Re: [OSL | CCIE_Voice] UCCX/IVR Script Repository
Hello: This is the most complet guide Scripting and Development Series: Volume 1 to 3, 7.0(1) You can find it at UCCX documentation area at Cisco hth Date: Fri, 12 Mar 2010 08:43:07 +0200 From: chip...@gmail.com To: tanner.ez...@gmail.com CC: ccie_voice@onlinestudylist.com; mjbe...@krollontrack.com Subject: Re: [OSL | CCIE_Voice] UCCX/IVR Script Repository a bit out of topic,what material is best for writing the UCCX scripts? On Thu, Mar 11, 2010 at 9:06 PM, Tanner Ezell tanner.ez...@gmail.com wrote: C:\program files\wfavvid\Scripts\Templates On Thu, Mar 11, 2010 at 1:53 PM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Does anyone know of where Cisco’s UCCX/IVR sample script repository is? I can’t find it. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Tanner Ezell ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- B _ ¿Quieres saber qué PC eres? ¡Descúbrelo aquí! http://www.quepceres.com/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can’t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ 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] RSVP WITH MLPoFR
Hello: The same happend with multicast moh traffic, after activating auto qos you need to move ip pim sparse-dense mode to the virtual template interface thanks From: roger.kallb...@cygate.se To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Date: Fri, 12 Mar 2010 06:18:37 +0100 Subject: RE: [OSL | CCIE_Voice] RSVP WITH MLPoFR That’s the expected behavior. Auto qos won’t move the ip rsvp bandwith command. That’s one of the quirks with auto qos. Brgds, Roger Källberg Unified Communication Consultant Cygate AB From: Angel Perez [mailto:gorr...@hotmail.com] Sent: den 10 mars 2010 19:37 To: osl osl Subject: [OSL | CCIE_Voice] RSVP WITH MLPoFR Hello: I was configurin MLPoFR and LFI on a link between hq and br1, on the serial interface I had: interface Serial0/2/0.202 point-to-point ip rsvp bandwidth 64 Calls where progressing as configured (two g729 calls) Then after apply auto qos voip trust fr-atm new virtual templates and virtual access interfaces are created Then trying to test the policy-map just created and tuned I noticed that I could not make calls from hq to br1 (rsvp was rejecting the call) So I added the following at hq and br1: interface Virtual-Template200 ip rsvp bandwidth 64 And the problem get solved Is this the normal situation? I suppose it is but not 100% sure Thanks ¿Te gustaría tener Hotmail en tu móvil Movistar? ¡Es gratis! _ ¿Te gustaría tener Hotmail en tu móvil Movistar? ¡Es gratis! http://serviciosmoviles.es.msn.com/hotmail/movistar-particulares.aspx___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1
Hello Jason, E1's and T1's will always use a-law and u-law companding mechanism respectively, this is used to give more resolution to low voice frequencies when digitizing an analog signal (the mechanism is also used in the other end for digital to analogue conversion), each mechanism is designed exclusively to work with its voice digital standard and cannot be used conversely, In that sense, my guess is that before applying that command in your E1 port, the companding type was u-law, you can verify this using the sh voice port command (perhaps the default configuration of a-law was somehow overwritten by a cptone command in the same port configuration), and when you hardcoded the a-law companding type everything worked as expected, I also found a note in the Cisco IOS Voice Port Configuration Guide, which says that the command is used when cross-connecting in a local router, http://www.cisco.com/en/US/partner/docs/ios/voice/voiceport/configuration/guide/vp_cfg_digital_vps_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1009871 HTH, On Thu, Mar 11, 2010 at 2:37 PM, Jason Granat j...@slash128.com wrote: So I’ve got this partially figured out. It had to do with the compand-type. E1 was a-law and T1 was u-law. I set the E1 side for u-law and it sounds correct now. The final thing I am trying to figure out is how to ‘trans-compand’ (if that is the correct term) on the PSTN gateway. As it sits I had to change the compand-type between the PSTN and E1 gateway. I don’t have experience with foreign connectivity so maybe this is the way it is done in the real world but I am thinking that perhaps the E1 site may not want or be able to change their compand-type, so can it be changed at the PSTN level between a-law and u-law locations? Thanks, Jason *From:* Jason Granat *Sent:* Thursday, March 11, 2010 9:46 AM *To:* ccie_voice@onlinestudylist.com *Subject:* PSTN Call Distortion Between T1/E1 Perhaps this is something simple that I am overlooking but I have the generic setup running in my home lab with 3 gateways and one PSTN router. 2 of the gateways are T1 and one is E1. The PSTN router is also running CME with a 7960 to simulate PSTN destinations. Calls from any site to the PSTN phone are fine. Calls between T1 sites are fine. Calls between T1 and E1 sites are distorted, like the gain is way too high. I tried playing with the gain on the voice-port but no luck. I’m not finding much online or in Cisco docs. Any suggestions? Thanks, Jason -- http://slash128.com ___ 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] Lab 4 AB GK registration problem
Hi Mike,
I'm noticing from your initial debugs that the 156.26.1.70:1719 ip
address/port the one confirming the GRQ message from BR2-RTR
***
value RasMessage ::= gatekeeperConfirm :
{
requestSeqNum 126
protocolIdentifier { 0 0 8 2250 0 4 }
gatekeeperIdentifier {PL}
rasAddress ipAddress :
{
ip '*9C1A0146*'H
port *1719*
}
}
After the Angel's suggestion this should have been corrected, so would you
please send the same debugs now from both routers? plus a show gatek zone
status, also I see that you are not pinging from the br2 l0 interface but
the closest to hq l0 interface (which might be the serial interface), please
try to use ping with options to verify that loopbacks can see each other,
Thanks!,
On Thu, Mar 11, 2010 at 2:24 PM, Mike Peterson polobi...@yahoo.com wrote:
Hi Angel,
Thanks for helping me out with this GK issue. Yes indeed the GW doesn't
receive the message , that is why we are seeing GRQ and GCF.
I do have full connectivity b/w HQ/BR2/PUB/SUB .
Below are the ping's you sugested to post.
Thanks a lot in advance for your time and help.
HQ#ping 192.21.66.254-from HQ to loopback of BR2
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.21.66.254, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/5/12 ms
HQ#
BR2-RTR#ping 192.21.64.254--from BR2 to loopback of HQ
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.21.64.254, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 8/14/32 ms
BR2-RTR#
HQ#ping 192.168.0.11 from HQ to CUCM PUB
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.168.0.11, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/4/8 ms
HQ#ping 192.168.0.12 - from HQ to CUCM SUB
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.168.0.12, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 16/25/40 ms
HQ#
BR2-RTR#ping 192.168.0.11 -from BR2 CUCM PUB
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.168.0.11, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 16/27/44 ms
BR2-RTR#ping 192.168.0.12--- from BR2 to CUCM SUB
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.168.0.12, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 20/40/52 ms
BR2-RTR#
--
*From:* Angel Perez gorr...@hotmail.com**
*Sent:* Thu, March 11, 2010 1:17:17 PM
*Subject:* RE: [OSL | CCIE_Voice] Lab 4 AB GK registration problem
Just to verify, can you ping hq loo 0 192.21.64.254 from br2? And br2 loop
192.21.66.254 from hq?
It looks like br2 gw ask for registration GRQ, and then gk try to confirm
GCF but the gw can't recieve the message
hth
--
Date: Thu, 11 Mar 2010 08:54:30 -0800
Subject: Re: [OSL | CCIE_Voice] Lab 4 AB GK registration problem
Hi All,
I did tried your suggestion (to add loopback IP address :
zone local PL cisco.com 192.21.64.254 ) which does make sense but it
doesn't work.
I took a look at deb ras and I am seeing only GRQ (a message sent by
endpoint to GK ) and GCF (A reply from gatekeeper to endpoint
which indicates the transport address of the gatekeeper RAS channel) and I
am not seeing GRJ (the reject the endpoint request for registration) so
something I am missing or I am hitting a BUG!
The deb gatekeeper main 19 or deb h225 asn1 still doesn't give me a
clue of why GK is failing to register.
Once again thanks for your time and help.
Kind Regards,
Mike
Note: This is the change I made:
gatekeeper
zone local PL cisco.com 192.21.64.254
zone prefix PL 1... gw-priority 10 gk-trunk_2
zone prefix PL 1... gw-priority 9 gk-trunk_1
zone prefix PL 1... gw-priority 0 BR2-RTR
zone prefix PL 5... gw-priority 10 gk-trunk_2
zone prefix PL 5... gw-priority 9 gk-trunk_1
zone prefix PL 5... gw-priority 0 BR2-RTR
no shutdown
!
--
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___
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visit www.ipexpert.com
--
Regards,
Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
___
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www.ipexpert.com
Re: [OSL | CCIE_Voice] via gatekeeper invia key word
Hi Jeff, According to your lab results, you are describing the expected behavior, more information at: http://www.cisco.com/en/US/partner/docs/ios/voice/cubegk/configuration/guide/ve-gk-config_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1225776 Thanks!, On Thu, Mar 11, 2010 at 9:55 AM, Jeff Cotter jcot...@voxns.com wrote: I am struggling a bit with the invia concept. I think I understand the “outvia”. When I lab this up I find the following. Invia only applies to calls coming from a remote GK. In order for call to use cube I had to configure the invia key word on the actual remote zone…..not on the destination zone. Sample config of my invia GK gk zone local ucm cisco.com 1.1.1.1 gk zone local cube gk zone local cme gk zone remote gk2 lab.com 2.2.2.2 invia cube zone prefixs omitted So calls coming FROM gk2 destined for either ucm or cme zone used the cube. If I applied the invia key word on either ucm or cme zone directly, the cube was not invoked. This seems to conflict with the proctor guide mock lab 1 statement “invia command when defining the UCME zone would invoke the cube for calls coming in from a remote zone”. In my lab applying invia directly to destination zone had no affect and cube was not invoked. Am I missing something. ___ 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 Hardware Transcoder
Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.com wrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can’t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ 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 http://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] IPad Support
I want to green up my library. I am considering purchase of an IPad. Does anyone know if the IPExpert PDF's security will be supported VIA IPad? If not currently supported, is there a roadmap to get to that point? thanks Chris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Globalized + Dialing - Applying Called Party Number Type for Secondary Route Scenario (non-TEHO)
All - I am setting up + dialing on a self-made lab. A question has come up as to where the Called Party Number Type should be set. For this exercise, I want to find the best way to route calls through a system, utilizing alternate paths for failover scenarios. Those this does not take TEHO into account, I want a format that can easily accommodate TEHO situations. I believe my method below will do that. PSTN is expecting: Subscriber: Seven digits, Subscriber National: Eleven digits (incl. 1), National Intl: Undefined digits, Intl I have also setup translation patterns in PT_US_MN_EP_PSTN setup as: 9.952[2-9]XX-- Predot, Prefix +1 -- Result: +19525163748 (local MN) 9.1[2-9]XX[2-9]XX -- Predot, Prefix + -- Result: +1615444 (remote long-distance in TN) 9.011!-- Predot, Prefix +-- Result: +3432141861 (remote international) I have route patterns in PT_US_MN_EP_PSTN setup as: \+1952[2-9]XX-- +19525163748 \+1[2-9]XX[2-9]XX-- +1615444 \+! -- +3432141861 \+!# -- +3432141861 Note: I am not using predot in the route patterns!! At this point, all dialed numbers have been globalized from their localized variants. I am trying to figure out if it would be best to apply the Called Party Number Type (Sub, Nat, Intl) at the Translation Pattern, Route Pattern, Route List (i.e. Route Group settings), or Called Party Transformation on the gateway level. If I did not modify the Called Party Number Type at the Translation Pattern or Route Pattern, it would allow me to configure different call treatment at the route list level for TEHO. However, I think it would be best to apply that setting at the Called Party Transformation pattern on the MGCP gateways (avoiding H.323 for sake of conversation). I have two locations that calls will be sent out, MN and TN. For example: RL_US_MN_PSTN RG_US_MN RG_US_TN RL_US_TN_PSTN RG_US_TN RG_US_MN At this point, I need to convert the globalized numbers to their localized variants at the specific locations' voice gateways Called Party Transformation on MN gateway: +\1952.XXX -- Strip predot, Subscriber +\.1[2-9]XX[2-9]XX -- Strip predot, National +\.!-- Strip predot, Prefix 011, International Called Party Transformation on TN gateway: +\1615.XXX -- Strip predot, Subscriber +\.1[2-9]XX[2-9]XX -- Strip predot, National +\.!-- Strip predot, Prefix 011, International Primary Route (MN call out MN gateway) Verification: 1. User in MN dials local number as 9.9525163748 (my desk phone, give me a call :)) 2. Translation pattern changes to +19525163748 3. Matches route pattern of +1952[2-9]XX 4. Route pattern sent via RL_US_MN_PSTN to RG_US_MN 5. RG_US_MN sends call to US_MN_Gateway1 6. US_MN_Gateway1 has a called transformation pattern of +\1952.XXX (Subscriber) 7. Call goes out the US_MN_Gateway1 as 5163748 (Subscriber) Secondary Route (MN call out TN gateway) Verification: 1. User in MN dials local number as 9.9525163748 (my desk phone, give me a call :)) 2. Translation pattern changes to +19525163748 3. Matches route pattern of +1952[2-9]XX 4. Route pattern sent via RL_US_MN_PSTN to RG_US_TN (RG_US_MN is down and not functioning) 5. RG_US_TN sends call to US_TN_Gateway1 6. US_TN_Gateway1 has a called transformation pattern of +\.1[2-9]XX[2-9]XX (National) 7. Call goes out the US_TN_Gateway1 as 19525163748 (National) Any feedback would be appreciated. It took me about 30 minutes to think this through and type it out. Because it takes so long, I am trying to build a strawman structure that I can easily drop into the lab and modify to support my needs. What say ye? Matthew Berry ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MVA
Hi Folks I am doing MVA , When i dial the MVA number , I am able to hear the prompt. I dial a PSTN number , but the call disconnect . Can any body suggest me what can be the reason . Rgds Anu. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA
Check the CSS on the remote destination profile you're calling from. If you do a debug isdn q931 on the PSTN gateway, do you see the call hit the gateway? Your rerouting CSS on the RDP is used for calls out to your RD. Your CSS on the RDP is used for calls through MVA that are routed out through your PSTN gateway. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of anupam TYAGI Sent: Friday, March 12, 2010 9:27 AM To: ccie_voice-requ...@onlinestudylist.com; ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA Hi Folks I am doing MVA , When i dial the MVA number , I am able to hear the prompt. I dial a PSTN number , but the call disconnect . Can any body suggest me what can be the reason . Rgds Anu. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA
i saw the call hit the gateway . RDP is having the same CSS as phone CSS On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Check the CSS on the remote destination profile you’re calling from. If you do a “debug isdn q931” on the PSTN gateway, do you see the call hit the gateway? Your rerouting CSS on the RDP is used for calls out to your RD. Your CSS on the RDP is used for calls through MVA that are routed out through your PSTN gateway. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI *Sent:* Friday, March 12, 2010 9:27 AM *To:* ccie_voice-requ...@onlinestudylist.com; ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] MVA Hi Folks I am doing MVA , When i dial the MVA number , I am able to hear the prompt. I dial a PSTN number , but the call disconnect . Can any body suggest me what can be the reason . Rgds Anu. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA
are you maybe calling to a remote location (g.729) and therefore a xcoder is required, but not set up correctly? 2010/3/12 anupam TYAGI anuf...@gmail.com i saw the call hit the gateway . RDP is having the same CSS as phone CSS On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Check the CSS on the remote destination profile you’re calling from. If you do a “debug isdn q931” on the PSTN gateway, do you see the call hit the gateway? Your rerouting CSS on the RDP is used for calls out to your RD. Your CSS on the RDP is used for calls through MVA that are routed out through your PSTN gateway. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI *Sent:* Friday, March 12, 2010 9:27 AM *To:* ccie_voice-requ...@onlinestudylist.com; ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] MVA Hi Folks I am doing MVA , When i dial the MVA number , I am able to hear the prompt. I dial a PSTN number , but the call disconnect . Can any body suggest me what can be the reason . Rgds Anu. ___ 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] MVA
if i dial that external number without MVA it goes through ,but when in MVA i get a disconnect when calling this external number ( so don't seems to be codec issue ) On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer myciscov...@gmail.comwrote: are you maybe calling to a remote location (g.729) and therefore a xcoder is required, but not set up correctly? 2010/3/12 anupam TYAGI anuf...@gmail.com i saw the call hit the gateway . RDP is having the same CSS as phone CSS On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Check the CSS on the remote destination profile you’re calling from. If you do a “debug isdn q931” on the PSTN gateway, do you see the call hit the gateway? Your rerouting CSS on the RDP is used for calls out to your RD. Your CSS on the RDP is used for calls through MVA that are routed out through your PSTN gateway. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI *Sent:* Friday, March 12, 2010 9:27 AM *To:* ccie_voice-requ...@onlinestudylist.com; ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] MVA Hi Folks I am doing MVA , When i dial the MVA number , I am able to hear the prompt. I dial a PSTN number , but the call disconnect . Can any body suggest me what can be the reason . Rgds Anu. ___ 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] CCIE_Voice Digest, Vol 49, Issue 68
Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ 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] PSTN Call Distortion Between T1/E1
Hi Otto, Thanks for the advice. In your second paragraph the opposite was actually the case. The E1 voice-ports were originally showing a-law, and had distortion. I hard set u-law on the E1 ports between the gateway and PSTN router and the distortion went away. Perhaps that is what you meant? I took a look at the link you included. I'll have to do some testing but my main question is how is this handled in the real world at the provider level? Thanks, Jason From: Otto Sanchez [mailto:o...@ipexpert.com] Sent: Friday, March 12, 2010 4:59 AM To: Jason Granat Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1 Hello Jason, E1's and T1's will always use a-law and u-law companding mechanism respectively, this is used to give more resolution to low voice frequencies when digitizing an analog signal (the mechanism is also used in the other end for digital to analogue conversion), each mechanism is designed exclusively to work with its voice digital standard and cannot be used conversely, In that sense, my guess is that before applying that command in your E1 port, the companding type was u-law, you can verify this using the sh voice port command (perhaps the default configuration of a-law was somehow overwritten by a cptone command in the same port configuration), and when you hardcoded the a-law companding type everything worked as expected, I also found a note in the Cisco IOS Voice Port Configuration Guide, which says that the command is used when cross-connecting in a local router, http://www.cisco.com/en/US/partner/docs/ios/voice/voiceport/configuration/guide/vp_cfg_digital_vps_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1009871 HTH, On Thu, Mar 11, 2010 at 2:37 PM, Jason Granat j...@slash128.commailto:j...@slash128.com wrote: So I've got this partially figured out. It had to do with the compand-type. E1 was a-law and T1 was u-law. I set the E1 side for u-law and it sounds correct now. The final thing I am trying to figure out is how to 'trans-compand' (if that is the correct term) on the PSTN gateway. As it sits I had to change the compand-type between the PSTN and E1 gateway. I don't have experience with foreign connectivity so maybe this is the way it is done in the real world but I am thinking that perhaps the E1 site may not want or be able to change their compand-type, so can it be changed at the PSTN level between a-law and u-law locations? Thanks, Jason From: Jason Granat Sent: Thursday, March 11, 2010 9:46 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: PSTN Call Distortion Between T1/E1 Perhaps this is something simple that I am overlooking but I have the generic setup running in my home lab with 3 gateways and one PSTN router. 2 of the gateways are T1 and one is E1. The PSTN router is also running CME with a 7960 to simulate PSTN destinations. Calls from any site to the PSTN phone are fine. Calls between T1 sites are fine. Calls between T1 and E1 sites are distorted, like the gain is way too high. I tried playing with the gain on the voice-port but no luck. I'm not finding much online or in Cisco docs. Any suggestions? Thanks, Jason http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] via gatekeeper invia key word
Thanks Otto, if this is the case then I believe the explanation Mark S. gives on the VOD is incorrect. As he references the invia between local zones. From: Otto Sanchez [mailto:o...@ipexpert.com] Sent: Friday, March 12, 2010 6:14 AM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] via gatekeeper invia key word Hi Jeff, According to your lab results, you are describing the expected behavior, more information at: http://www.cisco.com/en/US/partner/docs/ios/voice/cubegk/configuration/guide/ve-gk-config_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1225776 Thanks!, On Thu, Mar 11, 2010 at 9:55 AM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: I am struggling a bit with the invia concept. I think I understand the outvia. When I lab this up I find the following. Invia only applies to calls coming from a remote GK. In order for call to use cube I had to configure the invia key word on the actual remote zone.not on the destination zone. Sample config of my invia GK gk zone local ucm cisco.comhttp://cisco.com 1.1.1.1 gk zone local cube gk zone local cme gk zone remote gk2 lab.comhttp://lab.com 2.2.2.2 invia cube zone prefixs omitted So calls coming FROM gk2 destined for either ucm or cme zone used the cube. If I applied the invia key word on either ucm or cme zone directly, the cube was not invoked. This seems to conflict with the proctor guide mock lab 1 statement invia command when defining the UCME zone would invoke the cube for calls coming in from a remote zone. In my lab applying invia directly to destination zone had no affect and cube was not invoked. Am I missing something. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://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
[OSL | CCIE_Voice] ip phone on layer 3 interface
Hello: I want to connect an ip phone to a 2811 fast ether 0/0 interface (pstn router) this way I wouldn't need a switch for the pstn phone A xcable is needed but I'm not sure if a layer 3 interface is l2 switching capable Any suggestion? Thanks _ Recibe en tu móvil un SMS con tu Hotmail recibido. ¡Date de alta ya! http://serviciosmoviles.es.msn.com/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja _ From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja -- *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* Friday, March 12, 2010 12:33 PM *To:* Vishal Preenja *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ 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] CCIE_Voice Digest, Vol 49, Issue 68
Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja _ From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPCCx 7.x
Gent's, Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..? I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said non Cisco hardware install will now stop T E-mail:thomas.k...@compucom.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja Sent: Friday, March 12, 2010 12:46 PM To: 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja _ From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hi, You want to transcode at the br2 rtr as I suppose your requirement is to transport the call using g.729 over the wan right?, if that's the case, make sure the incoming dial-peer codec is set to g.729, in that case the transcoder at br2 shoud be invoked if the sip phone codec is set to g.711, On Fri, Mar 12, 2010 at 10:03 AM, Omotayo adefilabi...@gmail.com wrote: Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.com wrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can’t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ 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 http://www.ipexpert.com/ ___ 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] IPCCx 7.x
Hi Thomas, Are ladies also allowed to reply? :-) Pls check the hardware compatibility matrix below: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/cr s/express_compatibility/matrix/crscomtx.pdf HP DL380 G3 is not supported with IPCC 7.x onwards. Thanks and Regards, Vishal Preenja _ From: Thomas Koch [mailto:koch1...@comcast.net] Sent: Friday, March 12, 2010 1:50 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Gent's, Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..? I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said non Cisco hardware install will now stop T E-mail:thomas.k...@compucom.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja Sent: Friday, March 12, 2010 12:46 PM To: 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja _ From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPCCx 7.x
Thomas, I can't verify this will work for you (and I certainly wouldn't do this for a production machine) however, you can apply the following registry hack (which on a plain windows install, will allow UCCX to install). If it works, let me know. Cheers, Tanner Ezell On Fri, Mar 12, 2010 at 2:39 PM, Vishal Preenja vpree...@cisco.com wrote: Hi Thomas, Are ladies also allowed to reply? J Pls check the hardware compatibility matrix below: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf HP DL380 G3 is not supported with IPCC 7.x onwards. Thanks and Regards, Vishal Preenja From: Thomas Koch [mailto:koch1...@comcast.net] Sent: Friday, March 12, 2010 1:50 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Gent’s, Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..? I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said “non Cisco hardware” install will now stop” T E-mail:thomas.k...@compucom.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja Sent: Friday, March 12, 2010 12:46 PM To: 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ 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 -- Regards, Tanner Ezell win2k3-reg-cisco.reg Description: Binary data ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPCCx 7.x
Yes.of course Thanks. Thomas J Koch Consultant CCDA, CCNA, CCVP Cisco IPT Design Specalist CompuCom Cell: 630-808-4910 E-mail:thom.k...@compucom.com From: Vishal Preenja [mailto:vpree...@cisco.com] Sent: Friday, March 12, 2010 1:39 PM To: 'Thomas Koch'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Hi Thomas, Are ladies also allowed to reply? J Pls check the hardware compatibility matrix below: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/cr s/express_compatibility/matrix/crscomtx.pdf HP DL380 G3 is not supported with IPCC 7.x onwards. Thanks and Regards, Vishal Preenja _ From: Thomas Koch [mailto:koch1...@comcast.net] Sent: Friday, March 12, 2010 1:50 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Gent's, Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..? I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said non Cisco hardware install will now stop T E-mail:thomas.k...@compucom.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja Sent: Friday, March 12, 2010 12:46 PM To: 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja _ From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPCCx 7.x
Hi All, I tried installing it on a VMWare 2k3 Server, but never succeeded so far, if anyone has done it, please could you do me a favor by sharing your secret? Regards Wilson Samuel From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Thomas Koch Sent: Friday, March 12, 2010 2:49 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x Yes...of course Thanks... Thomas J Koch Consultant CCDA, CCNA, CCVP Cisco IPT Design Specalist CompuCom Cell: 630-808-4910 E-mail:thom.k...@compucom.com From: Vishal Preenja [mailto:vpree...@cisco.com] Sent: Friday, March 12, 2010 1:39 PM To: 'Thomas Koch'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Hi Thomas, Are ladies also allowed to reply? :) Pls check the hardware compatibility matrix below: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf HP DL380 G3 is not supported with IPCC 7.x onwards. Thanks and Regards, Vishal Preenja From: Thomas Koch [mailto:koch1...@comcast.net] Sent: Friday, March 12, 2010 1:50 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Gent's, Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..? I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said non Cisco hardware install will now stop T E-mail:thomas.k...@compucom.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja Sent: Friday, March 12, 2010 12:46 PM To: 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.commailto:vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.commailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.commailto:vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPad Support
IMHO, The best way to read PDFs is to read on a computer (aka Laptop, Palmtop, Tablet PC or Mac versions of the equivalent) and find yourself the most portable one that suits you. I guess the E-Readers (Kindle, Sony et al) are not that user friendly when it comes to PDF, I tried doing it with Amazon Kindle and I guess I'm not really getting benefitted out of it. However I'm keeping the Kindle because, all the newspaper and magazine subscriptions are a bit ok deal and its really portable, also most of the books are so far cheaper on Kindle edition (excluding the Cisco Press). HTH Kind Regards Wilson Samuel -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of bontacommunications Sent: Friday, March 12, 2010 9:42 AM To: ccie_voice-boun...@onlinestudylist.com Cc: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] IPad Support I want to green up my library. I am considering purchase of an IPad. Does anyone know if the IPExpert PDF's security will be supported VIA IPad? If not currently supported, is there a roadmap to get to that point? thanks Chris ___ 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] IPCCx 7.x
Do you have the OS ISOs? I created a VM with 2 nics (one left disabled), 1 proc, 2G RAM, 80G IDE Drive, and optical drive. I used the HP OS 1.3a disc. After the OS is loaded and rebooted it will start the add new hardware wizard. Don't click that. Just run the install VM Ware tools. After the tools are installed the drivers get updated and the wizard goes away. Then reboot again. Assign an IP to one NIC. Leave the second NIC disabled. Install the UCCX IP IVR disc. That's it... You could install VNC if you want instead of managing from the VMWare console. I'm sure other VM settings combos might work but this has always worked for me. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of wilson.sam...@usc-bt.com Sent: Friday, March 12, 2010 12:41 PM To: koch1...@comcast.net; vpree...@cisco.com; adefilabi...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x Hi All, I tried installing it on a VMWare 2k3 Server, but never succeeded so far, if anyone has done it, please could you do me a favor by sharing your secret? Regards Wilson Samuel From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Thomas Koch Sent: Friday, March 12, 2010 2:49 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x Yes...of course Thanks... Thomas J Koch Consultant CCDA, CCNA, CCVP Cisco IPT Design Specalist CompuCom Cell: 630-808-4910 E-mail:thom.k...@compucom.com From: Vishal Preenja [mailto:vpree...@cisco.com] Sent: Friday, March 12, 2010 1:39 PM To: 'Thomas Koch'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Hi Thomas, Are ladies also allowed to reply? :) Pls check the hardware compatibility matrix below: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf HP DL380 G3 is not supported with IPCC 7.x onwards. Thanks and Regards, Vishal Preenja From: Thomas Koch [mailto:koch1...@comcast.net] Sent: Friday, March 12, 2010 1:50 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Gent's, Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..? I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said non Cisco hardware install will now stop T E-mail:thomas.k...@compucom.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja Sent: Friday, March 12, 2010 12:46 PM To: 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.commailto:vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.commailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.commailto:vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you
Re: [OSL | CCIE_Voice] IPad Support
I challenge IP Expert to build interactive learning content for the iPad (as many publishers are now looking into doing). If done properly, would give IPX an incredible advantage over the competition. As they say, Evolve or Die. :) On Fri, Mar 12, 2010 at 3:56 PM, wilson.sam...@usc-bt.com wrote: IMHO, The best way to read PDFs is to read on a computer (aka Laptop, Palmtop, Tablet PC or Mac versions of the equivalent) and find yourself the most portable one that suits you. I guess the E-Readers (Kindle, Sony et al) are not that user friendly when it comes to PDF, I tried doing it with Amazon Kindle and I guess I'm not really getting benefitted out of it. However I'm keeping the Kindle because, all the newspaper and magazine subscriptions are a bit ok deal and its really portable, also most of the books are so far cheaper on Kindle edition (excluding the Cisco Press). HTH Kind Regards Wilson Samuel -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of bontacommunications Sent: Friday, March 12, 2010 9:42 AM To: ccie_voice-boun...@onlinestudylist.com Cc: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] IPad Support I want to green up my library. I am considering purchase of an IPad. Does anyone know if the IPExpert PDF's security will be supported VIA IPad? If not currently supported, is there a roadmap to get to that point? thanks Chris ___ 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 -- Regards, Tanner Ezell ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPad Support
Personally, I wouldn't prefer to buy Apple products (sorry mates, after the ebook price debate, I have lost respect for Mr. Jobs and APPL in general). I would prefer them to develop it for Kindle and Sony E-Readers (I guess these at the moment have the max. marketshare) also iPad comes at a very high price and hasn't been even launched, though the demand seems to be looming. I'm not sure if people have around 600 USD to buy E-readers just for the sake of a reader and some computing, esp. when Mac OS challenged people like me have remained loyal to Linux and Windows OS. Again, it's the survival of the fittest, May the Best Product Win!! Regards Wilson Samuel -Original Message- From: Tanner Ezell [mailto:tanner.ez...@gmail.com] Sent: Friday, March 12, 2010 4:10 PM To: Samuel, Wilson Cc: bontacommunicati...@gmail.com; ccie_voice-boun...@onlinestudylist.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPad Support I challenge IP Expert to build interactive learning content for the iPad (as many publishers are now looking into doing). If done properly, would give IPX an incredible advantage over the competition. As they say, Evolve or Die. :) On Fri, Mar 12, 2010 at 3:56 PM, wilson.sam...@usc-bt.com wrote: IMHO, The best way to read PDFs is to read on a computer (aka Laptop, Palmtop, Tablet PC or Mac versions of the equivalent) and find yourself the most portable one that suits you. I guess the E-Readers (Kindle, Sony et al) are not that user friendly when it comes to PDF, I tried doing it with Amazon Kindle and I guess I'm not really getting benefitted out of it. However I'm keeping the Kindle because, all the newspaper and magazine subscriptions are a bit ok deal and its really portable, also most of the books are so far cheaper on Kindle edition (excluding the Cisco Press). HTH Kind Regards Wilson Samuel -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of bontacommunications Sent: Friday, March 12, 2010 9:42 AM To: ccie_voice-boun...@onlinestudylist.com Cc: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] IPad Support I want to green up my library. I am considering purchase of an IPad. Does anyone know if the IPExpert PDF's security will be supported VIA IPad? If not currently supported, is there a roadmap to get to that point? thanks Chris ___ 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 -- Regards, Tanner Ezell ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPCCx 7.x
No.Just WIN2k3 enterprise. Thomas J Koch Unified Communicatons Consultant CCDA, CCNA, CCVP Cisco IPT Design Specalist CompuCom Cell: 630-808-4910 E-mail:thom.k...@compucom.com From: Jason Granat [mailto:j...@slash128.com] Sent: Friday, March 12, 2010 3:05 PM To: wilson.sam...@usc-bt.com; koch1...@comcast.net; vpree...@cisco.com; adefilabi...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Do you have the OS ISOs? I created a VM with 2 nics (one left disabled), 1 proc, 2G RAM, 80G IDE Drive, and optical drive. I used the HP OS 1.3a disc. After the OS is loaded and rebooted it will start the add new hardware wizard. Don't click that. Just run the install VM Ware tools. After the tools are installed the drivers get updated and the wizard goes away. Then reboot again. Assign an IP to one NIC. Leave the second NIC disabled. Install the UCCX IP IVR disc. That's it. You could install VNC if you want instead of managing from the VMWare console. I'm sure other VM settings combos might work but this has always worked for me. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of wilson.sam...@usc-bt.com Sent: Friday, March 12, 2010 12:41 PM To: koch1...@comcast.net; vpree...@cisco.com; adefilabi...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x Hi All, I tried installing it on a VMWare 2k3 Server, but never succeeded so far, if anyone has done it, please could you do me a favor by sharing your secret? Regards Wilson Samuel From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Thomas Koch Sent: Friday, March 12, 2010 2:49 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x Yes.of course Thanks. Thomas J Koch Consultant CCDA, CCNA, CCVP Cisco IPT Design Specalist CompuCom Cell: 630-808-4910 E-mail:thom.k...@compucom.com From: Vishal Preenja [mailto:vpree...@cisco.com] Sent: Friday, March 12, 2010 1:39 PM To: 'Thomas Koch'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Hi Thomas, Are ladies also allowed to reply? J Pls check the hardware compatibility matrix below: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/cr s/express_compatibility/matrix/crscomtx.pdf HP DL380 G3 is not supported with IPCC 7.x onwards. Thanks and Regards, Vishal Preenja _ From: Thomas Koch [mailto:koch1...@comcast.net] Sent: Friday, March 12, 2010 1:50 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Gent's, Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..? I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said non Cisco hardware install will now stop T E-mail:thomas.k...@compucom.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja Sent: Friday, March 12, 2010 12:46 PM To: 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja _ From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have
Re: [OSL | CCIE_Voice] IPad Support
iPad sucks big time! Sorry! It's not that I am not an Apple fan..have got iPhone, iPod etc but this big bulky thing looks very awkward.. Ash wilson.sam...@usc-bt.com wrote: Personally, I wouldn't prefer to buy Apple products (sorry mates, after the ebook price debate, I have lost respect for Mr. Jobs and APPL in general). I would prefer them to develop it for Kindle and Sony E-Readers (I guess these at the moment have the max. marketshare) also iPad comes at a very high price and hasn't been even launched, though the demand seems to be looming. I'm not sure if people have around 600 USD to buy E-readers just for the sake of a reader and some computing, esp. when Mac OS challenged people like me have remained loyal to Linux and Windows OS. Again, it's the survival of the fittest, May the Best Product Win!! Regards Wilson Samuel -Original Message- From: Tanner Ezell [mailto:tanner.ez...@gmail.com] Sent: Friday, March 12, 2010 4:10 PM To: Samuel, Wilson Cc: bontacommunicati...@gmail.com; ccie_voice-boun...@onlinestudylist.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPad Support I challenge IP Expert to build interactive learning content for the iPad (as many publishers are now looking into doing). If done properly, would give IPX an incredible advantage over the competition. As they say, Evolve or Die. :) On Fri, Mar 12, 2010 at 3:56 PM, wilson.sam...@usc-bt.com wrote: IMHO, The best way to read PDFs is to read on a "computer" (aka Laptop, Palmtop, Tablet PC or Mac versions of the equivalent) and find yourself the most portable one that suits you. I guess the E-Readers (Kindle, Sony et al) are not that user friendly when it comes to PDF, I tried doing it with Amazon Kindle and I guess I'm not really getting benefitted out of it. However I'm keeping the Kindle because, all the newspaper and magazine subscriptions are a bit ok deal and its really portable, also most of the books are so far cheaper on Kindle edition (excluding the Cisco Press). HTH Kind Regards Wilson Samuel -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of bontacommunications Sent: Friday, March 12, 2010 9:42 AM To: ccie_voice-boun...@onlinestudylist.com Cc: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] IPad Support I want to "green up" my library. I am considering purchase of an IPad. Does anyone know if the IPExpert PDF's security will be supported VIA IPad? If not currently supported, is there a roadmap to get to that point? thanks Chris ___ 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
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hello Otto, Yes requirement is to transport g729 over the WAN if i want to transcoder on the trunk. What do i need to do because quetion says use the hq resources The last time i applied the transcoder to the trunk, When hq phone call the sip phone on br2, i get a reorder tone When the sip phone on the br2 calls the hq phone, it disconnects on pick up and continues to ring on the sip phone Thanks On Fri, Mar 12, 2010 at 8:37 PM, Otto Sanchez o...@ipexpert.com wrote: Hi, You want to transcode at the br2 rtr as I suppose your requirement is to transport the call using g.729 over the wan right?, if that's the case, make sure the incoming dial-peer codec is set to g.729, in that case the transcoder at br2 shoud be invoked if the sip phone codec is set to g.711, On Fri, Mar 12, 2010 at 10:03 AM, Omotayo adefilabi...@gmail.com wrote: Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.comwrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can’t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ 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 http://www.ipexpert.com/ ___ 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 http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPCCx 7.x
I haven't tried that route. If you get the NFR kit it comes with the OS discs. Sent while mobile. On Mar 12, 2010, at 13:22, Thomas Koch koch1...@comcast.netmailto:koch1...@comcast.net wrote: No…Just WIN2k3 enterprise… Thomas J Koch Unified Communicatons Consultant CCDA, CCNA, CCVP Cisco IPT Design Specalist CompuCom Cell: 630-808-4910 E-mail:thom.k...@compucom.com From: Jason Granat [mailto:j...@slash128.com] Sent: Friday, March 12, 2010 3:05 PM To: wilson.sam...@usc-bt.commailto:wilson.sam...@usc-bt.com; koch1...@comcast.netmailto:koch1...@comcast.net; vpree...@cisco.commailto:vpree...@cisco.com; mailto:adefilabi...@gmail.com adefilabi...@gmail.commailto:adefilabi...@gmail.com Cc: mailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Do you have the OS ISOs? I created a VM with 2 nics (one left disabled), 1 proc, 2G RAM, 80G IDE Drive, and optical drive. I used the HP OS 1.3a disc. After the OS is loaded and rebooted it will start the add new hardware wizard. Don’t click that. Just run the install VM Ware tools. After the tools are installed the drivers get updated and the wizard goes away. Then reboot again. Assign an IP to one NIC. Leave the second NIC disabled. Install the UCCX IP IVR disc. That’s it… You could install VNC if you want instead of managing from the VMWare console. I’m sure other VM settings combos might work but this has always worked for me. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of wilson.sam...@usc-bt.commailto:wilson.sam...@usc-bt.com Sent: Friday, March 12, 2010 12:41 PM To: koch1...@comcast.net; vpree...@cisco.com; adefilabi...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x Hi All, I tried installing it on a VMWare 2k3 Server, but never succeeded so far, if anyone has done it, please could you do me a favor by sharing your secret? Regards Wilson Samuel From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Thomas Koch Sent: Friday, March 12, 2010 2:49 PM To: 'Vishal Preenja'; 'Omotayo' Cc: mailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x Yes…of course Thanks… Thomas J Koch Consultant CCDA, CCNA, CCVP Cisco IPT Design Specalist CompuCom Cell: 630-808-4910 E-mail:thom.k...@compucom.com From: Vishal Preenja [mailto:vpree...@cisco.com] Sent: Friday, March 12, 2010 1:39 PM To: 'Thomas Koch'; 'Omotayo' Cc: mailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Hi Thomas, Are ladies also allowed to reply? ☺ Pls check the hardware compatibility matrix below: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdfhttp://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf HP DL380 G3 is not supported with IPCC 7.x onwards. Thanks and Regards, Vishal Preenja From: Thomas Koch [mailto:koch1...@comcast.net] Sent: Friday, March 12, 2010 1:50 PM To: 'Vishal Preenja'; 'Omotayo' Cc: mailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Gent’s, Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..? I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said “non Cisco hardware” install will now stop” T E-mail:thomas.k...@compucom.com From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja Sent: Friday, March 12, 2010 12:46 PM To: 'Omotayo' Cc: mailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: mailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja
Re: [OSL | CCIE_Voice] Pt2. Globalized + Dialing - Applying Called Party Number Type for Secondary Route Scenario (non-TEHO)
Hi Mathew - excellent post, and looks like you have successfully answered your own question! I'm going through the CUCM normalization study section (again) and i would agreed that we set the Called Party Number type at the exiting PSTN gateway using called party xform pattern as: - The call may take multiple routes, if we apply at the xlate pattern, the number type may need to be changed again before the call exits to the PSTN. - RP / RL is still not close enough to the actual PSTN gateway (especially when using Loc RG) to avoid the type to be changed again. - Called Party xform pattern is set at the PSTN gateway so no risk of additional type changes before exiting to PSTN. We could also apply this at DP level for phones and gateways (as the phones will ignore called party xform rules). Just my opinion - but always open to ideas! thks b From: mjbe...@krollontrack.com To: ccie_voice@onlinestudylist.com Date: Fri, 12 Mar 2010 09:22:08 -0600 Subject: [OSL | CCIE_Voice] Pt2. Globalized + Dialing - Applying Called Party Number Type for Secondary Route Scenario (non-TEHO) I could also simplify my translation patterns and make them available to ALL sites. By creating a separate partition called PT_US_Translate and setting up the following patterns: 9.!à Predot, Prefix +1 à Result: +19525163748 (local MN) 9.1!à Predot, Prefix +à Result: +1615444 (remote long-distance in TN) 9011.! à Predot, Prefix +à Result: +3432141861 (remote international) From: Berry, Matthew J. Sent: Friday, March 12, 2010 9:12 AM To: OSL Group Subject: Globalized + Dialing - Applying Called Party Number Type for Secondary Route Scenario (non-TEHO) All – I am setting up + dialing on a self-made lab. A question has come up as to where the Called Party Number Type should be set. For this exercise, I want to find the best way to route calls through a system, utilizing alternate paths for failover scenarios. Those this does not take TEHO into account, I want a format that can easily accommodate TEHO situations. I believe my method below will do that. PSTN is expecting: Subscriber: Seven digits, Subscriber National: Eleven digits (incl. 1), National Intl: Undefined digits, Intl I have also setup translation patterns in PT_US_MN_EP_PSTN setup as: 9.952[2-9]XXà Predot, Prefix +1 à Result: +19525163748 (local MN) 9.1[2-9]XX[2-9]XX à Predot, Prefix + à Result: +1615444 (remote long-distance in TN) 9.011!à Predot, Prefix +à Result: +3432141861 (remote international) I have route patterns in PT_US_MN_EP_PSTN setup as: \+1952[2-9]XXà +19525163748 \+1[2-9]XX[2-9]XXà +1615444 \+! à +3432141861 \+!# à +3432141861 Note: I am not using predot in the route patterns!! At this point, all dialed numbers have been globalized from their localized variants. I am trying to figure out if it would be best to apply the Called Party Number Type (Sub, Nat, Intl) at the Translation Pattern, Route Pattern, Route List (i.e. Route Group settings), or Called Party Transformation on the gateway level. If I did not modify the Called Party Number Type at the Translation Pattern or Route Pattern, it would allow me to configure different call treatment at the route list level for TEHO. However, I think it would be best to apply that setting at the Called Party Transformation pattern on the MGCP gateways (avoiding H.323 for sake of conversation). I have two locations that calls will be sent out, MN and TN. For example: RL_US_MN_PSTN RG_US_MN RG_US_TN RL_US_TN_PSTN RG_US_TN RG_US_MN At this point, I need to convert the globalized numbers to their localized variants at the specific locations’ voice gateways Called Party Transformation on MN gateway: +\1952.XXX à Strip predot, Subscriber +\.1[2-9]XX[2-9]XX à Strip predot, National +\.!à Strip predot, Prefix 011, International Called Party Transformation on TN gateway: +\1615.XXX à Strip predot, Subscriber +\.1[2-9]XX[2-9]XX à Strip predot, National +\.!à Strip predot, Prefix 011, International Primary Route (MN call out MN gateway) Verification: 1. User
Re: [OSL | CCIE_Voice] IPCCx 7.x
Jason, Thanks…I have our sales team trying get me a copy… Thomas J Koch Unified Communicatons Consultant CCDA, CCNA, CCVP Cisco IPT Design Specalist CompuCom Cell: 630-808-4910 E-mail:thom.k...@compucom.com From: Jason Granat [mailto:j...@slash128.com] Sent: Friday, March 12, 2010 3:31 PM To: Thomas Koch Cc: wilson.sam...@usc-bt.com; vpree...@cisco.com; adefilabi...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x I haven't tried that route. If you get the NFR kit it comes with the OS discs. Sent while mobile. On Mar 12, 2010, at 13:22, Thomas Koch koch1...@comcast.net wrote: No…Just WIN2k3 enterprise… Thomas J Koch Unified Communicatons Consultant CCDA, CCNA, CCVP Cisco IPT Design Specalist CompuCom Cell: 630-808-4910 E-mail:thom.k...@compucom.com From: Jason Granat [mailto:j...@slash128.com] Sent: Friday, March 12, 2010 3:05 PM To: wilson.sam...@usc-bt.com; koch1...@comcast.net; vpree...@cisco.com; adefilabi...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Do you have the OS ISOs? I created a VM with 2 nics (one left disabled), 1 proc, 2G RAM, 80G IDE Drive, and optical drive. I used the HP OS 1.3a disc. After the OS is loaded and rebooted it will start the add new hardware wizard. Don’t click that. Just run the install VM Ware tools. After the tools are installed the drivers get updated and the wizard goes away. Then reboot again. Assign an IP to one NIC. Leave the second NIC disabled. Install the UCCX IP IVR disc. That’s it… You could install VNC if you want instead of managing from the VMWare console. I’m sure other VM settings combos might work but this has always worked for me. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of wilson.sam...@usc-bt.com Sent: Friday, March 12, 2010 12:41 PM To: koch1...@comcast.net; vpree...@cisco.com; adefilabi...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x Hi All, I tried installing it on a VMWare 2k3 Server, but never succeeded so far, if anyone has done it, please could you do me a favor by sharing your secret? Regards Wilson Samuel From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Thomas Koch Sent: Friday, March 12, 2010 2:49 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCCx 7.x Yes…of course Thanks… Thomas J Koch Consultant CCDA, CCNA, CCVP Cisco IPT Design Specalist CompuCom Cell: 630-808-4910 E-mail:thom.k...@compucom.com From: Vishal Preenja [mailto:vpree...@cisco.com] Sent: Friday, March 12, 2010 1:39 PM To: 'Thomas Koch'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Hi Thomas, Are ladies also allowed to reply? J Pls check the hardware compatibility matrix below: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_compatibility/matrix/crscomtx.pdf HP DL380 G3 is not supported with IPCC 7.x onwards. Thanks and Regards, Vishal Preenja _ From: Thomas Koch [mailto:koch1...@comcast.net] Sent: Friday, March 12, 2010 1:50 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Gent’s, Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..? I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said “non Cisco hardware” install will now stop” T E-mail:thomas.k...@compucom.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja Sent: Friday, March 12, 2010 12:46 PM To: 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja _ From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL |
Re: [OSL | CCIE_Voice] UC and cme sip integration
Hello, it work ok now I was using the wrong ip address on the unity connection all the while Thanks On Fri, Mar 12, 2010 at 8:30 AM, Flemming Ortvald f...@netdesign.dk wrote: Unity connection can do both g729 and g711, you can use “voice class codec” on “voice register dn” to expand codec support for sip. Med venlig hilsen Flemming Ortvald Network System Eng. NetDesign A/S +45 4435 8346 Tænk på miljøet inden udskrivning af denne e-post og tilknyttede vedhæftninger *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* 11 March, 2010 20:58 *To:* Flemming Ortvald *Subject:* Re: [OSL | CCIE_Voice] UC and cme sip integration Hello, I have to configure a transcoder on the br2 router? Unity connection support g729 only? Rgd On Thu, Mar 11, 2010 at 8:24 PM, Flemming Ortvald f...@netdesign.dk wrote: You will need a transcoder or chnage the sip endpoints to support g.711, natively it only supports g.729 Best regards Flemming Ortvald Network System Eng. NetDesign A/S +45 4435 8346 Tænk på miljøet inden udskrivning af denne e-post og tilknyttede vedhæftninger *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo *Sent:* 11 March, 2010 20:07 *To:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] UC and cme sip integration Hello all, As anyone been able to get the SIP integration between Unity Connection and Cme to work? I followed the Proctorlabs Guide I posted this sometime lat week and revised as advised but keep getting a reorder tone( Number Unknown) when the message button is pressed Below is the relevant configuration voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 600 min 60 voice register global mode cme source-address 10.10.110.3 port 5060 max-dn 3 max-pool 6 authenticate register mwi reg-e164 voicemail 3600 tftp-path flash: create profile sync 0006855418337003 ! voice register dn 1 number 3002 call-forward b2bua busy 3600 call-forward b2bua mailbox 3002 call-forward b2bua noan 3600 timeout 12 name br2 phone 2 no-reg label br2 phone 2 mwi ! voice register dn 2 number 3003 call-forward b2bua busy 3600 call-forward b2bua mailbox 3003 call-forward b2bua noan 3600 timeout 12 name br2 phone 3 no-reg label br2 phone 3 mwi ! voice register pool 1 id mac .. type 7941 number 1 dn 1 dtmf-relay rtp-nte username 3002 password cisco ! voice register pool 2 id mac 001F.6C7E.D6FE type 7941 number 1 dn 2 dtmf-relay rtp-nte username 3003 password cisco dial-peer voice 200 voip max-conn 1 destination-pattern 3600 session protocol sipv2 session target ipv4:10.10.210.13 dtmf-relay rtp-nte codec g711ulaw ! ! telephony-service no auto-reg-ephone em logout 0:0 0:0 0:0 max-ephones 8 max-dn 8 ip source-address 10.10.202.1 port 2000 voicemail 3600 mwi relay max-conferences 8 gain -6 transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Mar 10 2010 15:22:39 ! ! ephone-dn 1 dual-line number 3001 no-reg primary label Br2 pHone 1 name Br2 Phone 1 call-forward busy 3600 call-forward noan 3600 timeout 12 ! ! sip-ua mwi-server ipv4:10.10.210.13 expires 3600 port 5060 transport udp unsolicited ! ! ephone 1 device-security-mode none mac-address 001E.EC15.996D type CIPC button 1:1 ! Thanks for the anticipated support ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
FYI, I was only able to get this to work using transcoder on CME. Had to match the codec between UCM trunk and incoming dial-peer on CME...then xcoder would engage on CME for the SIP phone. I have a hardware limitation in my home lab so I am not able to configure a xcoder on both UCM and CME simultaneously. From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 6:33 AM To: Otto Sanchez Cc: Jeff Cotter; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.commailto:o...@ipexpert.com wrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can't seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hello Jeff, All calls worked when i configure the xcoder on the cme The question says use the hq router resources- that is where i have issues thanks On Fri, Mar 12, 2010 at 11:33 PM, Jeff Cotter jcot...@voxns.com wrote: FYI, I was only able to get this to work using transcoder on CME. Had to match the codec between UCM trunk and incoming dial-peer on CME…then xcoder would engage on CME for the SIP phone. I have a hardware limitation in my home lab so I am not able to configure a xcoder on both UCM and CME simultaneously. *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* Friday, March 12, 2010 6:33 AM *To:* Otto Sanchez *Cc:* Jeff Cotter; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] SIP Hardware Transcoder Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.com wrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can’t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ 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 http://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
Re: [OSL | CCIE_Voice] MVA
i have the route pattern partion assigned to the CSS and this CSS is assigned to RDP but still the call disconnect when i dial the external number in MVZ On Fri, Mar 12, 2010 at 11:06 PM, Omotayo adefilabi...@gmail.com wrote: Hello, Berry is right. create a partition called pt-mva crease a CSS called css-mva put the partition in the css create a route pattern like 9.011! in partition pt-mva. the gateway can be the hq gateway if you wish discard predot assign the css to the remote destination profile this will work for you On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI anuf...@gmail.com wrote: if i dial that external number without MVA it goes through ,but when in MVA i get a disconnect when calling this external number ( so don't seems to be codec issue ) On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer myciscov...@gmail.comwrote: are you maybe calling to a remote location (g.729) and therefore a xcoder is required, but not set up correctly? 2010/3/12 anupam TYAGI anuf...@gmail.com i saw the call hit the gateway . RDP is having the same CSS as phone CSS On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Check the CSS on the remote destination profile you’re calling from. If you do a “debug isdn q931” on the PSTN gateway, do you see the call hit the gateway? Your rerouting CSS on the RDP is used for calls out to your RD. Your CSS on the RDP is used for calls through MVA that are routed out through your PSTN gateway. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI *Sent:* Friday, March 12, 2010 9:27 AM *To:* ccie_voice-requ...@onlinestudylist.com; ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] MVA Hi Folks I am doing MVA , When i dial the MVA number , I am able to hear the prompt. I dial a PSTN number , but the call disconnect . Can any body suggest me what can be the reason . Rgds Anu. ___ 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] CME busy-trigger-per-button
10.10.202.1 port 2000 voicemail 3600 mwi relay max-conferences 8 gain -6 transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Mar 10 2010 15:22:39 ! ! ephone-dn 1 dual-line number 3001 no-reg primary label Br2 pHone 1 name Br2 Phone 1 call-forward busy 3600 call-forward noan 3600 timeout 12 ! ! sip-ua mwi-server ipv4:10.10.210.13 expires 3600 port 5060 transport udp unsolicited ! ! ephone 1 device-security-mode none mac-address 001E.EC15.996D type CIPC button 1:1 ! Thanks for the anticipated support -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100312/76526596/attachment-0001.htm -- Message: 2 Date: Fri, 12 Mar 2010 14:33:41 -0800 From: Jeff Cotter jcot...@voxns.com Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder To: Omotayo adefilabi...@gmail.com, Otto Sanchez o...@ipexpert.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: 54cc1bd3093b6e41b86926c1657432f187a06...@ssfex1 Content-Type: text/plain; charset=us-ascii FYI, I was only able to get this to work using transcoder on CME. Had to match the codec between UCM trunk and incoming dial-peer on CME...then xcoder would engage on CME for the SIP phone. I have a hardware limitation in my home lab so I am not able to configure a xcoder on both UCM and CME simultaneously. From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 6:33 AM To: Otto Sanchez Cc: Jeff Cotter; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.commailto:o...@ipexpert.com wrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can't seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100312/bab98ebb/attachment-0001.htm -- Message: 3 Date: Sat, 13 Mar 2010 02:04:50 +0100 From: Omotayo adefilabi...@gmail.com Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder To: Jeff Cotter jcot...@voxns.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: 3082f9d41003121704j6e368e5egc989092a7343e...@mail.gmail.com Content-Type: text/plain; charset=windows-1252 Hello Jeff, All calls worked when i configure the xcoder on the cme The question says use the hq router resources- that is where i have issues thanks On Fri, Mar 12, 2010 at 11:33 PM, Jeff Cotter jcot...@voxns.com wrote: FYI, I was only able to get this to work using transcoder on CME. Had to match the codec between UCM trunk and incoming dial-peer on CME?then xcoder would engage on CME for the SIP phone. I have a hardware limitation in my home lab so I am not able to configure a xcoder on both UCM and CME simultaneously. *From:* Omotayo [mailto:adefilabi...@gmail.com] *Sent:* Friday, March 12, 2010 6:33 AM *To:* Otto Sanchez *Cc:* Jeff Cotter; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] SIP Hardware Transcoder Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.com wrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks
[OSL | CCIE_Voice] CME busy-trigger-per-button
Hey Folks Is it possible to set different busy-trigger-per-button for each button (line) on a phone on CME. For example : If I have line 1 : 1000 And line 2 : 1001 I need to limit 4 incoming calls on line 1 and limit 2 incoming calls on line 2. How can I achieve this. I guess busy-trigger-per-button sets limits for all the buttons on the particular ephone. Thanks Aman ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CIPC strange behavior !
Hi all. I am using CUCM 6. CIPC is installed on my pc. I am using single callmanager server that is also my tftp server for endpoints. Now while doing my studies i wanted to check how DNS might cause issues. In System-Server, i am using hostname instead of IP. Now what happens is that, CIPC after getting the .cnf.xml file, registers with tftp server successfully (which is also my callmanager server). Now what i read in student guide was, ip phone should not be able to register if its not able to resolve the hostname through DNS. (i am not using any DNS server and nor the entry for the hostname is present in my pc host file). Can someone tell me why is this the case ? i have searched alot but i couldnt find any thing stating that ip phone will fall back to tftp server in case primary callmanager fails !! Any input on this pls ? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
