Re: [OSL | CCIE_Voice] Gatekeeper understanding
Matched zone is UCM for this call since no prefix was matched. GK tries to terminate call in UCM zone. Consider technology prefix is matched to 3 but the remaining should be the zone prefix itself. You may need to send calls like 33... format to direct the call to UCME zone. Peter - Original Message - From: A NN prince_karim...@yahoo.com Date: Wednesday, May 30, 2012 11:47 am Subject: Re: [OSL | CCIE_Voice] Gatekeeper understanding To: Kevin Spicer ke...@kevinspicer.co.uk Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi Kevin, Yes, gateway command was there and the show gatek en showed the BR2-RTR as registered. How does the technology prefix works in this situation (two zones)? From: Kevin Spicer To: A NN prince_karim...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Wednesday, 30 May 2012, 5:56 Subject: Re: [OSL | CCIE_Voice] Gatekeeper understanding Hi, is ucme registered to the gk? (gateway command missing?) On 30 May 2012 03:08, A NN wrote: Hi list, I configured a GK with two zones: Zone UCM for CUCM Zone UCME for Br2 router When I make Calls from CUCM to CME (3xxx) call fails as per debugs below. I'm not using any CUBE. Can please someone explain what's wrong with my config. zone local UCM cisco.com 10.10.200.3 zone local UCME cisco.com zone prefix UCM 1... gw-priority 10 gk-trunk_2 zone prefix UCM 1... gw-priority 9 gk-trunk_1 zone prefix UCM 1... gw-priority 0 BR2-RTR zone prefix UCME 3... zone prefix UCM 5... gw-priority 10 gk-trunk_2 zone prefix UCM 5... gw-priority 9 gk-trunk_1 zone prefix UCM 5... gw-priority 0 BR2-RTR no shutdown interface Loopback0 ip address 10.10.110.3 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id UCME ipaddr 10.10.200.3 1719 h323-gateway voip h323-id BR2-RTR h323-gateway voip tech-prefix 3 h323-gateway voip bind srcaddr 10.10.110.3 May 26 18:37:05.630: //80849A2C1900/80849A2C1900/GK/rassrv_get_addrinfo: (3002) Matched tech-prefix 3 May 26 18:37:05.630: //80849A2C1900/80849A2C1900/GK/rassrv_get_addrinfo: (3002) unresolved zone prefix, using source zone UCM May 26 18:37:05.630: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 May 26 18:37:05.630: //80849A2C1900/80849A2C1900/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x48C6E830 May 26 18:37:05.630: //80849A2C1900/80849A2C1900/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_invianamelen=0 May 26 HQ-RTR(config-18:37:05.630: //80849A2C1900/80849A2C1900/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x48C6E830 May 26 18:37:05.630: //80849A2C1900/80849A2C1900/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_outvianamelen=0 May 26 18:37:05.630: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 May 26 18:37:05.630: //80849A2C1900/80849A2C1900/GK/rassrv_get_addrinfo: (3002) tech-prefix gateway selection failed. May 26 18:37:05.630: //80849A2C1900/80849A2C1900/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x107)gk)# HQ-RTR(config-gk)#e May 26 18:37:15.518: ////GK/gk_process: got a TIMER event ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] dtmf-relay h.245-signal or H.245 Alphanumeric
Definitely SRND clearly recommend signal type of methods. Actually this includes length in time the button was pressed, so it is more accurate. I think this is the reason for their choice. Regarding the interoperibility all H.323 version 2 compliant systems are required to support the h245-alphanumeric method, while support of the h245-signal method is optional. Dtmf-relay command allows us to give priority in DTMF methods. Does it make sense if we configure dtmf-relay h245-signal h245-alphanumeric? Peter - Original Message - From: Maik Stokman maikstok...@hotmail.com Date: Thursday, April 26, 2012 11:47 am Subject: [OSL | CCIE_Voice] dtmf-relay h.245-signal or H.245 Alphanumeric To: CCIE Voice ccie_voice@onlinestudylist.com Hi, In the SRND guide I read that the h.245-signal is the preferred dtmf method. But I think we need always H.245 Alphanumeric to have no issues with dtmf. True? Regards, Maik ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] UCCX Scripts End Step
You can insert a Goto step to direct the contact to the final and single End step. I would also put a Terminate step just before the End to free up the IVR port. Peter - Original Message - From: Ken Wyan kew...@gmail.com Date: Wednesday, April 25, 2012 11:26 am Subject: [OSL | CCIE_Voice] UCCX Scripts End Step To: ccie_voice@onlinestudylist.com Sometimes , our scripts send calls to an agent at the middle of the script. In that case should we include an End step right below call-contact step ? Is it recommended / not recommended to have multiple End steps in a single script? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] UCCX Scripts End Step
The default script is only initiated in some circumstances such as the script is not valid. If the main script works well the call will not get down there. - Original Message - From: Ken Wyan kew...@gmail.com Date: Wednesday, April 25, 2012 11:49 am Subject: Re: [OSL | CCIE_Voice] UCCX Scripts End Step To: wormh...@sch.hu Cc: ccie_voice@onlinestudylist.com But UCCX default scripts don't have terminate step ? reason? On Wed, Apr 25, 2012 at 3:03 PM, Farkas Péter wormh...@sch.bme.hu wrote: You can insert a Goto step to direct the contact to the final and single End step. I would also put a Terminate step just before the End to free up the IVR port. Peter - Original Message - From: Ken Wyan kew...@gmail.com Date: Wednesday, April 25, 2012 11:26 am Subject: [OSL | CCIE_Voice] UCCX Scripts End Step To: ccie_voice@onlinestudylist.com Sometimes , our scripts send calls to an agent at the middle of the script. In that case should we include an End step right below call-contact step ? Is it recommended / not recommended to have multiple End steps in a single script? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IOS NTP
You should use ntp master in only case if you don't have any external NTP source you can synch to. Also if your gateway has been already synched ex. by configuring the ntp server command than you can simply synch any other device to this gateway. In other words it is neither recommended or necessary the gateway to being a master. hth, Peter - Original Message - From: Maik Stokman maikstok...@hotmail.com Date: Wednesday, April 25, 2012 2:06 pm Subject: Re: [OSL | CCIE_Voice] IOS NTP To: Ken Wyan kew...@gmail.com Cc: ccie_voice@onlinestudylist.com I understand. But the calendar-update is needed for the clock summertime I think. Or is there no relation between them? Regards, Maik From: Ken Wyan Sent: Wednesday, April 25, 2012 11:43 AM To: Maik Stokman Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IOS NTP No you don't need. if you put ntp master , HQ Router won't synchronize with Backbone ntp server. I should have told more , but can't because of Cisco NDA. Thanks On Wed, Apr 25, 2012 at 1:54 PM, Maik Stokman maikstok...@hotmail.com wrote: Ken, What about Cisco Callmanager. Don’t we need ntp master for that? I see in the ipexpert study books that the command is used. I lost some point on the lab for the basic configuration (vlan/dhcp/ntp) I think is was for the ntp. But looks like I did nothing wrong there. That’s the reason I look for commands that cisco expects. I know that without the master command everything works fine. But is that good enough. From: Ken Wyan Sent: Wednesday, April 25, 2012 10:10 AM To: Maik Stokman Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IOS NTP don't use ntp master if this router needs to get time updates from another NTP server. In fact HQ Router don't need ntp master ntp source is needed for HQ to sync with a priority ntp server. It doesn't affect time-syncing of SB SC. On Wed, Apr 25, 2012 at 11:59 AM, Maik Stokman maikstok...@hotmail.com wrote: Hi, For configuring IOS ntp I have 3 questions: 1. Do I need the ntp master command? Which stratum is best practice? 2. Do I need the ntp update-calendar command? 3. Do I need the configure “ntp source loopback 0” when site b and c must use the HQ loopback interface as NTP server At this moment I use the following configuration: HQ conf t ntp x.x.x.x ntp master ntp calendar-update ntp source loopback 0 clock timezone PST -8 clock summer-time PST recurring SB ntp x.x.x.x clock time-zone CST -6 clock summer-time CST recurring SC ntp x.x.x.x clock time-zone HKT 8 clock summer-time HKT recurring Regards, Maik ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab Location Decission
...@gmail.com het volgende: I didn't have problem accessing notepad in local PC. But never try to use notepad via VNC to candidate PC it's slow.. I think , we can use Remote Desktop to connect to UCCX use notepad there , rather than struggling with Test PC (through VNC small display) On Wed, Apr 18, 2012 at 8:51 PM, Mathew Miller miller.mat...@gmail.comwrote: You can use notepad on the test PC but it is not enabled on the PC you are sitting at. So basically you have to use notepad through VNC which sucks. 2012/4/18 Farkas Péter wormh...@sch.bme.hu Notepad is not enabled by default at each location? Peter - Original Message - From: Mathew Miller miller.mat...@gmail.com Date: Wednesday, April 18, 2012 5:04 pm Subject: Re: [OSL | CCIE_Voice] Lab Location Decission To: Juan Carlos Anzola juancarlosanz...@gmail.com, Online Study ccie_voice@onlinestudylist.com I think it depends on how early you like to get up and how close you are to each. RTP Test starts at 7:10. You get a 20 minutes lunch in a conference room and it is catered in and are done by 3:45. SJ Test starts at 8:30. You get a 40-45 minute lunch in a cafeteria with lots of choices. You are done with the test about 5:05. Computers are about the same, but you get access to notepad on your computer in RTP but not in SJ. I felt like the proctor at RTP is more helpful. From: Juan Carlos Anzola juancarlosanz...@gmail.com Date: Wed, 18 Apr 2012 10:44:36 -0400 To: Online Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab Location Decission Hi Guys, I am scheduling my first attempt today. I have heard many myths and rumors about different locations. I am trying to decide between RTP and San Jose. Someone want to share te pros and cons? (In case they really exist) Regards, -- Juan Carlos Anzola ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Hardware Conferencing with different codecs
Sorry but you are wrong. HW CFB do xcoding to enable mixed mode conferences. What else should we enable multiple codecs under? Here is a reference to CUCM SRND: In a mixed-mode conference, the hardware conference bridge transcodes G.729 and G.723 streams into G.711 streams, mixes them, and then encodes the resulting stream into the appropriate stream type for transmission back to the user. Peter - Original Message - From: Mohammed Al Baqari baqari.voic...@gmail.com Date: Wednesday, April 18, 2012 1:02 pm Subject: Re: [OSL | CCIE_Voice] Hardware Conferencing with different codecs To: 'Julien Krieger' krieger.jul...@gmail.com, 'Ken Wyan' kew...@gmail.com Cc: ccie_voice@onlinestudylist.com Nop . You need to have a separate transcoder. CFB can't do transcoding. Regards, Mohammed Al Baqari From: ccie_voice-boun...@onlinestudylist.com [ On Behalf Of Julien Krieger Sent: Tuesday, April 17, 2012 11:11 AM To: Ken Wyan Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Hardware Conferencing with different codecs Hi, You're all good. Hardware conference bridge does the transcoding theirself. 2012/4/17 Ken Wyan kew...@gmail.com We configure HW conf bridge this way dspfarm profile 1 conference codec g711u codec g711a codec g729r8 codec g729ar8 codec g729br8 codec g729abr8 maximum sessions 4 associate application sccp no shut Is it possible for different phones (which use different codecs as per different region) to enter into a single conference using this conference bridge ? Does hardware conference bridge does internal transcoding by itself or do we need to configure another dspfarm profile for transcoding as well? Thank You ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab Location Decission
Notepad is not enabled by default at each location? Peter - Original Message - From: Mathew Miller miller.mat...@gmail.com Date: Wednesday, April 18, 2012 5:04 pm Subject: Re: [OSL | CCIE_Voice] Lab Location Decission To: Juan Carlos Anzola juancarlosanz...@gmail.com, Online Study ccie_voice@onlinestudylist.com I think it depends on how early you like to get up and how close you are to each. RTP Test starts at 7:10. You get a 20 minutes lunch in a conference room and it is catered in and are done by 3:45. SJ Test starts at 8:30. You get a 40-45 minute lunch in a cafeteria with lots of choices. You are done with the test about 5:05. Computers are about the same, but you get access to notepad on your computer in RTP but not in SJ. I felt like the proctor at RTP is more helpful. From: Juan Carlos Anzola juancarlosanz...@gmail.com Date: Wed, 18 Apr 2012 10:44:36 -0400 To: Online Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab Location Decission Hi Guys, I am scheduling my first attempt today. I have heard many myths and rumors about different locations. I am trying to decide between RTP and San Jose. Someone want to share te pros and cons? (In case they really exist) Regards, -- Juan Carlos Anzola ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] UCCX and Media Resources
- Have you checked by issuing debug sccp events on hq gw how SCCP works on during the call? - Also have you resetted all involved devices: xcoder, br phones, cti ports? - Hq xoder device is in the hq-xcoder MRG? Peter - Original Message - From: Chris devsin2...@gmail.com Date: Tuesday, April 17, 2012 12:02 pm Subject: Re: [OSL | CCIE_Voice] UCCX and Media Resources To: Gurpreet Singh Kukreja tycoononway1...@gmail.com Cc: ccie_voice@onlinestudylist.com Yes. This is all true. I had double checked before my original post. But I did it again after seeing your response. Best Regards Thanks Chris On Tue, Apr 17, 2012 at 7:27 PM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Hi Chris, I would verify the following: 1) Region setting between HQ and Branch sites uses G.729. 2) All your CTI Route Points should show in HQ DP on the CM and the CCX. 3) The Media Resource (Xcoder) should be configured on the HQ router. 4) The codec selected on the CCX is G.711. 5) Your IP phones show in the correct Device Pool. Last but not the least, make sure that the CTI Route point you dial should also be in the HQ DP with an Xcoder in the MRGL of HQ DP. Let me know if all the above stands true. Regards Gurpreet On Tue, Apr 17, 2012 at 4:17 AM, Chris devsin2...@gmail.com wrote: My UCCX is in HQ device pool. The DP has MRGL allocated to with registered transcoder resources. However, when I try to dial from BR1/BR2. The call fails to connect. The SDI traces on the call manager show following messages: *04/17/2012 15:28:30.231 CCM|MediaManager(9)::disconnOnResourceAllocationFailure, ERROR disconnOnResourceAllocationFailure - fails to allocate MTP/XCoder,connCount=2|CLID::StandAloneClusterNID::10.10IP::10.10.100.14DEV::UCCX_5701LVL::ErrorMASK::0800 * Xcoder resource is configured as *Transcoding Oper State: ACTIVE - Cause Code: NONE* *Active Call Manager: * *10.10.100**.12, Port Number: 2000* *TCP Link Status: CONNECTED, Profile Identifier: 1* *Reported Max Streams: 6, Reported Max OOS Streams: 0* *Supported Codec: g711ulaw, Maximum Packetization Period: 30* *Supported Codec: g711alaw, Maximum Packetization Period: 30* *Supported Codec: g729ar8, Maximum Packetization Period: 60* *Supported Codec: g729abr8, Maximum Packetization Period: 60* *Supported Codec: g729r8, Maximum Packetization Period: 60* *Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30* *Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30* *Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30* * * *MTP Oper State: ACTIVE - Cause Code: NONE* *Active Call Manager: * *10.10.100**.12, Port Number: 2000* *TCP Link Status: CONNECTED, Profile Identifier: 3* *Reported Max Streams: 20, Reported Max OOS Streams: 0* *Supported Codec: pass-thru, Maximum Packetization Period: N/A* *Supported Codec: g729r8, Maximum Packetization Period: 60* *Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30* *Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30* *Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30* *RSVP : ENABLED* MRGL [image: Inline image 1] Can someone tell me what am I doing wrong. Thanks Chris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] UCCX Editor Error
Have you tried to run the editor in windows compatibility mode? Peter - Original Message - From: Humayun Sami humayun_s...@hotmail.com Date: Thursday, April 5, 2012 8:16 am Subject: [OSL | CCIE_Voice] UCCX Editor Error To: ccie_voice@onlinestudylist.com Any one with the solution. I am not logging in with the without the access to the machine, I need to make a script. Logging in Anonymously. I get the attached error. Any ideas. Thanks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] the directories button display
You mean position flag as the the priority field?
Priority from low to high defines where in a list a Service should appear: 1 =
top of list, 50 (default) = middle,100 = bottom
Peter
- Original Message -
From: Rick Long rick.l...@ensi.com
Date: Wednesday, April 4, 2012 4:45 am
Subject: Re: [OSL | CCIE_Voice] the directories button display
To: Joe Martini joem...@cisco.com, chase mergenthal cm3_...@hotmail.com
Cc: ccie voice ccie_voice@onlinestudylist.com, kew...@gmail.com
kew...@gmail.com
Joe,
Your solution is confirmed.
Works great and is much easier and quick!
Also, I must add that I had to leave the external directories url blank on
the phone else it
shows host not found and the directories header displays..
I am running CUCM System version: 7.0.1.11000-2 not sure if that matters or
not.
I also confirmed that the order of the listings is determined by the
aforementioned position
flag in the insert statement.
Because Intercom is not listed as an Enterprise parameter it doesn't fill
its 4th position as
it should and Corp directory ends up being 5th instead of 6th as the flag
indicates.
However, if the intercom service is associated with a phone Corp Directory
is listed 6th as it
should be.
So in order to change the order, you must remove all the services except for
voicemail and
re-insert them back in and change the position value to change the order in
which they are
listed. The insert must be done using the run sql insert statement from a
SSH session to the
server. Inserting manually will not give the desired order and will cause
services to be listed
alphabetically.
Thank you so much for your input, it has been very helpful and appreciated.
Rick Long
From: Joe Martini [
Sent: Tuesday, April 03, 2012 12:59 PM
To: chase mergenthal
Cc: felipe_segn...@hotmail.com; kew...@gmail.com; ccie voice; Rick Long
Subject: Re: [OSL | CCIE_Voice] the directories button display
No, but you can configure on the phone(s) with external services provisioned
a Messages URL of
If you test this out though you'll see that the messages button acts a
little differently
than the Application:Cisco/Voicemail.
On Apr 3, 2012, at 12:35 PM, chase mergenthal wrote:
Would voice mail still work?
-Chase
--
If winners never quit and quitters never win, then who coined the phrase,
Quit while you're
still ahead.?
From: felipe_segn...@hotmail.com
To: kew...@gmail.com
Date: Tue, 3 Apr 2012 08:47:13 -0600
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] the directories button display
Hello All,
Basically if what you want to accomplish is restrict a user to have access
to the Directories
button and keep the order of the of the Missed Calls/Received Calls etc; I
think that we do not
even have to delete the URLs from the phone services.
What I think would work, is to use on this particular phone where you want
to restrict the
directories the service provisioning setting as External URL.
If you do this, the IP Phone will try to use the URL configured under
Enterprise parameters,
by default: If we leave this URL configured
and set the service provisioning to external, the directories will still
show up. So in order
to get the No service configured message, just delete the Directories URL
from enterprise
parameters (leave it blank).
By default, the other IP Phones, will still continue using Service
Provisioning Internal
which means they will get the URLs from their configuration file.
So, summary:
1. Set service provisioning to External URL to the phone where you want to
restrict
Directories access.
2. Leave the directories URL from enterprise parameter blank.
3. Reset the phone.
Reference:
HTH
Felipe Segnini.
Date: Tue, 3 Apr 2012 18:58:57 +0530
From: kew...@gmail.com
To: joem...@cisco.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] the directories button display
Hi,
Let's try to understand this sql command to find a way to change order.
We should run below to restore Corporate Directory.
run sql insert into telecasterservice
(pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
values('7eca2cf1-0c8d-4df4-a807-124b18fe89a4','Corporate
Directory','Corporate
Directory','Corporate
Directory','Application:Cisco/CorporateDirectory',1,'t',6)
Does anybody know about these parameters?
tkPhoneService default 1
Prioritydefault 6
On Tue, Apr 3, 2012 at 4:52 PM, Joe Martini joem...@cisco.com wrote:
If you use the restore sql statement you'll
Re: [OSL | CCIE_Voice] the directories button display
I always build and reference sql commands from the dev guide: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/datadict/7_0_1/DD_701.pdf Peter - Original Message - From: Ken Wyan kew...@gmail.com Date: Wednesday, April 4, 2012 11:21 am Subject: Re: [OSL | CCIE_Voice] the directories button display To: wormh...@sch.hu Cc: Rick Long rick.l...@ensi.com, Joe Martini joem...@cisco.com, chase mergenthal cm3_...@hotmail.com, ccie voice ccie_voice@onlinestudylist.com from which document you found this? On Wed, Apr 4, 2012 at 1:10 PM, Farkas Péter wormh...@sch.bme.hu wrote: You mean position flag as the the priority field? Priority from low to high defines where in a list a Service should appear: 1 = top of list, 50 (default) = middle,100 = bottom Peter - Original Message - From: Rick Long rick.l...@ensi.com Date: Wednesday, April 4, 2012 4:45 am Subject: Re: [OSL | CCIE_Voice] the directories button display To: Joe Martini joem...@cisco.com, chase mergenthal cm3_...@hotmail.com Cc: ccie voice ccie_voice@onlinestudylist.com, kew...@gmail.com kew...@gmail.com Joe, Your solution is confirmed. Works great and is much easier and quick! Also, I must add that I had to leave the external directories url blank on the phone else it shows host not found and the directories header displays.. I am running CUCM System version: 7.0.1.11000-2 not sure if that matters or not. I also confirmed that the order of the listings is determined by the aforementioned position flag in the insert statement. Because Intercom is not listed as an Enterprise parameter it doesn't fill its 4th position as it should and Corp directory ends up being 5th instead of 6th as the flag indicates. However, if the intercom service is associated with a phone Corp Directory is listed 6th as it should be. So in order to change the order, you must remove all the services except for voicemail and re-insert them back in and change the position value to change the order in which they are listed. The insert must be done using the run sql insert statement from a SSH session to the server. Inserting manually will not give the desired order and will cause services to be listed alphabetically. Thank you so much for your input, it has been very helpful and appreciated. Rick Long From: Joe Martini [ Sent: Tuesday, April 03, 2012 12:59 PM To: chase mergenthal Cc: felipe_segn...@hotmail.com; kew...@gmail.com; ccie voice; Rick Long Subject: Re: [OSL | CCIE_Voice] the directories button display No, but you can configure on the phone(s) with external services provisioned a Messages URL of If you test this out though you'll see that the messages button acts a little differently than the Application:Cisco/Voicemail. On Apr 3, 2012, at 12:35 PM, chase mergenthal wrote: Would voice mail still work? -Chase -- If winners never quit and quitters never win, then who coined the phrase, Quit while you're still ahead.? From: felipe_segn...@hotmail.com To: kew...@gmail.com Date: Tue, 3 Apr 2012 08:47:13 -0600 CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] the directories button display Hello All, Basically if what you want to accomplish is restrict a user to have access to the Directories button and keep the order of the of the Missed Calls/Received Calls etc; I think that we do not even have to delete the URLs from the phone services. What I think would work, is to use on this particular phone where you want to restrict the directories the service provisioning setting as External URL. If you do this, the IP Phone will try to use the URL configured under Enterprise parameters, by default: If we leave this URL configured and set the service provisioning to external, the directories will still show up. So in order to get the No service configured message, just delete the Directories URL from enterprise parameters (leave it blank). By default, the other IP Phones, will still continue using Service Provisioning Internal which means they will get the URLs from their configuration file. So, summary: 1. Set service provisioning to External URL to the phone where you want to restrict Directories access. 2. Leave the directories URL from enterprise parameter blank. 3. Reset the phone. Reference: HTH Felipe Segnini
Re: [OSL | CCIE_Voice] BACD across GK controlled clusters
You are required to put an xcoder at BR2. That is because BACD only supports G.711 but here GK is configured (as leaved the default codec in place) to use G.729. Peter - Original Message - From: Chris devsin2...@gmail.com Date: Tuesday, April 3, 2012 12:44 pm Subject: [OSL | CCIE_Voice] BACD across GK controlled clusters To: ccie_voice@onlinestudylist.com Hi All, BACD is configured on BR2. It works fine for calls from PSTN and local BR2. However, when I try to dial BACD pilot from HQ/BR1, I receive reorder tone. The problem I think is - BR2 and HQ/BR1 routing is through GK. Therefore if a call comes from BR1/HQ to 3500, the dial-peer used doesn't have ras enabled. Or is there another reason. Does anyone has solution for this? dial-peer voice 3500 voip service aa destination-pattern 3500 session target ipv4:10.10.110.3 incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 15000 voip destination-pattern [15]... session target ras incoming called-number . tech-prefix 1# dtmf-relay h245-alphanumeric no vad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUPS integration
CtiGw application user is required for MS OCS/Lynch integration not for CUPC. Peter - Original Message - From: Jurassic Labs jurassicl...@gmail.com Date: Saturday, March 31, 2012 3:29 pm Subject: Re: [OSL | CCIE_Voice] CUPS integration To: Chris devsin2...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com I would add that on each phone that you want presence, make sure to select the presence CSS and also set the userID for phone device and line DN. I have not entered the CtiGw application userID before. Without it, i've been able to do everything. I'll need to look at that a bit closer. On Saturday, March 31, 2012, Chris wrote: Hi All, I prepared list of steps for CUPS integration. Have I missed anything. - ON CUCM 1. Ensure you have users created 2. Create Application Server 3. Licensing - Assign capabilites 4. Change SIP non secure profile 5. Create SIP Trunk 6. Go to Service Parameter CallManager and select created above - CUP PUBLISH Trunk 7. Ensure you have AXL user creatd 8. Create Application User - CtiGw with CTI enabled and Cti access to all users 9. Create Application User - PhoneMessenger with CTI enabled and Cti access to all users On CUPS 1. Complete Post install setup 2. Go to Servicebility Section on CUPS and Enable all services 3. Go to CUP SIP Proxy service parameter and set the Domain Name. At this stage you should see users populated. Go From Left to Right 1. Systems 1.1 Security - Enter ACLs input and output 2. Presence 2.1 Settings -Confirm The SIP Trunk 2.2 Gateways - Enter CUCM , description and IP Address 3. Application 3.1 Cisco Unified Personal Comunicator - CUPC 3.1.1settings - Enter IP Address of the PUB and the SUB 3.1.2Voicemail Server - Add New - choose unity connection and enter all details 3.1.3Voicemail Profile - Add new - enter name, select voicemail pilot, and primary voicemail server. Make this profiledefault by ticking the checkbox. Also assign all users to it. 3.1.4 Deskphone control Settings - Set Status to ON and enter the CtiGw passwords 3.1.5 PI Phone Messanger Settings - Set Status to ON and enter the CtiGw passwords On CUC 1. Go to relevant CoS and Check the tick box for - 1.1 - Allow Users to Access Voice Mail Using an IMAP Client 1.2 - Allow Users to Use Unified Client to Access Voice MailLog on from CUPC and check/test. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] DTMF Tones from CME to UC
Try to add: ! dial-peer voice 100 voip dtmf-interworking rtp-nte ! Peter - Original Message - From: Jason Murray murr...@usa.com Date: Friday, March 23, 2012 4:48 am Subject: Re: [OSL | CCIE_Voice] DTMF Tones from CME to UC To: Shirley, Kris C. kcshir...@tmhs.org, ccie_voice@onlinestudylist.com I don't think there is a setting for that in UC. Between CUCM and UC its a SCCP integration. But between UCME and UC its a SIP integration. But there is no trunk you have to create in UC. You just create the Phone System, Port Group and Ports. Didnt see anything about MTP though. Jason - Original Message - From: Shirley, Kris C. Sent: 03/22/12 07:44 PM To: Jason Murray, ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] DTMF Tones from CME to UC Do you have MTP required checked on the SIP Trunk in UC? Kris From: ccie_voice-boun...@onlinestudylist.com [ *On Behalf Of *Jason Murray *Sent:* Thursday, March 22, 2012 6:11 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] DTMF Tones from CME to UC So I am running through lab6.2 from the vol2 workbooks and seem to run into a little snag. I have integrated CME to UC using sip as it has you do. So I can leave messages and MWI works just fine. But if I try calling in from the br2 user and try to enter a pin the digits arent coming across. To test to make sure it recognizes the user I check the box in the user settings to bypass pin if recognized and I go straight into the box. So for some reason the dtmf tones arent getting sent correctly. Here is my config for the CME. If someone could give it a once over and see if you can see anything thats out of the ordinary and prevent the digits from working right. Thanks 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 Vlan400 bind media source-interface Vlan400 registrar server voice register global mode cme source-address 10.10.202.1 port 5060 max-dn 10 max-pool 2 authenticate register timezone 21 date-format Y-M-D hold-alert mwi reg-e164 voicemail 3600 create profile sync 0008581038415029 ntp-server 10.10.100.2 mode directedbroadcast ! voice register dn 1 number 3001 call-forward b2bua busy 3600 call-forward b2bua noan 3600 timeout 12 mwi ! voice register dn 2 number 3002 call-forward b2bua busy 3600 call-forward b2bua noan 3600 timeout 12 mwi ! voice register dn 5 number 3015 call-forward b2bua all 3600 mwi ! voice register template 1 softkeys idle Newcall Redial softkeys connected Confrn Endcall Hold ! voice register pool 1 id mac 001E.7A25.4329 type 7961 number 1 dn 1 template 1 cor incoming css-ld 1 3001 dtmf-relay rtp-nte username 3001 password cisco description 5623001 codec g711ulaw ! voice register pool 2 id mac 001C.581C.48B6 type 7961 number 1 dn 2 number 2 dn 5 dtmf-relay rtp-nte username 3002 password cisco description 5623002 codec g711ulaw ! voice hunt-group 1 parallel final 3015 list 3001,3002 timeout 16 pilot 3000 ! dial-peer voice 15 voip translation-profile outgoing ANI destination-pattern [15]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric no vad ! dial-peer voice 1 pots incoming called-number . direct-inward-dial ! dial-peer voice 5000 pots preference 1 destination-pattern 5... port 0/0/0:15 prefix 12123945 ! dial-peer voice 1000 pots preference 1 destination-pattern 1... port 0/0/0:15 prefix 16178631 ! dial-peer voice 112 pots destination-pattern 9%112 port 0/0/0:15 forward-digits 3 dial-peer voice 7 pots destination-pattern 9[4-9].. port 0/0/0:15 forward-digits 7 ! dial-peer voice 900 pots corlist outgoing pt-int destination-pattern 900T port 0/0/0:15 prefix 00 ! dial-peer voice 1212 voip translation-profile outgoing ANI destination-pattern 9001212394 session target ras tech-prefix 1#5 dtmf-relay h245-alphanumeric no vad ! dial-peer voice 12121 pots translation-profile outgoing ANI preference 1 destination-pattern 9001212394 port 0/0/0:15 prefix 001212394 ! dial-peer voice 3700 voip destination-pattern 370. session protocol sipv2 session target ipv4:10.10.202.2 dtmf-relay sip-notify no vad ! dial-peer voice 100 voip translation-profile incoming GK incoming called-number . dtmf-relay h245-alphanumeric no vad dial-peer voice 3600 voip max-conn 1 destination-pattern 3600 session protocol sipv2
Re: [OSL | CCIE_Voice] DTMF Tones from CME to UC
/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Reason: Q.850;cause=16 Date: Fri, 23 Mar 2012 14:08:14 GMT From: 3002 sip:3002@10.10.202.1;tag=001c581c48b60015721b9ab6-ef2a7901 Content-Length: 0 To: sip:3600@10.10.202.1;tag=2D1998-1266 Call-ID: 001c581c-48b60011-549fa838-a908c34b@10.10.202.51 Via: SIP/2.0/UDP 10.10.202.51:5060;branch=z9hG4bKad780522 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 BYE Mar 23 14:08:14.390: //-1//SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:3600@10.10.210.13:5060;transport=tcp SIP/2.0 Reason: Q.850;cause=16 Date: Fri, 23 Mar 2012 14:08:06 GMT From: 3002 sip:3002@10.10.202.1;tag=2D197C-14AD Timestamp: 1332511694 Content-Length: 0 User-Agent: Cisco-SIPGateway/IOS-12.x To: sip:3600@10.10.210.13;tag=ed7cb8457d534cf4a2233373e335271c Call-ID: 726F0885-742811E1-805F9D83-EBD77B10@10.10.202.1 Via: SIP/2.0/TCP 10.10.202.1:5060;branch=z9hG4bK2A2707 CSeq: 102 BYE Max-Forwards: 70 Mar 23 14:08:14.398: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK From: 3002 sip:3002@10.10.202.1;tag=2D197C-14AD To: sip:3600@10.10.210.13;tag=ed7cb8457d534cf4a2233373e335271c Via: SIP/2.0/TCP 10.10.202.1:5060;branch=z9hG4bK2A2707 Call-ID: 726F0885-742811E1-805F9D83-EBD77B10@10.10.202.1 CSeq: 102 BYE Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,SUBSCRIBE Content-Length: 0 - Original Message - From: Farkas Péter Sent: 03/23/12 03:50 AM To: Jason Murray Subject: Re: [OSL | CCIE_Voice] DTMF Tones from CME to UC Try to add: ! dial-peer voice 100 voip dtmf-interworking rtp-nte ! Peter - Original Message - From: Jason Murray murr...@usa.com Date: Friday, March 23, 2012 4:48 am Subject: Re: [OSL | CCIE_Voice] DTMF Tones from CME to UC To: Shirley, Kris C. kcshir...@tmhs.org, ccie_voice@onlinestudylist.com I don't think there is a setting for that in UC. Between CUCM and UC its a SCCP integration. But between UCME and UC its a SIP integration. But there is no trunk you have to create in UC. You just create the Phone System, Port Group and Ports. Didnt see anything about MTP though. Jason - Original Message - From: Shirley, Kris C. Sent: 03/22/12 07:44 PM To: Jason Murray, ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] DTMF Tones from CME to UC Do you have MTP required checked on the SIP Trunk in UC? Kris From: ccie_voice-boun...@onlinestudylist.com [ *On Behalf Of *Jason Murray *Sent:* Thursday, March 22, 2012 6:11 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] DTMF Tones from CME to UC So I am running through lab6.2 from the vol2 workbooks and seem to run into a little snag. I have integrated CME to UC using sip as it has you do. So I can leave messages and MWI works just fine. But if I try calling in from the br2 user and try to enter a pin the digits arent coming across. To test to make sure it recognizes the user I check the box in the user settings to bypass pin if recognized and I go straight into the box. So for some reason the dtmf tones arent getting sent correctly. Here is my config for the CME. If someone could give it a once over and see if you can see anything thats out of the ordinary and prevent the digits from working right. Thanksvoice 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 Vlan400 bind media source-interface Vlan400 registrar server voice register global mode cme source-address 10.10.202.1 port 5060 max-dn 10 max-pool 2 authenticate register timezone 21 date-format Y-M-D hold-alert mwi reg-e164 voicemail 3600 create profile sync 0008581038415029 ntp-server 10.10.100.2 mode directedbroadcast ! voice register dn 1 number 3001 call-forward b2bua busy 3600 call-forward b2bua noan 3600 timeout 12 mwi ! voice register dn 2 number 3002 call-forward b2bua busy 3600 call-forward b2bua noan 3600 timeout 12 mwi ! voice register dn 5 number 3015 call-forward b2bua all 3600 mwi ! voice register template 1 softkeys idle Newcall Redial softkeys connected Confrn Endcall Hold ! voice register pool 1 id mac 001E.7A25.4329 type 7961 number 1 dn 1 template 1 cor incoming css-ld 1 3001 dtmf-relay rtp-nte username 3001 password cisco description 5623001 codec g711ulaw ! voice register pool 2 id mac 001C.581C.48B6 type 7961 number 1 dn 2 number 2 dn 5 dtmf-relay rtp-nte username 3002 password cisco description 5623002 codec g711ulaw ! voice hunt-group 1 parallel final 3015 list 3001,3002 timeout 16 pilot 3000 ! dial-peer voice 15 voip translation-profile outgoing ANI destination-pattern [15
Re: [OSL | CCIE_Voice] CUPC Signalling
CUPC retrives voice messages via IMAP. Peter - Original Message - From: Juan Lopez lopez.hernandez.j...@gmail.com Date: Thursday, March 22, 2012 9:17 am Subject: Re: [OSL | CCIE_Voice] CUPC Signalling To: Ken Wyan kew...@gmail.com Cc: ccie_voice@onlinestudylist.com not sure as just starting with this, but I believe the CUPC, when logging in, will download the VM profile setup in CUPS, and then use a direct connection to CUC over HTTPS to access the VM/MWI - whether it uses SIP (softphone mode) or CTI (deskphone mode) to talk to CUCM. Op 22 maart 2012 07:33 schreef Ken Wyan kew...@gmail.com het volgende: Take typical integration of CUCM to Unity Connection sccp integration CUPC client is used to check voicemail / mwi CUPC Client --sip signalling- CUPS Server CUCM Server --sccp signalling- Unity Connection CUPC Client --sip signalling- CUCM Server CUCM Server -sip signalling- CUPS Server CUPC Client - ? signalling Unity Connection When CUPC client access voice mail mwi indications , does it use SCCP signalling ? OR does CUCM acts as a signalling proxy between CUPC client Unity connection server for sccp/sip translation? Ken ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUC integration - AXL API
Have you added UC server in CUCM under System/Application Server? Peter - Original Message - From: Chris devsin2...@gmail.com Date: Tuesday, March 20, 2012 11:54 am Subject: [OSL | CCIE_Voice] CUC integration - AXL API To: ccie_voice@onlinestudylist.com I am stuck in trying to CUC to Import users through AXL API access. When I click Import Users it gives No AXL Remote or LDAP Servers were found. A Unified Communications Manager and/or LDAP Directory server integration is required to synchronize users. message. The application user has Standard AXL API access role assigned. The I have tried following: 1. Delete and Add phone system, port group ( ports BTW are registered with UCM) 2. Change user name to administrator under AXL Server Settings in Phone System' Edit to no avail. 3. Delete and Add of the servers under Port Group Edit. 4. Change port numbers of the servers under Port Group Edit, based on www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080874c9b.shtml. The default ports it gives are 2000/2443. One clue/symptom I have is - when I click Ping, in Edit Server configuration page, I get timeout - for AXL as well as tftp server. Similar integration using AXL application user with Presence and CCX worked fine. Best Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Scoring 100% in High Availability
Could we assume that we may loose points by doing some extra? Peter - Original Message - From: AJ BG ciscoie2...@gmail.com Date: Monday, March 12, 2012 1:14 am Subject: [OSL | CCIE_Voice] Scoring 100% in High Availability To: ccie_voice@onlinestudylist.com I have concerns about High Availability section in the lab. 1. Has anyone scored 100% in High Availability? 2. Is there anything beside SRST and BACD that is graded in this section? I have not been able to score 100% in HA yet. My configurations consist of the standard SRST settings as well as the following settings. · Call routing requirements, · Voicemail requirements, · Date and time requirement, · Music on hold, · And additional specific requirements which are specifically requested. At the end of my lab I test the requirements to make sure things are working as expected. and ephone-dns are not in invalid state. And Yet so far I have got very low scores in HA. I was once told that we should not assume requirements in CCIE LAB exams. But I suppose I am either doing something extra, or I am missing some sort of requirments that are not specifcally mentioned . I don’t know what am I missing? Those who are strong in HA, please advice what else should I watch for? you are also welcome to send me a sample configuration. Thanks, AJ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] UC SIP integration question
Hi, Creating a CUCM-UC SIP integration do we need to configure SIP authentication and registration under port group configuration page of UC? It seems to be working w/o but DSG for W2Lab7 fills these items, as well. Peter ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] DSP`s for CFB
Is it really a PVDM2? Next gen PVDM3 do share. Peter - Original Message - From: Rynard Coetzee rynard.coet...@bytes.co.za Date: Wednesday, March 7, 2012 4:09 pm Subject: [OSL | CCIE_Voice] DSP`s for CFB To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi All I have a question about DSP`s and Conference Bridges ,from what i`ve read a CFB on IOS can not share a DSP with signalling channels ,but I have a Cisco 2911 router that I use for my lab that has only one PVDM2-16 in it which I use for the Signalling channels on my PRI and I have configured a CFB on the router ,and it works ! This is confusing me ,or am I maybe misreading this. Regards Rynard ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] B-ACD
You can download the whole bacd as a .zip from CCO. It includes the audio, too. Peter - Original Message - From: Emanuel Damasceno aedamasc...@gmail.com Date: Tuesday, March 6, 2012 2:40 pm Subject: [OSL | CCIE_Voice] B-ACD To: ccie_voice@onlinestudylist.com Hello Experts, I am wondering here about b-acd. I don't remember reaching that far on my first attempt, so I am considering myself totally new in this. So, what I wanted to know is this: - When I go to the support page on Cisco, I find the B-ACD examples, but these examples point me to a file on flash. I don't have that file, so can anybody share it? - Ok, I don't have the file for B-ACD, however, when I go on my home router, I see this: *SiteB(config-app)#service ? dsapp ipsla-responder clid_authen clid_col_npw_npw AFW_THIRD_PARTY_CC CALLIndSs_SErviCe Default RetrProxy CTAPP clid_authen_col_npw fax_hop_on ipsla-testcall app-b-acd-aa clid_authen_npw session app-b-acd clid_authen_collect clid_col_npw_3 WORD Name of the service/package SiteB(config-app)#service* I am using IOS (C2800NM-ADVENTERPRISEK9_IVS_LI-M), Version 12.4(24)T4, RELEASE SOFTWARE (fc2) (c2800nm-adventerprisek9_ivs_li-mz.124-24.T4.bin). My question is: We have a few prompts being referenced, where are those prompts? Thanks *Emanuel Damasceno* CCNP Voice ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cBarge in CME SRST
What srst provision in your scenario: none/dn/all? - Original Message - From: Rynard Coetzee rynard.coet...@bytes.co.za Date: Thursday, February 23, 2012 7:00 am Subject: RE: [OSL | CCIE_Voice] cBarge in CME SRST To: wormh...@sch.hu wormh...@sch.hu Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Yes I have tried it under the template and under the ephone. -Original Message- From: Farkas Péter [ Sent: 22 February 2012 02:58 PM To: Rynard Coetzee Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] cBarge in CME SRST Have you tried to turn off privacy and enable remote-in-use sofktkey through an ephone-template attached to the ephone? Privacy setting on ephone has a bug in SRST mode. Peter - Original Message - From: Rynard Coetzee rynard.coet...@bytes.co.za Date: Wednesday, February 22, 2012 1:51 pm Subject: [OSL | CCIE_Voice] cBarge in CME SRST To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi All I have an issue to get the cBarge to work when my H323 GW goes into SRST ,the shared line shows up on both phones ,but when I have an active call on one phone ,I don`t see the number on the other phone ,and the other phone does not go into remote in use state when I press the shared line button. I have privacy turned off under the ephones and also under the telephony service. Also my CFB is registered to the router when in srst mode ,I am able to make a normal ad-hoc conference when in srst mode. Any ideas ? Regards Rynard ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cBarge in CME SRST
Have you tried to turn off privacy and enable remote-in-use sofktkey through an ephone-template attached to the ephone? Privacy setting on ephone has a bug in SRST mode. Peter - Original Message - From: Rynard Coetzee rynard.coet...@bytes.co.za Date: Wednesday, February 22, 2012 1:51 pm Subject: [OSL | CCIE_Voice] cBarge in CME SRST To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi All I have an issue to get the cBarge to work when my H323 GW goes into SRST ,the shared line shows up on both phones ,but when I have an active call on one phone ,I don`t see the number on the other phone ,and the other phone does not go into remote in use state when I press the shared line button. I have privacy turned off under the ephones and also under the telephony service. Also my CFB is registered to the router when in srst mode ,I am able to make a normal ad-hoc conference when in srst mode. Any ideas ? Regards Rynard ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] VXML on MGCP Gateways
I am not familiar with VXML but you may do it as configuring the gateway being also as a H323 gw and send the call back to it from CUCM. You set up a dial-peer for incoming h323 call and enable bacd on it. Also if you have CU or CUC than it can do it for you. Peter - Original Message - From: Ricardo ricardoareval...@gmail.com Date: Wednesday, February 22, 2012 3:51 pm Subject: [OSL | CCIE_Voice] VXML on MGCP Gateways To: ccie_voice@onlinestudylist.com Hi Guys, I am doing a lab thinking out of the box... what if we want to set a b-acd tcl-like but for a gateway running MGCP, so a message is played before sending to receptionist or . I know TCL doesn't work with MGCP, but VXML does... do you have any idea how to set it up? best regards //r.a. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Embedded BACD Prompts and DropThrough
Also can we configure multiple AAs using the embeded script itself? It seems to me not since usage of 'service app-b-acd-aa' command does not allow us to determine a different service name like flash based BACD 'service aa flash://', where 'aa' is name of the service. Peter - Original Message - From: datucha123 datucha123 datucha...@gmail.com Date: Sunday, February 19, 2012 11:53 am Subject: Re: [OSL | CCIE_Voice] Embedded BACD Prompts and DropThrough To: AJ BG ciscoie2...@gmail.com Cc: ccie_voice@onlinestudylist.com 1. Are prompts also embedded in the IOS? Or do they need to be copied in the router’s flash? No, the Prompts are not embedded in the IOS, you need to manually add them into Flash. 2. Does drop through mode work with embedded BACD? Yes, embedded BACD works for Drop Through Mode very well. You can find the configuration examples here: On Sun, Feb 19, 2012 at 7:59 AM, AJ BG ciscoie2...@gmail.com wrote: Two questions about embedded BACD. 1. Are prompts also embedded in the IOS? Or do they need to be copied in the router’s flash? 2. Does drop through mode work with embedded BACD? Does anyone have a working copy of embedded BACD configuration? Thanks AJ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa)
It seems clocks are unsynched. Please try clock source line on HQ E1 0/3/0. Peter - Original Message - From: muhammad nouman nouman_n...@yahoo.com Date: Tuesday, February 14, 2012 3:26 pm Subject: Re: [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa) To: Emanuel Damasceno aedamasc...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, michael.se...@compucom.com michael.se...@compucom.com Yes, I have used isdn protocol-emulate network command, please find here both side config PSTN-FRS# network-clock-participate wic 2 isdn switch-type primary-net5 controller E1 0/2/0 clock source internal pri-group timeslots 1-3,16 interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn protocol-emulate network isdn incoming-voice voice no cdp enable HQ-Router# network-clock-participate wic 3 isdn switch-type primary-net5 controller E1 0/3/0 clock source internal pri-group timeslots 1-3,16 interface Serial0/3/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn protocol-emulate network isdn incoming-voice voice no cdp enable Thanks Nomi From: Emanuel Damasceno aedamasc...@gmail.com To: muhammad nouman nouman_n...@yahoo.com Cc: michael.se...@compucom.com michael.se...@compucom.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Wednesday, 15 February 2012 12:47 AM Subject: Re: [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa) Did you use the command: isdn protocol-emulate network on your Serial interfaces for T1/E1 on the PSTN side? Can you send your config? Emanuel Damasceno CCNP Voice On Tue, Feb 14, 2012 at 11:22 AM, muhammad nouman nouman_n...@yahoo.com wrote: Hi All, I have connected both E1 back to back with PRI crossover but I am getting following error on both side. Please let me know is this normal or it will create problem. HQ-Router#sh controllers e1 E1 0/3/0 is up. Applique type is Channelized E1 - balanced No alarms detected. alarm-trigger is not set Version info Firmware: 20090113, FPGA: 20, spm_count = 0 Framing is CRC4, Line Code is HDB3, Clock Source is Internal. Data in current interval (202 seconds elapsed): 0 Line Code Violations, 0 Path Code Violations 3 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 3 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Total Data (last 2 15 minute intervals): 3 Line Code Violations, 5 Path Code Violations, 30 Slip Secs, 0 Fr Loss Secs, 3 Line Err Secs, 0 Degraded Mins, 32 Errored Secs, 1 Bursty Err Secs, 0 Severely Err Secs, 0 Unavai HQ-Router# PSTN-FRS#sh controllers e1 E1 0/2/0 is up. Applique type is Channelized E1 - balanced No alarms detected. alarm-trigger is not set Version info Firmware: 20090113, FPGA: 20, spm_count = 0 Framing is CRC4, Line Code is HDB3, Clock Source is Internal. Data in current interval (315 seconds elapsed): 0 Line Code Violations, 0 Path Code Violations 4 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 4 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs Total Data (last 2 15 minute intervals): 3 Line Code Violations, 3 Path Code Violations, 31 Slip Secs, 0 Fr Loss Secs, 2 Line Err Secs, 0 Degraded Mins, 33 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 2 Unavail Secs I am also not able to register this E1 with MGCP, please help me Thanks Nomi From: michael.se...@compucom.com michael.se...@compucom.com To: ccie_voice@onlinestudylist.com Sent: Saturday, 31 December 2011 5:40 AM Subject: [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa) I read in the thread below I am using simple crossover cable (Ethernet crossover). This cable will not work. You need a T1 cross over cable: !!! If your using an Ethernet cross over it won't work need T1 cross over 1--4, 2--5, 4--1, 5--2 Usually you can pick one up at local computer store or ebay real cheap depending on the length. Hope this helps. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a
Re: [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa)
Also what show network-clocks sais? U may need to set E1 to be the source instead of backplane. Issue network-clock select command. Peter - Original Message - From: Kevin Spicer ke...@kevinspicer.co.uk Date: Tuesday, February 14, 2012 5:00 pm Subject: Re: [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa) To: muhammad nouman nouman_n...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, michael.se...@compucom.com michael.se...@compucom.com Hi, You have used protocol-emulate network on both sides. This should only be on one side. On 14 Feb 2012 14:26, muhammad nouman nouman_n...@yahoo.com wrote: Yes, I have used isdn protocol-emulate network command, please find here both side config PSTN-FRS# network-clock-participate wic 2 isdn switch-type primary-net5 controller E1 0/2/0 clock source internal pri-group timeslots 1-3,16 interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn protocol-emulate network isdn incoming-voice voice no cdp enable HQ-Router# network-clock-participate wic 3 isdn switch-type primary-net5 controller E1 0/3/0 clock source internal pri-group timeslots 1-3,16 interface Serial0/3/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn protocol-emulate network isdn incoming-voice voice no cdp enable Thanks Nomi *From:* Emanuel Damasceno aedamasc...@gmail.com *To:* muhammad nouman nouman_n...@yahoo.com *Cc:* michael.se...@compucom.com michael.se...@compucom.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Wednesday, 15 February 2012 12:47 AM *Subject:* Re: [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa) Did you use the command: isdn protocol-emulate network on your Serial interfaces for T1/E1 on the PSTN side? Can you send your config? *Emanuel Damasceno* CCNP Voice On Tue, Feb 14, 2012 at 11:22 AM, muhammad nouman nouman_n...@yahoo.comwrote: Hi All, I have connected both E1 back to back with PRI crossover but I am getting following error on both side. Please let me know is this normal or it will create problem. HQ-Router#sh controllers e1 E1 0/3/0 is up. Applique type is Channelized E1 - balanced No alarms detected. alarm-trigger is not set Version info Firmware: 20090113, FPGA: 20, spm_count = 0 Framing is CRC4, Line Code is HDB3, Clock Source is Internal. Data in current interval (202 seconds elapsed): 0 Line Code Violations, 0 Path Code Violations 3 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 3 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Total Data (last 2 15 minute intervals): 3 Line Code Violations, 5 Path Code Violations, 30 Slip Secs, 0 Fr Loss Secs, 3 Line Err Secs, 0 Degraded Mins, 32 Errored Secs, 1 Bursty Err Secs, 0 Severely Err Secs, 0 Unavai HQ-Router# PSTN-FRS#sh controllers e1 E1 0/2/0 is up. Applique type is Channelized E1 - balanced No alarms detected. alarm-trigger is not set Version info Firmware: 20090113, FPGA: 20, spm_count = 0 Framing is CRC4, Line Code is HDB3, Clock Source is Internal. Data in current interval (315 seconds elapsed): 0 Line Code Violations, 0 Path Code Violations 4 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 4 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs Total Data (last 2 15 minute intervals): 3 Line Code Violations, 3 Path Code Violations, 31 Slip Secs, 0 Fr Loss Secs, 2 Line Err Secs, 0 Degraded Mins, 33 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 2 Unavail Secs I am also not able to register this E1 with MGCP, please help me Thanks Nomi *From:* michael.se...@compucom.com michael.se...@compucom.com *To:* ccie_voice@onlinestudylist.com *Sent:* Saturday, 31 December 2011 5:40 AM *Subject:* [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa) I read in the thread below I am using simple crossover cable (Ethernet crossover). This cable will not work. You need a T1 cross over cable: !!! If your using an Ethernet cross over it won't work need T1 cross over 1--4, 2--5, 4--1, 5--2 Usually you can pick one up at local computer store or ebay real cheap depending on the length. Hope this helps. Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or
[OSL | CCIE_Voice] mlpp vs frf.12
Gents, Qos in wb2/6.2 requires the most efficient lfi technique. SG selected MLPP. Why? What is the main advantage one to the other? Thanks, Peter ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Application server in UCM 7
I have found this forum with more details: https://supportforums.cisco.com/thread/1002696 Peter - Original Message - From: Gurpreet Singh Kukreja tycoononway1...@gmail.com Date: Sunday, January 29, 2012 11:44 pm Subject: Re: [OSL | CCIE_Voice] Application server in UCM 7 To: donny f f.faraday...@gmail.com Cc: ccie_voice@onlinestudylist.com You can use the Application Server windows in Cisco Unified Communications Manager Administration to maintain associations between the Cisco Unified Communications Manager and off-cluster, external applications, such as Cisco Unity Connection and Cisco Unified Presence, and to synchronize Cisco Unified Communications Manager systems and other applications. HTH Gurpreet On Sun, Jan 29, 2012 at 2:25 PM, donny f f.faraday...@gmail.com wrote: hi all, anyone know why we need setup Application Server in UCM for Unity Connection ? tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] QoS question from new 5 labs, lab 2, question 10.1
Think the same, since 3rd threshold performs tail drop. Peter - Original Message - From: John McGaughey (jomcgaug) jomcg...@cisco.com Date: Friday, January 27, 2012 2:56 pm Subject: [OSL | CCIE_Voice] QoS question from new 5 labs, lab 2, question 10.1 To: ccie_voice@onlinestudylist.com Hello, From the new 5 labs, Lab 2 question 10.1 it asks the following. For traffic being sent to the Site A gateway ensure that traffic marked with COS 5 is dropped if queue 1 is 75% full. The solution guide says to add queue-set 2 to the fastethernet port and change the following 2 line like so. mls qos queue-set output 2 threshold 1 75 100 100 100 mls qos srr-queue output cos-map queue 1 threshold 3 5 the 2nd line looks like a typo. It should be the following for putting COS 5 into q1t1, correct? mls qos srr-queue output cos-map queue 1 threshold 1 5 John ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] QoS question from new 5 labs, lab 2, question 10.1
I meant my opinion the same as your. - Original Message - From: John McGaughey (jomcgaug) jomcg...@cisco.com Date: Friday, January 27, 2012 3:05 pm Subject: RE: [OSL | CCIE_Voice] QoS question from new 5 labs, lab 2,question 10.1 To: wormh...@sch.hu Cc: ccie_voice@onlinestudylist.com 3rd threshold is 100% implicit. So that wouldn't work since it will drop when 100% full. -Original Message- From: Farkas Péter [ Sent: Friday, January 27, 2012 8:04 AM To: John McGaughey (jomcgaug) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QoS question from new 5 labs, lab 2,question 10.1 Think the same, since 3rd threshold performs tail drop. Peter - Original Message - From: John McGaughey (jomcgaug) jomcg...@cisco.com Date: Friday, January 27, 2012 2:56 pm Subject: [OSL | CCIE_Voice] QoS question from new 5 labs, lab 2, question 10.1 To: ccie_voice@onlinestudylist.com Hello, From the new 5 labs, Lab 2 question 10.1 it asks the following. For traffic being sent to the Site A gateway ensure that traffic marked with COS 5 is dropped if queue 1 is 75% full. The solution guide says to add queue-set 2 to the fastethernet port and change the following 2 line like so. mls qos queue-set output 2 threshold 1 75 100 100 100 mls qos srr-queue output cos-map queue 1 threshold 3 5 the 2nd line looks like a typo. It should be the following for putting COS 5 into q1t1, correct? mls qos srr-queue output cos-map queue 1 threshold 1 5 John ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voicemail access during AAR
- Can AAR CSS assigned to HQ DN/Device reach Voicemail Pilot over PSTN? GW AAR CSS is used when there is an incoming call from PSTN to BR IP phone and the phone has call forwarding set to voicemail. Also VM is in other CAC location that has insufficient bandwidth which triggers AAR function. Call is terminated on gw and will do forwarding instead of IP Phone at the branch. Here gw's AAR CSS will route the call toward the Voicemail's pilot. Peter - Original Message - From: Vega Wong vega2...@yahoo.com.au Date: Tuesday, January 17, 2012 2:01 pm Subject: [OSL | CCIE_Voice] Voicemail access during AAR To: ccie_voice@onlinestudylist.com Hi experts I am working on Vol 2. Lab 7 and try to fully understand the topic of voice mail access during AAR, but I am struggling to have a clear picture. Here is the setup HQ GW - MGCP BR GW - H323 HQ Phones, and BR Phones are both SCCP. The line on both phone been assigned to the same AAR group. Only HQ phone assigned with AAR group and CSS AAR on the device level. voicemail for BR is on CUE which is integrated with CUCM using CTI route point. All integration is configured and tested ok AAR between the phones are working, the HQ Phone will reroute out to PSTN to reach BR Phones when there is not enough bandwidth. Also, HQ Phone can directly dial in to the voicemail pilot (reroute out to PSTN) and reach the CUE log in. BR phone can press the message button and reach the sign-in prompt for CUE (only asking for PIN) However, when HQ phone calls BR phone and BR Phone doesnt pick up, just as it should transfer to voicemail, I get fast busy tone on HQ Phone. I am trying to understand, at this instance, I am still using the AAR CSS on HQ Phone to reach the voice mail pilot right? I imagine if HQ Phone can successfully call to voicemail directly during AAR, it shouldnt be different when it is transferred by BR Phone? Also, I am trying to understand when do we need to assign AAR group and AAR CSS to the gateway? and why? Please help Thanks in advance Vega ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] incoming called-number .
Good to know: When the Cisco IOS router or gateway receives a call setup request, a dial peer match is made for the incoming call in order to facilitate routing the call to different session applications. This is not a digit-by-digit match, rather the full digit string received in the setup request is used to match against configured dial peers. So I vote dp 2. Peter - Original Message - From: Ken Wyan kew...@gmail.com Date: Friday, January 13, 2012 3:10 pm Subject: Re: [OSL | CCIE_Voice] incoming called-number . To: Emanuel Damasceno aedamasc...@gmail.com Cc: ccie_voice@onlinestudylist.com If we had 2 incoming dial-peers with same incoming called-number , I think selection should be random. Anybody found a cisco document for this? On Fri, Jan 13, 2012 at 7:05 PM, Emanuel Damasceno aedamasc...@gmail.comwrote: Right. =) *Emanuel Damasceno* CCNP Voice On Fri, Jan 13, 2012 at 10:41 AM, datucha123 datucha123 datucha...@gmail.com wrote: First of all Preference command works only for Outgoing calls. It does not make any sense for incoming dial-peer matching. also in that particular case, the preference command will not make any sense, because those dial-peers are having a differenct destination patterns. On Fri, Jan 13, 2012 at 4:31 PM, Emanuel Damasceno aedamasc...@gmail.com wrote: Wouldn't the command preference X work in that situation? *Emanuel Damasceno* CCNP Voice On Fri, Jan 13, 2012 at 6:36 AM, datucha123 datucha123 datucha...@gmail.com wrote: For incoming calls, the 23548 is more specific match for Called Number of 235482345 then 235. And that is why the Dial-peer 2 is matched. For outgoing calls, if you place a call to the same number (235482345) and the destination patterns are the same (235 and 23548) then the dial-peer 2 would be matched as again, it is more specific match and if the Enblock is used. If you will use Overlap Dialing, (UCME for instance) then dial-peer 1 would be matched On Fri, Jan 13, 2012 at 10:58 AM, Ken Wyan kew...@gmail.com wrote: Is there a cisco doc for incoming called-number selection order? What I mean is say dial-peer voice 1 incoming called-number 235 xxx dial-peer voice 2 incoming called-number 23548 xx If a call arrives with DNIS 235482345 which incoming dial-peer will be matched? As per my testing it seems dial-peer 2 is selected. If this was an outgoing dial-peer (with default dial-peer hunt 0) , dial-peer 1 should be the match this is well documented. For incoming calls ?? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Preference within policy-map
Here it is confirmed: http://www.cisco.com/en/US/tech/tk543/tk757/technologies_tech_note09186a0080160fc1.shtml Peter - Original Message - From: datucha123 datucha123 datucha...@gmail.com Date: Friday, January 13, 2012 4:25 pm Subject: Re: [OSL | CCIE_Voice] Preference within policy-map To: Ken Wyan kew...@gmail.com Cc: ccie_voice@onlinestudylist.com, OSL Routing and Switching ccie...@onlinestudylist.com That is a great question. Based on my knowledge, the Set DSCP command is executed first, because otherwise the exceeding traffic will become EF. You can also refer to this Linke, where the Auto QoS is duscussed: On Fri, Jan 13, 2012 at 5:47 PM, Ken Wyan kew...@gmail.com wrote: When I want to mark all voip packets to ef mark exceeding packets to 8 I do following in a Catalyst Switch. policy-map AutoQos-Policy-Untrust class AutoQos-VoIP-RTP-Untrust police 128 exceed-action policed-dscp-transmit set dscp ef mls qos map policed-dscp 46 to 8 Here will the set dscp ef command will be executed before police command? Otherwise exceeding packets (which are remarked to 8 by police statement ) can again marked with dscp ef. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SIP MWI issues with CUE
SIP stack on CUE has to be given IP address of binded SIP IP address of voice gateway. Try to correct on CUE: ! ccn subsystem sip gateway address 10.10.110.3 ! It may require reset of CUE, as well. Peter - Original Message - From: Rajasekar Shanmugam rajaseka...@gmail.com Date: Monday, January 2, 2012 4:29 pm Subject: [OSL | CCIE_Voice] SIP MWI issues with CUE To: ccie_voice@onlinestudylist.com Experts - I`m running into some issues with the CUE MWI , when working with SRST. I have the required configs using the unsolicited notify for the MWI. Attached my configs the ccsip debug output. Not sure , wher I`m going wrong. Please help. -- Raj ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Getting CUPS/CUCM restart/reload reason
We are curious to know the reason, not the command how to reboot. Peter - Original Message - From: Cisco Nut rafayc...@gmail.com Date: Thursday, November 10, 2011 3:44 pm Subject: Re: [OSL | CCIE_Voice] Getting CUPS/CUCM restart/reload reason To: brajesh kumaR brjku...@gmail.com Cc: CCIE-V邮件列表 ccie_voice@onlinestudylist.com utils system restart On Thu, Nov 10, 2011 at 6:15 AM, brajesh kumaR brjku...@gmail.com wrote: Hello , Is there any way to know using CLI for server restart reason for CUPS/CUCM. Regards, Brajesh. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] can cuc demo license run vpim
No, demo license not cover VPIM so it requries VPIM license to be added. However proctorlabs should have. Peter - Original Message - From: bruno bruno.juni...@gmail.com Date: Wednesday, November 9, 2011 11:18 am Subject: [OSL | CCIE_Voice] can cuc demo license run vpim To: CCIE-V邮件列表 ccie_voice@onlinestudylist.com When I attempt to add a VPIM location is Unity Connection I receive the following license error. Anyone attempt VPIM in these labs yet? Status The requested operation would result in a license violation. Unable to create VPIM Location -- Best Regards, Bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Triggering +
From IP phones you cannot dial the + only CUPC and CIPC can support. Peter - Original Message - From: Emanuel Damasceno aedamasc...@gmail.com Date: Thursday, October 27, 2011 4:41 pm Subject: [OSL | CCIE_Voice] Triggering + To: ccie_voice@onlinestudylist.com Hello Experts, I am on Lab 5A and I am now wondering how I trigger the + on the phones. I was told that 7940s don't support + dial, but I don't want to know that, I want to know what I need to press so it becomes a +. On my cell phone, I press 0 and hold it. Is it the same with Cisco Phones? Thanks *Antonio Emanuel Damasceno* CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written) CompTIA Network+ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] [UCCX COMPONENT ACTIVATION]
System/Control Center there is a link right upper side: Component Activation. Peter - Original Message - From: michael.se...@compucom.com Date: Monday, October 10, 2011 8:57 am Subject: [OSL | CCIE_Voice] [UCCX COMPONENT ACTIVATION] To: ccie_voice@onlinestudylist.com Where can I check in UCCX to determine what components are activated? Thanks --ms ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] AAR Issue
I think you mean BR1 Phone1's AAR Dest Mask is +1617863, right? HQ gateway is registered and working properly in all other cases but AAR? HQ Phone1's AAR CSS consists the +.! Route Pattern that should point to HQ GW? Peter - Original Message - From: Satoshi Funabashi (sfunabas) sfuna...@cisco.com Date: Thursday, August 11, 2011 9:33 am Subject: [OSL | CCIE_Voice] AAR Issue To: OSL Group ccie_voice@onlinestudylist.com Hi, I'm testing AAR. When I call from HQ Phone1(5001) to BR1 Phone1(1001), it seems that AAR is invoked, but call is not routed through PSTN. Does anyone know how to resolve this issue? Following is my troubleshooting. - On HQ Phone1, Network Congestion. Rerouting. message was shown. - I could not see any output with debug isdn q931 on HQ Router. - I could not see any output with debug mgcp packets on HQ Router. （HQ Router uses MGCP) Following is my CUCM Configuration. - Automated Alternate Routing Enable service parameter is set to True. - AAR Group AAR_UCM is configured and its Prefix Digits is blank. - HQ Phone 1 belongs to AAR_UCM - BR1 Phone 1 belongs to AAR_UCM - HQ Phone 1's AAR Calling Search Space is unrestricted. - DN 5001's AAR Destination Mask is +1617863 - Route Pattern which is used with AAR is ¥+! - Location of HQ Router and HQ Phone1 is Hub_None - Location of BR1 Phone1 is LOC_BR1 If you need more information, please let me know. Thanks and Regards, Satoshi Satoshi Funabashi Systems Engineer Cisco Systems G.K. Tel:81-3-6434-2824(direct) 81-3-6434-6500(group) 81-90-4050-1574(mobile) E-mail: sfuna...@cisco.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST - max buttons on phone
You can use 'limit-dn 7965 2' command under call-manager-fallback mode. Peter - Original Message - From: Victor Malyuga victor_maly...@yahoo.com Date: Friday, July 29, 2011 11:15 am Subject: [OSL | CCIE_Voice] SRST - max buttons on phone To: ccie_voice@onlinestudylist.com Is there a way to limit the number of buttons supported on the phone in SRST mode? For instance, I have 7965 with 6 buttons configured but want only 2 of them to be available in SRST mode. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] 2651xm and MGCP gateway - help required
- try to reset the MGCP stack on gateway by issuing 'no mgcp' than 'mgcp' commands - if no change, reset the Cisco RIS Data Collector under Network Services at Servicability page of CUCM. This will synchronize the admin page to internal database. Peter - Original Message - From: steven moran smoran...@gmail.com Date: Wednesday, July 20, 2011 9:24 am Subject: [OSL | CCIE_Voice] 2651xm and MGCP gateway - help required To: ccie_voice@onlinestudylist.com Dear all, I posted this a couple of days ago and have tried all of the suggestions and many more from trawling the websites and still I cannot pinpoint the issue. In brief, this is part of the IPExpets vol1 ex 4 I have a vwic-2mft-e1 in my 2651 running (C2600-ADVENTERPRISEK9_IVS-M), Version 12.4(15)T14, RELEASE SOFTWARE (fc2) sh ccm shows that the MGCP gateway is registered, however it does not show as registered in the CUCM 7.02 interface sh isdn stat - shows no problem sh controller E1 shows no recent issues for line code, path code or slips sh network-clocks shows the E1 0/1 as priority 1 the E1 clock source is set to line (with the PSTN-WAN switch set to internal) Does anyone know how to configure a 2651xm for MGCP using E1 card The ccie voice that sold me the equipment assured me all would work - (Unfortunately for me he is now travelling in remote parts and uncontactable). I have his config and mine is the same, however I do not have the CUCM 7 that he used - are there any advanced parameters in CUCM 7 needed to make this work? Best regards, Steve ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA hairpin in MGCP
Internal phone can dial 911 via HQ gw? - Original Message - From: donny f f.faraday...@gmail.com Date: Thursday, July 14, 2011 2:49 am Subject: Re: [OSL | CCIE_Voice] MVA hairpin in MGCP To: Brian Mulgrew btmulg...@gmail.com Cc: ccie_voice@onlinestudylist.com here is my debug isdn q931 , after calling MVA 2999, press 1 and dial 911 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Facility i = 0x9F8B0100A10F02010106072A8648CE1500040A0100 Protocol Profile = Networking Extensions 0xA10F02010106072A8648CE1500040A0100 Component = Invoke component Invoke Id = 1 Operation = InformationFollowing (calling_name) Name information in subsequent FACILITY message Called Party Number i = 0x81, '911' Plan:ISDN, Type:Unknown Redirecting Number i = 0x01008A, '911' Plan:ISDN, Type:Unknown *Jul 14 03:52:58.687: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8005 Channel ID i = 0xA98383 Exclusive, Channel 3 *Jul 14 03:52:58.695: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8 callref = 0x8005 Progress Ind i = 0x8188 - In-band info or appro HQ(config)#priate now available *Jul 14 03:52:58.739: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x0005 Cause i = 0x82AF - Resource unavailable, unspecified *Jul 14 03:52:58.747: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x8005 *Jul 14 03:52:58.791: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x0005 *Jul 14 03:52:58.815: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x808D Cause i = 0x8290 - Normal call clearing On Wed, Jul 13, 2011 at 2:37 PM, Brian Mulgrew btmulg...@gmail.com wrote: Hi - ensure your 911 call is going out the same gateway (rdp device and line css) and ensure the rdp/h323/mgcp gateway are all set to g711u only hth On Wed, Jul 13, 2011 at 7:52 PM, donny f f.faraday...@gmail.com wrote: hi all, I config Hq router as MGCP hairpinDN :2999 and MVA 2999 - CAll fro PSTN to 2999, MVA work and also I am able to dial any UCM extension, just the call to PSTN like : 911 , local call (it rings , btu after few second just release/blink) What I have missed here ( I have reset UCM and router) here is my dial-peer in HQ : application service cmm ! ! Voice serv voip Allow connection h232 to h323 ! dial-peer voice 1234567 voip description Trigger to VXML --- RP 2999 from UCM service cmm session target ipv4:10.10.210.10 incoming called-number 2999 codec g711ulaw no vad dial-peer voice 5010 voip description TO Make a call --Internal and PSTN out destination-pattern 2999 session target ipv4:10.10.210.10 dtmf-relay h245-alphanumeric codec g711ulaw no vad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CME EM Host not found
Gents, I am facing an EM issue on CME cannot step over. EM is configured like: voice logout-profile 1 number 3002 type normal ! voice user-profile 1 max-idle-time 10 user br2phn3 password adgjm number 3005 type normal speed-dial 1 3006 ! ephone 2 privacy off device-security-mode none mac-address 0005.9A3C.7800 ephone-template 2 type CIPC logout-profile 1 ! telephony-service authentication credential secretname psswrd url services http://10.10.202.1/CMEserverForPhone/serviceurl url authentication http://10.10.202.1/CCMCIP/authenticate.asp secretname psswrd ! When I press the Services button on CIPC only Host not found error message apperars on the phone. So I cannot reach EM menu at all. I can see phone tries to reach CME: Jul 8 14:07:07.436: Fri, 08 Jul 2011 14:07:07 GMT 192.168.208.1 /CMEserverForPhone/serviceurl ok Protocol = HTTP/1.1 Method = GET Query = locale=English_United_Statesname=SEP00059A3C7800 Any thought to resolve? Peter ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] issue with DID number..
What deb isdn q931 sais when you dial this number from PSTN? Peter - Original Message - From: Rashid Khan me_rashid...@yahoo.com Date: Thursday, July 7, 2011 10:56 am Subject: [OSL | CCIE_Voice] issue with DID number.. To: ccie voice ccie_voice@onlinestudylist.com Hi Friends. I am facing an issue with Direct Inward Dialing.. We have an Pri number provided by service provider. also we have purchased 200 DID numbers... All other numbers working fine. but only one DID number is having some issue.. I also raise a ticket at service provider... but they close that ticket by saying, Problem is at your end. Can anyone please help me in this regard.. Regards Rashid ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] issue with DID number..
*Jul 7 10:02:17.597: ISDN Se0/0/0:15 Q931: RX - DISCONNECT pd = 8 callref = 0 xB666 Cause i = 0x8281 - Unallocated/unassigned number Debug states gateway unable to route the call, and vendor is correct u do receive the call. Check if call is correctly routed toward phone: dial-peers, gw incoming css, etc. Peter - Original Message - From: Rashid Khan me_rashid...@yahoo.com Date: Thursday, July 7, 2011 11:57 am Subject: Re: [OSL | CCIE_Voice] issue with DID number.. To: wormh...@sch.hu Cc: ccie voice ccie_voice@onlinestudylist.com I Hope this will help. debug isdn q931 is ON. *Jul 7 10:02:17.201: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 i s 0x0 0x0, Calling num 2360 *Jul 7 10:02:17.205: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x3665 call ID = 0xB289 switch = primary-net5 interface = User *Jul 7 10:02:17.205: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8 callref = 0x3665 Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98393 Exclusive, Channel 19 Calling Party Number i = 0x0081, '2360' My IP Phone ext.. Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '35202154' DID number im trying to call Plan:Unknown, Type:Unknown *Jul 7 10:02:17.233: ISDN Se0/0/0:15 Q931: RX - SETUP_ACK pd = 8 callref = 0x B665 Channel ID i = 0xA98393 Exclusive, Channel 19 *Jul 7 10:02:17.233: ISDN Se0/0/0:15 Q931: RX - CALL_PROC pd = 8 callref = 0x B665 Progress Ind i = 0x8288 - In-band info or appropriate now available *Jul 7 10:02:17.257: ISDN Se0/0/0:15 Q931: RX - SETUP pd = 8 callref = 0x0271 Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA1838B Preferred, Channel 11 Calling Party Number i = 0x2183, '2135205493' Pri Number Plan:ISDN, Type:National Called Party Number i = 0xA1, '2135202154' DID Number Plan:ISDN, Type:National *Jul 7 10:02:17.257: ISDN Se0/0/0:15 Q931: Received SETUP callref = 0x8271 cal lID = 0x2DC2 switch = primary-net5 interface = User *Jul 7 10:02:17.273: ISDN Se0/0/0:15 Q931: TX - CALL_PROC pd = 8 callref = 0x 8271 Channel ID i = 0xA9838B Exclusive, Channel 11 *Jul 7 10:02:17.297: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 i s 0x2 0x1, Calling num 2135205493 *Jul 7 10:02:17.297: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x3666 call ID = 0xB28A switch = primary-net5 interface = User *Jul 7 10:02:17.297: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8 callref = 0x3666 *Jul 7 10:02:17.325: ISDN Se0/0/0:15 Q931: RX - CALL_PROC pd = 8 callref = 0x B666 Progress Ind i = 0x8288 - In-band info or appropriate now available *Jul 7 10:02:17.345: ISDN Se0/0/0:15 Q931: TX - PROGRESS pd = 8 callref = 0x8 271 Progress Ind i = 0x8188 - In-band info or appropriate now available *Jul 7 10:02:17.373: ISDN Se0/0/0:15 Q931: RX - ALERTING pd = 8 callref = 0xB 665 Progress Ind i = 0x8188 - In-band info or appropriate now available Progress Ind i = 0x8188 - In-band info or appropriate now available *Jul 7 10:02:17.525: %ISDN-6-CONNECT: Interface Serial0/0/0:22 is now connected to 03002318643 N/A *Jul 7 10:02:17.597: ISDN Se0/0/0:15 Q931: RX - DISCONNECT pd = 8 callref = 0 xB666 Cause i = 0x8281 - Unallocated/unassigned number Progress Ind i = 0x8288 - In-band info or appropriate now available *Jul 7 10:02:17.597: ISDN Se0/0/0:15 Q931: call_disc: PI received in disconnect ; Postpone sending RELEASE for callid 0xB28A *Jul 7 10:02:18.385: ISDN Se0/0/0:15 Q931: RX - DISCONNECT pd = 8 callref = 0 xB65F Cause i = 0x8290 - Normal call clearing Progress Ind i = 0x8288 - In-band info or appropriate now available ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] sip trunk delay offer
You are right, I thought early offer when it is the case. - Original Message - From: Adam Frankel (afrankel) afran...@cisco.com Date: Wednesday, June 22, 2011 12:22 am Subject: Re: [OSL | CCIE_Voice] sip trunk delay offer To: wormh...@sch.hu, Farkas Péter wormh...@sch.bme.hu Cc: donny f f.faraday...@gmail.com, ccie_voice@onlinestudylist.com This is false. If region configuration is set to G729 and the calling device supports G729, CUCM can select G729 in its answer, if it is offered. Adam Original Message-- From: Farkas Péter wormh...@sch.bme.hu Sent: Tue, Jun 21, 2011 3:32:21 Am To: donny f f.faraday...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] sip trunk delay offer Have you introduce MTP to the call? By default CUCM only capable outgoing delay offer based on G.711. Peter - Original Message - From: donny ff.faraday...@gmail.com Date: Tuesday, June 21, 2011 1:09 am Subject: [OSL | CCIE_Voice] sip trunk delay offer To: ccie_voice@onlinestudylist.com hi, anybody know why i did not see the G729 in debug ? it only said PCMU in codec v=0 o=CiscoSystemsSIP-GW-UserAgent 6036 5234 IN IP4 10.20.100.2 s=SIP Call c=IN IP4 10.20.100.2 t=0 0 m=audio 16522 RTP/AVP 0 c=IN IP4 10.20.100.2 a=rtpmap:0 PCMU/8000- codec a=ptime:20 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] how to find b-acd on cisco documentation
under CME config guides: http://www.cisco.com/en/US/customer/docs/voice_ip_comm/cucme/bacd/configuration/guide/cme40tcl.html Peter - Original Message - From: Chris Green voice5...@yahoo.com Date: Wednesday, June 22, 2011 12:56 pm Subject: [OSL | CCIE_Voice] how to find b-acd on cisco documentation To: ccie_voice@onlinestudylist.com Hi All Can you guid me to find b-acd on cisco documentation. Is any one has the steps link. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] sip trunk delay offer
Have you introduce MTP to the call? By default CUCM only capable outgoing delay offer based on G.711. Peter - Original Message - From: donny f f.faraday...@gmail.com Date: Tuesday, June 21, 2011 1:09 am Subject: [OSL | CCIE_Voice] sip trunk delay offer To: ccie_voice@onlinestudylist.com hi, anybody know why i did not see the G729 in debug ? it only said PCMU in codec v=0 o=CiscoSystemsSIP-GW-UserAgent 6036 5234 IN IP4 10.20.100.2 s=SIP Call c=IN IP4 10.20.100.2 t=0 0 m=audio 16522 RTP/AVP 0 c=IN IP4 10.20.100.2 a=rtpmap:0 PCMU/8000- codec a=ptime:20 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] digit manipulate on VOICE-PORT or on POT
Manipulating on voice-affects all incoming calls on that port meanwhile doing on pots dial-peer only modify calls that match this peer. If not asked otherwise better to modify incoming pstn calls on voice-port. Peter - Original Message - From: Chris Green voice5...@yahoo.com Date: Wednesday, June 8, 2011 12:46 pm Subject: [OSL | CCIE_Voice] digit manipulate on VOICE-PORT or on POT To: ccie_voice@onlinestudylist.com Hi All, What is the difference manipulating the digit on VOICE-PORT or POT as follows? Which one expected for the exam and why? - voice translation-rule 1 rule 1 /^.*\(5...\)/ /\1/ voice translation-profile pstn-in translate called 1 voice-port 0/1/0:23 translation-profile incoming pstn-in dial-peer voice 1 pots incoming called-number . direct-inward-dial --- voice translation-rule 1 rule 1 /^.*\(5...\)/ /\1/ voice translation-profile pstn-in translate called 1 dial-peer voice 1 pots translation-profile incoming pstn-in incoming called-number . direct-inward-dial port 0/1/0:23 --- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] mva partial match does not work
Gents, I have a H323 gw only for PSTN and RD is 10 digits. PSTN also sends this ten digits. If complete match is configured MVA recognizes RD and asks only for PIN. However if I set partial match with any number of digit MVA always requires RD to be given. I have already restarted both CM and MVA services. Could you advice how to resolve and why partial match does not work but complete does? Peter ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] transcoder for voicemail ports
put in VM's mrgl since the region that does not support low bitrate codec will invoke the xcoder, in your case it's the VM's. Peter - Original Message - From: Michael Luo hout...@gmail.com Date: Friday, June 3, 2011 7:05 am Subject: [OSL | CCIE_Voice] transcoder for voicemail ports To: ccie_voice@onlinestudylist.com I'm testing with Unity Connection 7.0. Let say if the voicemail ports are in region A. IP phones and voice gateway are in region B. Cross-region codec was G.729. IP phones can call VM pilot just fine (with G.729). However, if PSTN calls IP phone and rollover (CFNA/CFB) to voicemail, I got fast busy (reorder tone). By looking at CCM trace, it seems to be codec issue: 06/02/2011 23:07:43.042 CCM|MediaResourceManager::sendAllocationResourceErr - ERROR - no transcoder device configured. My question is: if I configure transcoder to fix this problem, shall the transcoder be in the voicemail's mgrl or the voice gateway's mrgl? Thanks! Michael ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] mva partial match does not work
Here is the CSCsy60115 bug from the list: http://www.onlinestudylist.com/archives/ccie_voice/2009-December/065021.html Peter - Original Message - From: Bartosz Sokolowski ibartosz.sokolow...@gmail.com Date: Friday, June 3, 2011 12:07 pm Subject: Re: [OSL | CCIE_Voice] mva partial match does not work To: ccie_voice@onlinestudylist.com Hi, Partial match is very buggy in CM 7.0 and it doesn't work in general :) Search list archive for this matter. It was discussed something like a month ago. There were even CCM traces provided showing it's a bug. -- Regards, Bartosz 2011/6/3 Farkas Péter wormh...@sch.bme.hu Gents, I have a H323 gw only for PSTN and RD is 10 digits. PSTN also sends this ten digits. If complete match is configured MVA recognizes RD and asks only for PIN. However if I set partial match with any number of digit MVA always requires RD to be given. I have already restarted both CM and MVA services. Could you advice how to resolve and why partial match does not work but complete does? Peter ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SW Version for Lab
On lab CUCME 7.0 is required since Ben confirmed. Peter - Original Message - From: George Goglidze gogli...@gmail.com Date: Friday, April 29, 2011 11:16 am Subject: Re: [OSL | CCIE_Voice] SW Version for Lab To: Abel ... midga...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com In CUCME study for 7.1 Sent from my iPad On 28 Apr 2011, at 22:43, Abel ... midga...@gmail.com wrote: Hi everyone, the following list is the recommended software to be use on lab, is ok use the same version under the major release or must be use the higher version under minor release for v7.x of each one? Any major software release which has been generally available for six months is eligible for testing in the CCIE Voice Lab Exam. Cisco Unified Communications Manager 7.0 Cisco Unified Communications Manager Express 7.0 Cisco Unified Contact Center Express 7.0 Cisco Unified Presence 7.0 Cisco Unity Connection 7.0 Thanks Abel Mateo CCIE R/S 28546 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] TCS Capability Exchange
It is rather a timer expiration since either endpoint will send TCS. Peter - Original Message - From: George Goglidze gogli...@gmail.com Date: Sunday, April 17, 2011 6:35 pm Subject: Re: [OSL | CCIE_Voice] TCS Capability Exchange To: Naoufal Kerboute naou...@mhdinfotech.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com I don't have a call manager available right now, but if you keep searching for h245 it said something about capability failure if I remember correctly. Attach the trace file if you have it, I'll have a look. Sent from my iPad On 17 Apr 2011, at 05:54, Naoufal Kerboute naou...@mhdinfotech.com wrote: Hi guys, I’m working on CUBE, and I’m facing the TCS issue, I know that I have to uncheck “wait for Far End H.245 Terminal Capability Set”, but I’m looking how to identify this in the SDL traces. Anyone know the exact word who describe the issue in the logs? Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ 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] why is vad bad?
Also can source voice quality issues like hissing. Peter - Original Message - From: matt...@ciscovoiceguru.com matt...@ciscovoiceguru.com Date: Wednesday, February 9, 2011 5:16 pm Subject: Re: [OSL | CCIE_Voice] why is vad bad? To: Stutz, Bernhard st...@pandacom.de Cc: ccie_voice@onlinestudylist.com I've always understood that VAD results in a higher CPU utilization. For a site of 10 phones running a 2921 it wouldn't be an issue. However, if you're running several hundred (or thousand) users running off the same pool of devices then you'd run into a significant impact on CPU performance. Matthew Berry, CCIE #26721 Email: matt...@ciscovoiceguru.com Twitter: Blog: On Feb 9, 2011, at 9:38 AM, Stutz, Bernhard wrote: Hi, i am just wondering why vad is bad and we all learn as a rule of thumb to disable vad on all voip dial peers? When you have a look for what vad has been designed for it looks to me as a valuable algorithm ( Whats the reason we disable it all the time? Is Cisco not able to support vad correctly or is it user experience that they want to hear a noise otherwise they think of that the connection has been lost? But therefore you have comfort-noise isn’t it? Kindly regards, Bernhard ___ 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] Voicemail button in SRST mode
by voicemail command under SRST config. Peter - Original Message - From: Shrini linuxbos...@gmail.com Date: Wednesday, January 12, 2011 11:50 am Subject: [OSL | CCIE_Voice] Voicemail button in SRST mode To: ccie_voice@onlinestudylist.com Hi Experts, How can I get the Vmail speeddial button on phone worked in SRST mode. When I press the button, I get fast busy tone but if I dial Voicemail number I am able to connect via a pots dial-peer. Thanks Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call forward to UC failed
VM Pilot should have a CSS that can reach the hunt pilot for VM ports. try add PT-UC-VMPILOT to CSS-UC-VMPORT Peter - Original Message - From: ShinGei Yong shingei.y...@gmail.com Date: Wednesday, December 8, 2010 12:02 pm Subject: Re: [OSL | CCIE_Voice] Call forward to UC failed To: Miron Kobelski findko...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi Kobel, Not sure i'm understand your question correctly. But i believe the VM Pilot information already stated, it used CSS-UC-VMPORT (PT-VMPORT,INT-DN) as below. Shingei On Wed, Dec 8, 2010 at 6:48 PM, Miron Kobelski findko...@gmail.com wrote: OK, but it is not assigned to the VM pilot? 2010/12/8 ShinGei Yong shingei.y...@gmail.com Hi Kobel, i believed is already stated there, the CSS-INT-DN which comprised ( INT-DN and UC-VMPILOT) as well Shingei On Wed, Dec 8, 2010 at 6:42 PM, Miron Kobelski findko...@gmail.comwrote: It seems you don't have the PT-UC-VMPILOT partition in the CSS used for forwarding - this might be your issue. regards kobel On Wed, Dec 8, 2010 at 11:39, ShinGei Yong shingei.y...@gmail.comwrote: Hi Kobel, Below is the setup: 5600/PT-UC-VMPILOT 5601/PT-UC-VMPORT 5602/PT-UC-VMPORT 5001/PT-INT-DN (Phone B) 5002/PT-INT-DN (Phone A) CSS-INT-DN (PT-INT-DN,PT-UC-VMPILOT) CSS-UC-VMPORT (PT-VMPORT,PT-INT-DN) Voicemail Pilot Information Pilot Number: 5600 CSS: CSS-UC-VMPORT Phone A Line Setting: CFA (Checked) CSS-INT-DN CFB (Checked) CSS-INT-DN CFNA (Checked)CSS-INT-DN Both phone A B able to dial VM directly without issue. Anything goes wrong? Shingei On Wed, Dec 8, 2010 at 5:59 PM, Miron Kobelski findko...@gmail.comwrote: Hi, If you entered forward destination as a number, you need to also configure proper forwarding CSS. if you configured forwarding for the line using the VM checkbox, you also need to have proper CSS in VM pilot. HTH kobel 2010/12/8 ShinGei Yong shingei.y...@gmail.com Hi, Yes, both the cfna and cfb has already checked on line setting, and CSS that contain the internal DN and VM pilot did assigned also. It seem to be a CSS issue but can't figure out the cause. I tried monitor the UC port status, there is no any call answering. So the call just stuck some where in UCM Shingei On Wed, Dec 8, 2010 at 5:19 PM, mark.f.bunch mark.f.bu...@gmail.comwrote: Have you also configured the Calling Search Space on the line for CFA and CFNA? On 08/12/2010, at 8:12 PM, Stanislav Braichuk stanislav.braic...@gmail.com wrote: Are you set checkbox on cfna and cfb in line configuration? 2010 12 8 10:44 пользователь ShinGei Yong shingei.y...@gmail.com shingei.y...@gmail.com написал: Hi, I'm facing an issue which is either cfna or cfb failed on phone A when caller B call to caller A. Both the phone A and B able to access to their voicemail box in UC by pressing the voicemail button and enter correct pin. Both the phone is able to dial the Pilot number directly without issue. UC is integrate with UCM with SCCP. Am i miss out any setting? Shingei ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ 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] Conference by Select/Join (SIP)
I've tested with 7965 and works so it should be a limitation of CIPC. - Original Message - From: Peter Farkas To: ccie list Sent: Monday, May 31, 2010 3:27 PM Subject: [OSL | CCIE_Voice] Conference by Select/Join (SIP) Conference is supported at HQ site but CIPC(SIP) IP phone cannot use Select/Join softkey to bulid up a conference. It failes with Unavailable Feature message on the display. However Confrn softkey works as expected. CIPC or SIP supports buliding a conference by Select/Join method, at all? -- ___ 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] cannot dial from MVA
Gents,
I have an issue with MVA. MVA collects PIN and I press 1 to dial but it does
not proceed with any call instead the well known prompt sounds: The call
cannot be completed... Even if the called number is local and placed in the
None partition.
This prompt suggests CSS issue however as Vik advised before I created a
totally new CSS just for RDP but it does not solve the problem.
Service Parameters: Complete Match and RDP+Line CSS.
I have read near all the thread regarding MVA here, but the issue remains. I
attached the vxml debug.
Any suggestion?HQ-RTR#sh run
Building configuration...
Current configuration : 5107 bytes
!
! Last configuration change at 18:15:20 UTC Tue May 18 2010
! NVRAM config last updated at 18:15:21 UTC Tue May 18 2010
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
no service password-recovery
!
hostname HQ-RTR
!
boot-start-marker
boot system flash c2800nm-ipvoice_ivs-mz.124-20.T2.bin
warm-reboot
boot-end-marker
!
logging message-counter syslog
logging buffered 51200 warnings
!
no aaa new-model
memory-size iomem 20
network-clock-participate wic 0
network-clock-participate wic 1
network-clock-select 1 E1 0/1/0
!
no ip source-route
!
!
ip cef
!
!
no ip domain lookup
ip multicast-routing
multilink bundle-name authenticated
!
isdn switch-type primary-ni
!
!
!
voice service voip
allow-connections h323 to sip
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
/dial-p
filtering...
dial-peer voice 15 voip
voice-class codec 1
incoming called-number .
!
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
!
dial-peer voice 911 pots
destination-pattern 911
clid strip name
no digit-strip
port 0/1/0:15
!
dial-peer voice 7 pots
destination-pattern 9[2-9]..
port 0/1/0:15
forward-digits 7
!
dial-peer voice 11 pots
destination-pattern 91[2-9]..[2-9]..
port 0/1/0:15
forward-digits 11
!
dial-peer voice 5000 voip
destination-pattern 5...$
voice-class codec 1
voice-class h323 1
session target ipv4:10.10.210.11
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 5001 voip
preference 1
destination-pattern 5...$
voice-class codec 1
voice-class h323 1
session target ipv4:10.10.210.10
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 5999 pots
service cmm
incoming called-number 5999
!
!
!
gatekeeper
zone local US ipexpert.com
zone local Spain ipexpert.com
zone remote PSTN-WAN ipexpert.com 10.10.100.2 1719
zone prefix Spain 34*
zone prefix PSTN-WAN 91*
HQ-RTR#un all
All possible debugging has been turned off
HQ-RTR#deb voip appl vxml all
vxml all debugging is on
HQ-RTR#
May 18 18:57:48.209: //44//VXML:/Open_SetupIndication:
May 18 18:57:48.213: //44//AFW_:/vapp_vxmldialog: Trusted=0, DNIS Map URI=,
Code = {
}
May 18 18:57:48.213: //44//AFW_:/vapp_vxmldialog: After DNIS Map
URI=http://10.10.210.10/ccmivr/pages/IVRMainpage.vxml, Code = {
?xml version=1.0 encoding=iso-8859-1?
vxml version=2.0
form id=
}
May 18 18:57:48.213: //-1//AFW_:/AFW_VxmlModule_New:
May 18 18:57:48.213: //-1//VXM
HQ-RTR#L:/vxml_tree_lock:
vxmlp=47E35714 usage_cnt=0
May 18 18:57:48.217: //0/15E3016C800D/VXML:/vxml_parse:
May 18 18:57:48.217: vxml_parse: XML_Parse success err=0
May 18 18:57:48.217: //0/15E3016C800D/VXML:/vxml_session_delete:
May 18 18:57:48.217: vxml_session_delete:mem_mgr_mempool_free: mempool=NULL
May 18 18:57:48.217: vxml_session_delete:mem_mgr_mempool_free: mempool=NULL
May 18 18:57:48.217: //-1//VXML:/vxml_crc_generate_element_tree: 28 crc
generated
May 18 18:57:48.217: //-1//VXML:/vxml_create:
enter url=http://10.10.210.10/ccmivr/pages/IVRMainpage.vxml
tree_handle=47E35714
return_handle_add=48547838
May 18 18:57:48.245: //44//AFW_:/vapp_get_type_detail:
May 18 18:57:48.245: //44/15E3016C800D/VXML:/vxml_offramp_mailhdrs_get:
May 18 18:57:48.245: //44//AFW_:/vapp_get_incoming_gtd_list:
May 18 18:57:48.249: //44/15E3016C800D/VXML:/vxml_jse_add_gtd_obj_to_list:
Sig-event name = setup_indication, gtd-len = 176, gtd-buf =
IAM,
PRN,isdn*,,NI***,
USI,rate,c,s,c,1
USI,lay1,alaw
TMR,00
HQ-RTR#CPN,04,,1,2123945999
CGN,02,,1,y,1,2123942123
CPC,09
FCI,,,y,
GCI,15e3016c61e611df800d001e1335ff88
May 18 18:57:48.249: //44/15E3016C800D/VXML:/vxml_jse_add_gtd_obj_to_list:
gtd_obj for sig-event [setup_indication] added to session/shadow
var array [0x4850DE2C]
May 18 18:57:48.249: //44/15E3016C800D/VXML:/vxml_create_gtd_sess_vars:
Created object chain for com.cisco.signal.gtdlist
May 18 18:57:48.249: //-1//VXML:/vxml_create: Exit
May 18 18:57:48.249: //44/15E3016C800D/VXML:/vxml_start:
vxmlhandle=47E34858 vapphandle=47E31A24 status=0 async_status=0
May 18 18:57:48.249: //44/15E3016C800D/VXML:/vxml_vxml_proc:
vxml
Re: [OSL | CCIE_Voice] Unified Contact Center Express for CCIE Voice v 3
hi, Not answered yet but also would like to know: Which licence version of UCCX (premier, enhance or standard) is recomended for lab? Peter - Original Message - From: Pulos, Greg gpu...@doc.gov To: Talmadge Almand t...@ipexpert.com; akash patel akashapa...@yahoo.com Cc: ccie_voice@onlinestudylist.com Sent: Friday, December 04, 2009 12:53 PM Subject: Re: [OSL | CCIE_Voice] Unified Contact Center Express for CCIE Voice v 3 An excellent source is the ccx getting started with scripts document. http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_7_0/user/guide/uccx70edgs.pdf thank you. gpulos -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Talmadge Almand Sent: Thursday, December 03, 2009 4:15 PM To: akash patel Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Unified Contact Center Express for CCIE Voice v 3 Yes to the troubleshooting being fairgame, and you can also expect any variation for scripting, I would recommend the advanced scripting training, or practice some custom scenarios, the ones that IPExpert details are a good measure of what to expect. On Thu, Dec 3, 2009 at 3:30 PM, akash patel akashapa...@yahoo.com wrote: I understood from previous blog/post that UCCX will be pre integrated with UCM in the lab, however we will be expected to troubleshoot the integration. I can take help from SRND, however as far 2nd piece of UCCX which is scripting, what material do you recommend to study? Is there any document outlining what version of UCXX is in test (premier, enhance or standard) and /or what topics will be tesable (can imagine standard AA, ACD), but weather email, DB dig out, web application, java applet creation could be testable as well? Thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ -- Regards, Talmadge Almand CCIE #20901 (Voice) Sr, Support Engineer - IPexpert, Inc. URL: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] No Audio in TCL IVR prompts
There r different script for SRST and CUCME. Loading incompatible tcl script can also cause the same issue. Peter - Original Message - From: Daryl Smith darylpsm...@gmail.com To: han...@tiscali.it; ccie_voice@onlinestudylist.com Sent: Thursday, November 19, 2009 12:59 PM Subject: Re: [OSL | CCIE_Voice] No Audio in TCL IVR prompts Check the CME -BACD admin guide on cisco site. You can copy and paste the examples in Notepad and then edit to the phone number you want in the script and the dial-peer DPS There are no secrets to success. It is the result of preparation, hard work, and learning from failure On 11/19/09 12:26 AM, han...@tiscali.it han...@tiscali.it wrote: Dear All, I'm asking your help to understand how tcl ivr prompts work. In order to introduce myself to tlc ivr scripting I wanted to explore a classic of this kind: its-CISCO.2.0.2.0.tcl This script is provided by Cisco, and it is an example of what a gateway could do. By my side, Itryed this script installing it in my 1751v. But I couldn't get to hear any prompt. I read many post of this Mailinglist, and others. I also read of someone who said that ivr prompts could not be played to voip dial-peers but only to pots. I read official docs, not carefully, I have to admit it, but I didn't see a page saying this. By the way the script I have uploaded on my router, seems to work, except I can not hear any sound back in my handset. here I paste the config I placed on my router; I would appreciate any suggest on why this odd behaviour. application service aa tftp://10.10.10.1 /its-CISCO.2.0.2.0.tcl paramspace english language en paramspace english index 1 paramspace english location tftp://10.10.10.1/ param aa-pilot 5000 param operator 104 dial-peer voice 250 voip service aa destination- pattern 5000 incoming called-number 5000 session protocol sipv2 codec g711ulaw no vad session target ipv4: 10.10.13.2 ! which is my loopback address codec g711ulaw the following a debug showing what's happen when I call the number. Nov 17 22:07:11.277 GMT: //- 1//HIFS: /hifs_ifs_cb: hifs ifs file read succeeded. size=7164, url=tftp://10. 10.10.1/its-CISCO.2.0.2.0.tcl Nov 17 22:07:11.305 GMT: //-1//HIFS: /hifs_free_idata: hifs_free_idata: 0x85354B94 Nov 17 22:07: 11.305 GMT: //-1//HIFS:/hifs_hold_idata: hifs_hold_idata: 0x85354B94 Nov 17 22:07: 11.325 GMT: //-1//AFW_: /tcl_RequiredVersionObjCmd: Script requires version 2.0. Nov 17 22:07: 11.345 GMT: //-1//AFW_: EE84F68554000: /Tcl_Link: Linking script aa Nov 17 22:07:11.381 GMT: //-1//TCL : EE84F68554000: /tcl_RequiredVersionObjCmd: Script requires version 2.0. So 2.1 is OK Nov 17 22:07:11.441 GMT: //-1//TCL : EE84F68554000: /tcl_InfotagObjCmd: infotag get cfg_avpair_exists aa- pilot Nov 17 22:07: 11.445 GMT: //- 1//TCL :EE84F68554000: /tcl_InfotagGetObjCmd: infotag get cfg_avpair_exists aa-pilot Nov 17 22:07:11.445 GMT: //-1//AFW_: EE84F68554000:/vtr_cf_avpair_exists: argc 3 argindex 2 Nov 17 22:07: 11.445 GMT: //-1//TCL :EE84F68554000: /tcl_InfotagObjCmd: infotag get cfg_avpair_exists operator Nov 17 22: 07:11.449 GMT: //-1//TCL : EE84F68554000:/tcl_InfotagGetObjCmd: infotag get cfg_avpair_exists operator Nov 17 22:07:11.449 GMT: //- 1//AFW_:EE84F68554000: /vtr_cf_avpair_exists: argc 3 argindex 2 Nov 17 22:07:11.453 GMT: //- 1//TCL :EE84F68554000:/tcl_FSMObjCmd: fsm define fsm CALL_INIT Nov 17 22:07:11.457 GMT: //-1//TCL :EE84F68554000: /tcl_FSMDefineObjCmd: State Machine: Array fsm: Start State: CALL_INIT Nov 17 22:07:11.461 GMT: //- 1//TCL :EE84F68554000: /tcl_FSMDefineObjCmd: FSM Data structure Nov 17 22:07:11.461 GMT: (CALLDISCONNECT(2), ev_media_done(146)--(act_Cleanup) --(any_state (0)) Nov 17 22:07:11.461 GMT: (GETDEST(3), ev_collectdigits_done(190) --(act_GotDest)--(HANDOFF(4)) Nov 17 22:07: 11.461 GMT: (any_state (0), ev_disconnect_done(1Cool--(act_Cleanup)-- (any_state(0)) Nov 17 22:07:11.461 GMT: (any_state(0), ev_disconnected (17)--(act_Cleanup)-- (any_state(0)) Nov 17 22:07:11.461 GMT: (HANDOFF (4), ev_setup_done (184)--(act_CallSetupDone)--(CONTINUE(5)) Nov 17 22: 07:11.461 GMT: (CALL_INIT(1), ev_setup_indication(30)--(act_Setup)-- (GETDEST(3)) Nov 17 22:07:11.461 GMT: FSM start state CALL_INIT(1) Nov 17 22:07: 11.465 GMT: //-1//AFW_:EE84F68554000:/Tcl_Link: Script aa succesfully linked. Nov 17 22:07:21.337 GMT: //-1//AFW_:EE84F68F4C000: /Tcl_Link: Linking script aa Nov 17 22:07:21.377 GMT: //-1//TCL : EE84F68F4C000: /tcl_RequiredVersionObjCmd: Script requires version 2.0. So 2.1 is OK Nov 17 22:07:21.393 GMT: //-1//TCL :EE84F68F4C000: /tcl_InfotagObjCmd: infotag get cfg_avpair_exists aa-pilot Nov 17 22: 07: 21.393 GMT: //- 1//TCL :EE84F68F4C000:/tcl_InfotagGetObjCmd: infotag get
Re: [OSL | CCIE_Voice] transformation on transfers
Some points. At this time CUCM is not able to manipulate redirecting number (except VM profile). This causes side effect in a situation when final called party is off-net to a CUCM cloud (reached over a trunk) and redirecting number have to be a full E.164. This easily happens when a phone registered to a CUCM is forwarded to an off-net number. Possible scenarios over a trunk: - Send last redirecting party as ANI. At this case the original calling number disappears so final called party cannot see original calling party. - Send original calling number as ANI. Here are major issues of CUCM as it is able to populate redirecting number with enterprise number only. For this issue the following solutions can be given: - Cisco IOS gateway after the CUCM manipulates redirecting number. - Enterprise numbers are full E.164 numbers and TPs are used for abbrevated dialing. This might be the prefered way. - Usage of VM profiles gives the tool to modify redirecting number, but this is little worth. Peter - Original Message - From: Vik Malhi vma...@ipexpert.com To: Robert McGhee bobwmcg...@verizon.net; Daryl Smith darylpsm...@gmail.com; OSL Group ccie_voice@onlinestudylist.com Sent: Monday, November 16, 2009 5:08 PM Subject: Re: [OSL | CCIE_Voice] transformation on transfers Page 142 (chapter 7-8) of the Features and Services guide talks briefly about Globalization and Localization on transfer and forward. In effect the original calling number will be used when the transfer completes. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Robert McGhee bobwmcg...@verizon.net Date: Sun, 15 Nov 2009 10:28:17 -0500 To: Daryl Smith darylpsm...@gmail.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] transformation on transfers Thanks Daryl, I thought it was mentioned in the dial plan section of the vod's but maybe I misunderstood. I'll keep looking but It'd make life easier if it wasn't possible :) -Original Message- From: Daryl Smith darylpsm...@gmail.com Sent: Saturday, November 14, 2009 8:53 PM To: Robert McGhee bobwmcg...@verizon.net; 'OSL Group' ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] transformation on transfers I remember reading something in the SRND and I've been going through it tonight about the call once it has reached its original destination. Being the call from the PSTN reached the HQ phone. The Call leaving the DP shouldn't do anything to the original called number since it is being transferred. The new Called number is the HQ Phone1 calling BR1 Phon1 not the PSTN phone calling BR1 Phone1. Correct me if I'm wrong I'm still trying to find the page on how the transfer works in the SRND or some other book I read. DPS On 11/14/09 6:54 PM, Robert McGhee bobwmcg...@verizon.net wrote: That's exactly correct, I figured the xformation would be applied to the br1 dp but that didn't work... -Original Message- From: Daryl Smith darylpsm...@gmail.com Sent: Saturday, November 14, 2009 5:46 PM To: Robert McGhee bobwmcg...@verizon.net; 'OSL Group' ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] transformation on transfers [The entire original message is not included] ___ 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] 3750 QoS Question
At earlier time AF31 was the prefered PHB for voice signaling, but the latest rule is CS3. The reason is because AF PHB can mark down or drop packages in contrast of CS PHB. hey, in the ! police 32 8000 exceed-action policed-dscp-transmit command the 8000 means burst volume not degradation to 8k of speed. Peter - Original Message - From: Alex Hannah To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com ; Farkas Péter Sent: Thursday, November 12, 2009 8:11 AM Subject: Re: [OSL | CCIE_Voice] 3750 QoS Question Michael, My understanding was older CUCM servers ( 4.x and early 5.x ) sent signalling out at AF31, also I thought I remembered something about CIPC not sending traffic out with right markings. I was trying to do a catch all to match any type of signaling be it either CS3 or AF31. And the police statement I have verified on my 2811 running 12.4(22) T2 ( Same as v3 lab last month ). So I believe this to be correct. What exactly did you mean by checking it to meet ONLY my requirements? The exceed action would remark traffic above 32k down to 8k correct? Thanks again, Alex 2009/11/11 Michael Ciarfello mciarfe...@iplogic.com That's looking better. Check your policed-dscp line to ONLY meet your requirements. Check the command reference and 3750 Switch COnfiguration guide - QoS chapter on that police command. I haven't looked at that or remember if it's correct. Pay attention to what Farkas said. Look at other documents to find the source of that. Maybe the document I mentioned above on what he is saying is in there. Why CS3 and AF31? If you have a home lab or a partial home lab, use a sniffer and sniff around. Let us know what you find. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Hannah [alex.han...@gmail.com] Sent: Wednesday, November 11, 2009 6:56 PM To: Farkas P¨¦ter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] 3750 QoS Question Michael and Farkas, Okay, I have thought about what you mentioned. Here is my revised approach. Let me know what you think about this way: ! mls qos map policed-dscp 0 24 to 8 mls qos map cos-dscp 0 8 16 24 32 46 48 56 mls qos ! ! class-map match-any SCCP-Traffic match ip dscp cs3 af31 ! ! policy-map POLICE-MAP class SCCP-Traffic police 32 8000 exceed-action policed-dscp-transmit set dscp cs3 ! ! interface FastEthernet0/6 service-policy input POLICE-MAP ! What is the signifigance of matching both ip dscp cs3 af31? Since I have match-any will it match on both? New CUCM 7.x servers should send SCCP out at cs3 correct? Thanks, Alex 2009/11/11 Farkas P¨¦ter wormh...@sch.bme.hu AutoQoS cannot be configured until service-policy is attached to the interface so you cannot use it for correction. Also, AutoQos does not work on Eth. - Original Message - From: Michael Ciarfello mciarfe...@iplogic.com Date: Wednesday, November 11, 2009 8:56 pm Subject: Re: [OSL | CCIE_Voice] 3750 QoS Question To: Alex Hannah alex.han...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Here are some hints for you to research: I believe there is an error in one of the class-maps. See if you can find it or agree. I believe you have too much extra stuff configured, let’s eliminate the unneeded stuff. How about use match IP protocol instead of access-lists? Are you sure your access-list is correct for the inbound / outbound traffic you have? I think the data vlan people are going to be pissed and complain about slowness. I know it’s a lab. I believe you can get the entire config down to a much simplier 10-15 lines instead of all the stuff you have. From: ccie_voice-boun...@onlinestudylist.com [ On Behalf Of Alex Hannah Sent: Wednesday, November 11, 2009 2:41 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] 3750 QoS Question Hello everyone. I am attempting to create the following QoS policy on a 3750 port with an IP Phone plugged in behind it. The policy will police signalling ( SCCP ) 32k down to 8k and remark to DSCP 8. I have read through most of the SRND guide for the 3750, the model I am following is the: 2970/3560/3750–Conditionally-Trusted IP Phone + PC + Scavenger (Basic) Model Configuration on page 105 of the 3.3 QoS SRND. Can anyone validate my work below and let me know if you think this meets those requirements? Also, in this scenerio, Auto
[OSL | CCIE_Voice] cat 3750 vs. 3560
hi, What are the main differences between catalyst 3750 and 3560 switches regarding lab exam? As till i only have 3560. thx. regards, Peter___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] written topics
Hi, I'm new to study for written exam. Who is over that please help me to preapre. Detailed knowledge of VoFR and VoATM are still required? And also lab does not includes any analog interface, and what about for written? Version of Unified Communication entities are all 7.0 for written, too? Thank you. Peter___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] ccievoicelabX.com == X means s
Another ad. :( X is s or X is e could be a filtering base :) w - Original Message - From: Arif Muslim To: ccie_voice@onlinestudylist.com Sent: Thursday, June 25, 2009 2:39 PM Subject: [OSL | CCIE_Voice] ccievoicelabX.com == X means s Hi, I passed my lab yestarday and it was great feeling in my heart. Whatever people will say here but i would to explore really really something from the heart ccievoicelabX.com (X is s) is the best, Now some people will bark here but i know they will never ever can archieve the CCIE id because to pass ccie u required luck, exact questions of real lab and practise ccievoicelabX.com (X is s) can give you exact questions but still you need to do a practise and still you required luck to be with you. But if you do not have the exact questions other 2 things will be definetly nothing for you. I heartly appreciate ccievoicelabX.com (X is s) and all the people here who told me to take the ccievoicelabX.com labs for practise (And they have also guided me properly till my attempt, i got new mgcp lab) in SJC As there is 3 labs new mgcp with VPIM , old mgcp with MOH, bacd and H323 with moh, bacd is perfect and word to word in real lab from them because my friend got h323 lab in brussels from them and it was also word to word. I havent seen so close any vendor i was struggling with cert / passXccielab from last 1 year but they are unbelieveable this is not any kind of marketing. Because people know now, every one is passing just taking thier labs now like cert for R$S People can PM me if they have any queries. Whatever they said it to me it was perfect might be they have many customers making back to back attempts or they will be having exact thing but that was unbelieveable. I am so happy so happy cant even express. Real live implementating knowledge we cannot compare in CCIE becuase CCIE is like your graduation and graduation books never we use in real life, so comparing both is shit with each other. Best of luck guys. Regards Arif CCIE VOICE -- Love Cricket? Check out live scores, photos, video highlights and more. Click here.
Re: [OSL | CCIE_Voice] DSP Calculation
Also how does one use g729 on an FXO, FXS or PRI B channel? That's when these voice volume terminated at a gateway and it produce VoIP streams based on G.729. This needs when WAN link only supports compressed voice codec. Here there are costumers who only buy the WAN link with VoIP capability but still operate their own legacy PBXs. Also, 'G.729 Annex A' and 'G.729 Annex A annex B' are reduced complexity codecs, so they should consume less DSP resources. Is it valid that both G.729 and its Annex A variant hit the same MIPS resource? - Original Message - From: Michael Ciarfello To: Cristi Radescu ; Art Sandborgh ; ccie list Sent: Tuesday, June 16, 2009 4:10 PM Subject: Re: [OSL | CCIE_Voice] DSP Calculation Want to add the D channel doesn't count as a signaling channel. So If you have pri-group timeslots 1-4,24 then only 4 channels are needed. Also how does one use g729 on an FXO, FXS or PRI B channel? The CCM 7.x SRND, Table 6-2 says: At 15 MIPS per call: .G.711 (a-law, mu-law) .Fax/modem passthrough .Clear channel At 30 MIPS per call: .G.726 (32K, 24K, 16K) .Fax relay .G.729 .G.729 (a, b, ab) At 40 MIPS per call: .G.728 .G.723.1 (32K, 24K, 16K) .G.723.1a (5.3K, 6.3K) .Modem relay What document did you get the g729's are 40 MIPS per call? We'll have to validate this ourselves if there is a documentation inconsistency (no surprise.) But need to know how to test this. I think it might be incoming PRI channel (or FXO port) TERMINATING to a g729r8 only device will use 30 MIPS. There is no codec on an FXO. I'll post an updated doc sheet soon after we validate some of these and give people a chance to add / update. From: Cristi Radescu [mailto:cristian.rade...@crescendo.ro] Sent: Tuesday, June 16, 2009 5:31 AM To: Michael Ciarfello; Art Sandborgh; ccie list Subject: RE: [OSL | CCIE_Voice] DSP Calculation Hi Michael, Very nice doc. Thanks for that. I'll do some corrections on it from my point of view. Please see below with blue. Hope this helps, Cristi -- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Michael Ciarfello Sent: Tuesday, June 16, 2009 7:02 AM To: Art Sandborgh; ccie list Subject: Re: [OSL | CCIE_Voice] DSP Calculation Here are my DSP notes as promised: Maybe we can finalize this and have IPexpert post it on their web site. PVDM2-16 is one DSP chip (32=2, 48=3, 64=4) - PVDM2-16 can manage 16 voice termination channels - has one DSP C5510(new one); DSP C5510 can be shared between transcoding and voice termination. Can not be shared if it's used for conferencing! PVDM2-8 is one DSP chip but less processing capacity than DSP on the 16 - PVDM2-8 can manage 8 voice termination channels PVDM2-16 - Signaling - 8 calls per DSP (medium complexity codecs) - 6 calls per DSP (high complexity codecs) - 240 MIPS in flex mode - G711 uses 15 MIPS per call (240 / 15 = 16 calls per DSP) - G729a, G729ab uses 30 MIPS per call (not all g729 variants) = 240 MIPS/30 = 8 medium complexity calls - G729, G729b uses 40 MIPS per call = 240 MIPS/40 = 6 high complexity calls(ATT: g729r8 is a high complexity codec!) PVDM-12 - can manage 12 voice termination channels - has 3 x DSP C549(the old one); each of these DSPs can manage 4 calls/transcoder sessions/voice termination channels no matter what codec is used; - one DSP C549 can not be shared(not even between voice termination and transcoding); - e.g. if you have 5 x timeslots E1/T1 you will have 2 X DSPs blocked for voice termination = it remains only one DSP(4 sessions) for transcoding; MTP - CCM SW MTP is G711 only (all versions including CCM7.x) - IOS SW MTP - Supports G711 and any G729 variant. But can choose only one codec on the dspfarm profile at a time. - Need the capacities. - IOS HW MTP - 16 G711 sessions per DSP - 6 G729 sessions per DSP PVDM2-16 - Conferencing - Each DSP accomodates 8 conference participants - IOS 12.4(15)T has new capability for 32 participants (needs verification) - Each DSP supports 8 conferences if G.711 is only configrued codec on the dspfarm profile. - Each DSP supports 2 conferences (of 8 participants each) if G.729 is CONFIGURED on the dspfarm. (even if all participants on all conferences are using G.711.) - I think you have to turn off GSM to get 8 conferences (needs verification) - Can't share conference on DSP with xcode or voice signaling. - Config max-sessions in multiples of 2 or 8 (depending on configured codec). Doesn't make sense to configure less--wasting resources. PVDM2-16 - Transcoding - Can
Re: [OSL | CCIE_Voice] DSP Calculation
One DSP must be dedicated for only one function (conferencing, transcoding, etc.), but PVDM2-32 consits 32 DSPs. DSP consumption for voice termination depends on voice codec as there are low and high complexity ones. Ex. terminating calls based on G.729 Annex A codec needs double DSPs than G.711. w - Original Message - From: Kamran Ahsanullah To: ccie_voice@onlinestudylist.com Sent: Monday, June 15, 2009 5:49 PM Subject: [OSL | CCIE_Voice] DSP Calculation real life situation and would like to understand the following: If customer has PVDM2-48, what can be done and how do we calculate it? I don't understand the DSP calculator. I realise a PVDM2-32 is needed to do an E1, can the remainder be used for conferencing, ( if so how many?) I see somewhere that you cannot share the DSP. Please help thanks
