab 5 Question 4.2 CUE SRST
Hi all, I'm following up the same question that i posted last month, regarding the PSTN call diversion to BR2 CUE Voicemail during SRST.I was got confuse by the question,but looking at it to question PGuide, i think my understanding is correct. The question is asking, Ensure that the caller from PSTN who have dialed BR2 phone can be routed to Voicemail when there is a WAN outage at the BR2 site. In PGuide, the BR2 DN has CFUR external checked.So my understanding is that,when PSTN caller call BR2 phone,it should be routed to voicemail immediately,which CUE is locally located. My question is,how do achieve that PSTN caller enter Voicemail immediately without ringing the BR2 phone but allowing call from HQ/BR1 ring the destination and enter voicemail when CFNA or CFB. Anyone complete this section successfully?May share the idea? Shingei On Tue, Mar 8, 2011 at 11:16 PM, Rogers Ochieng rogersochi...@gmail.comwrote: Looking at that question the wording there does not specify that you need to send calls immediately to voicemail but that PSTN calls to BR2 can be routed to VM, it doesn't say at what state so to send VM so PSTN calls to BR2 busy and no answer states should meet the requirement. On 8 March 2011 18:05, ShinGei Yong shingei.y...@gmail.com wrote: Hi Rogers, Yes, and again, as stated below, i'm able to achieved CFB and CFNA during SRST, so in other words,the required dial-peer and setting to route call to CUE is already done,right? And HQ/BR1 phone is able to call BR2 phone via PSTN during SRST. Maybe put the question in this way: PSTN caller which originate from HQ/BR1, ring the destination during SRST.(I've done this) PSTN caller which originate from PSTN, to Voicemail without ringing BR2 phones during SRST Is the second requirement possible? Shingei. On Tue, Mar 8, 2011 at 10:26 PM, Rogers Ochieng rogersochi...@gmail.comwrote: AS you've stated you are using CUE which in normal operations you've integrated using jtapi CUE integration, i assuem the CUE module is on the BR2 router. So for SRST create a voip dial-peer using sip protocol and codec g711ulaw, dtmf sip-notify, to route calls to CUE and set CFB and CFNA, setup CUE sip settings. For HQ and BR1 to access BR2 setup CFUR On 8 March 2011 16:05, ShinGei Yong shingei.y...@gmail.com wrote: Hi Roger, As stated below, i'm able to achieved that PSTN caller routed to VM when CFB and CFNA. Alsothere's no CUC in this lab. how to achieve that PSTN caller will be route to VM while allowing HQ or BR1 ring the destination in SRST site? TIA Shingei 2011/3/8 Roger Källberg roger.kallb...@cygate.se You need to setup CFB CFNA in an SRST situation, so that it sends the call over PSTN to CUC VM. Sincerely *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ -- *Från:* ShinGei Yong [shingei.y...@gmail.com] *Skickat:* den 8 mars 2011 11:00 *Till:* ccie_voice@onlinestudylist.com *Ämne:* [OSL | CCIE_Voice] Vol2 Lab 5 Question 4.2 CUE SRST Hi, The question stated,caller from PSTN CAN BE routed to VM when there's WAN outage at BR2. Internal caller from HQ or BR1 must be able to reach BR2 phone and forward to VM if no answer. To me,there are two meaning of the sentense 1. PSTN caller routed to VM immediately when there's WAN outage at BR2, 2. PSTN caller routed to VM when CFB or CFNA. What confuse me is that,how to achieve that the PSTN caller routed to VM immediately when there's a WAN outage at BR2?I'm able to achieved that PSTN caller router to VM when CFB and CFNA. In proctor guide, Forward Unregisterd Int and External been checked(VM),but how the UCM instruct PSTN call to VM?The PSTN call will hitting the BR2 GW directly due to SRST. Am i thinking of too much? Shingei ___ 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] CUCM Calling CME -- CUE
send your CME configs On 11 April 2011 02:10, Divin Mathew John divinj...@gmail.com wrote: Call Flow ### HQ- Phone CUCM -- SIP CME Br3-Phone-1 CFA - CUE Now in this call flow, the problem is that, CUE has no idea that the Call was forwarded. So it plays the Login Prompt. Am I missing something here? Any leads? ___ 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
