Re: [OSL | CCIE_Voice] CME SRST Mode
You have the auto provision mode set to none. You should set to all so that CME will download the phone's config completely. This is most likely your problem. Try and change the command to srst mode auto all and let me know what happens. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot Sent: Wednesday, July 14, 2010 3:54 PM To: Edwin Dotson Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SRST Mode Done that, and phones get nothing. On Wed, Jul 14, 2010 at 5:51 PM, Edwin Dotson edot...@ams.net wrote: Have you tried removing the static ephone configuration? They may be picking that profile which don't have buttons specified. Thanks Edwin Sent from my Verizon Wireless BlackBerry -Original Message- From: ccieid1ot ccieid...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com Date: Wed, 14 Jul 2010 15:20:27 To: ccie_voice@onlinestudylist.comCCIE_Voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME SRST Mode Hi gangs, I can not seem to get the ephones to register with their assigned DN's. Tried all types of srst mode auto-pro commands. Someone please give me a sanity check. ! telephony-service sdspfarm units 5 sdspfarm tag 1 SC-CFB no privacy conference hardware srst mode auto-provision none srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 ip source-address 192.102.66.254 port 2000 system message Your phones are in fallback max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult create cnf-files version-stamp 7960 Apr 05 2009 12:47:04 ! ! ephone-dn-template 1 call-forward busy 91658200 call-forward noan 91658200 timeout 20 ! ! ephone-template 1 privacy off privacy-button softkeys remote-in-use Newcall CBarge softkeys idle Newcall Redial softkeys seized Redial Endcall Cfwdall Pickup Gpickup Meetme softkeys connected Hold Endcall Trnsfer Park Confrn ConfList ! ! ephone-dn 4 octo-line number B16A55 no-reg primary conference ad-hoc ! ! ephone-dn 10 octo-line number A100 conference ad-hoc ! ! ephone 1 privacy-button device-security-mode none max-calls-per-button 4 busy-trigger-per-button 1 ! ! ! ephone 2 privacy-button device-security-mode none max-calls-per-button 4 busy-trigger-per-button 1 ___ 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] CME SRST Mode
I have definitely had this problem before, but never really figure out why it happens. Try this: Copy your telephony-service config to Notepad do a no telephony-service Register the phones with CUCM again Paste in your telephony-service config Bring down the WAN connection. Eventually the phones will register with the proper DNs. It's not ideal, but it works most of the time. Hope this helps, Jeff -Original Message- From: ccieid1ot [mailto:ccieid...@gmail.com] Sent: Wednesday, July 14, 2010 4:10 PM To: Jeff Price (jeffpric) Cc: Edwin Dotson; ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SRST Mode Jeff, same thing. I had it registered back to CCM then failback to SRST. ccievoice-sitec#show telephony-service ephone-dn ephone-dn 4 octo-line number B16A55 no-reg primary preference 0 secondary 9 huntstop huntstop channel 8 call-waiting beep conference ad-hoc ephone-dn 10 octo-line number A100 preference 0 secondary 9 huntstop huntstop channel 8 call-waiting beep conference ad-hoc ccievoice-sitec#show run | s telephony telephony-service sdspfarm units 5 sdspfarm tag 1 SC-CFB no privacy conference hardware srst mode auto-provision all srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 ip source-address 142.102.66.254 port 2000 system message Your phones are in fallback max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult create cnf-files version-stamp 7960 Apr 05 2009 12:47:04 ccievoice-sitec# On Wed, Jul 14, 2010 at 5:57 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: You have the auto provision mode set to none. You should set to all so that CME will download the phone's config completely. This is most likely your problem. Try and change the command to srst mode auto all and let me know what happens. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot Sent: Wednesday, July 14, 2010 3:54 PM To: Edwin Dotson Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SRST Mode Done that, and phones get nothing. On Wed, Jul 14, 2010 at 5:51 PM, Edwin Dotson edot...@ams.net wrote: Have you tried removing the static ephone configuration? They may be picking that profile which don't have buttons specified. Thanks Edwin Sent from my Verizon Wireless BlackBerry -Original Message- From: ccieid1ot ccieid...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com Date: Wed, 14 Jul 2010 15:20:27 To: ccie_voice@onlinestudylist.comCCIE_Voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME SRST Mode Hi gangs, I can not seem to get the ephones to register with their assigned DN's. Tried all types of srst mode auto-pro commands. Someone please give me a sanity check. ! telephony-service sdspfarm units 5 sdspfarm tag 1 SC-CFB no privacy conference hardware srst mode auto-provision none srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 ip source-address 192.102.66.254 port 2000 system message Your phones are in fallback max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult create cnf-files version-stamp 7960 Apr 05 2009 12:47:04 ! ! ephone-dn-template 1 call-forward busy 91658200 call-forward noan 91658200 timeout 20 ! ! ephone-template 1 privacy off privacy-button softkeys remote-in-use Newcall CBarge softkeys idle Newcall Redial softkeys seized Redial Endcall Cfwdall Pickup Gpickup Meetme softkeys connected Hold Endcall Trnsfer Park Confrn ConfList ! ! ephone-dn 4 octo-line number B16A55 no-reg primary conference ad-hoc ! ! ephone-dn 10 octo-line number A100 conference ad-hoc ! ! ephone 1 privacy-button device-security-mode none max-calls-per-button 4 busy-trigger-per-button 1 ! ! ! ephone 2 privacy-button device-security-mode none max-calls-per-button 4 busy-trigger-per-button 1 ___ 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] CME SRST Mode
Good point. -Original Message- From: Randall Saborio [mailto:ill2...@gmail.com] Sent: Wednesday, July 14, 2010 4:10 PM To: Jeff Price (jeffpric) Cc: ccieid1ot; Edwin Dotson; ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SRST Mode srst mode auto-provision none is not a problem if you intend it this way. However, you must add the buttons to the ephone configuration: ephone 1 button 1:4 On Wed, Jul 14, 2010 at 4:57 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: You have the auto provision mode set to none. You should set to all so that CME will download the phone's config completely. This is most likely your problem. Try and change the command to srst mode auto all and let me know what happens. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot Sent: Wednesday, July 14, 2010 3:54 PM To: Edwin Dotson Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SRST Mode Done that, and phones get nothing. On Wed, Jul 14, 2010 at 5:51 PM, Edwin Dotson edot...@ams.net wrote: Have you tried removing the static ephone configuration? They may be picking that profile which don't have buttons specified. Thanks Edwin Sent from my Verizon Wireless BlackBerry -Original Message- From: ccieid1ot ccieid...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com Date: Wed, 14 Jul 2010 15:20:27 To: ccie_voice@onlinestudylist.comCCIE_Voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME SRST Mode Hi gangs, I can not seem to get the ephones to register with their assigned DN's. Tried all types of srst mode auto-pro commands. Someone please give me a sanity check. ! telephony-service sdspfarm units 5 sdspfarm tag 1 SC-CFB no privacy conference hardware srst mode auto-provision none srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 ip source-address 192.102.66.254 port 2000 system message Your phones are in fallback max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult create cnf-files version-stamp 7960 Apr 05 2009 12:47:04 ! ! ephone-dn-template 1 call-forward busy 91658200 call-forward noan 91658200 timeout 20 ! ! ephone-template 1 privacy off privacy-button softkeys remote-in-use Newcall CBarge softkeys idle Newcall Redial softkeys seized Redial Endcall Cfwdall Pickup Gpickup Meetme softkeys connected Hold Endcall Trnsfer Park Confrn ConfList ! ! ephone-dn 4 octo-line number B16A55 no-reg primary conference ad-hoc ! ! ephone-dn 10 octo-line number A100 conference ad-hoc ! ! ephone 1 privacy-button device-security-mode none max-calls-per-button 4 busy-trigger-per-button 1 ! ! ! ephone 2 privacy-button device-security-mode none max-calls-per-button 4 busy-trigger-per-button 1 ___ 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 -- Randall da ill Saborio CCIE Voice Wannabe #10054675811 (Real number coming this July 2010) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SRST Mode
Your source address changed from the first post, was it supposed to? I'm really reach out on a limb here trying to help you b/c your config looks correct. I would maybe suggest starting from square one and keep it simple. Remove all of the commands and only include the srst mode auto all command. See if your phones register like that. Then add your various other statements one by one. See if one of them is screwing everything up? Jeff -Original Message- From: CCIE Voice [mailto:ccievoiced...@gmail.com] Sent: Wednesday, July 14, 2010 4:32 PM To: Jeff Price (jeffpric); 'ccieid1ot' Cc: 'ccie_voice@onlinestudylist.com'; ccie_voice-boun...@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] CME SRST Mode Jeff is correct...whenever I have to make a change to CME SRST config, especially the ephones, I copy all of my related config to notepad and issue the no telephony-service command. I then paste everything back in and it works fine. Don't know if this is a bug or what but it happens every single time. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price (jeffpric) Sent: Wednesday, July 14, 2010 7:12 PM To: ccieid1ot Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SRST Mode I have definitely had this problem before, but never really figure out why it happens. Try this: Copy your telephony-service config to Notepad do a no telephony-service Register the phones with CUCM again Paste in your telephony-service config Bring down the WAN connection. Eventually the phones will register with the proper DNs. It's not ideal, but it works most of the time. Hope this helps, Jeff -Original Message- From: ccieid1ot [mailto:ccieid...@gmail.com] Sent: Wednesday, July 14, 2010 4:10 PM To: Jeff Price (jeffpric) Cc: Edwin Dotson; ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SRST Mode Jeff, same thing. I had it registered back to CCM then failback to SRST. ccievoice-sitec#show telephony-service ephone-dn ephone-dn 4 octo-line number B16A55 no-reg primary preference 0 secondary 9 huntstop huntstop channel 8 call-waiting beep conference ad-hoc ephone-dn 10 octo-line number A100 preference 0 secondary 9 huntstop huntstop channel 8 call-waiting beep conference ad-hoc ccievoice-sitec#show run | s telephony telephony-service sdspfarm units 5 sdspfarm tag 1 SC-CFB no privacy conference hardware srst mode auto-provision all srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 ip source-address 142.102.66.254 port 2000 system message Your phones are in fallback max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult create cnf-files version-stamp 7960 Apr 05 2009 12:47:04 ccievoice-sitec# On Wed, Jul 14, 2010 at 5:57 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: You have the auto provision mode set to none. You should set to all so that CME will download the phone's config completely. This is most likely your problem. Try and change the command to srst mode auto all and let me know what happens. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot Sent: Wednesday, July 14, 2010 3:54 PM To: Edwin Dotson Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SRST Mode Done that, and phones get nothing. On Wed, Jul 14, 2010 at 5:51 PM, Edwin Dotson edot...@ams.net wrote: Have you tried removing the static ephone configuration? They may be picking that profile which don't have buttons specified. Thanks Edwin Sent from my Verizon Wireless BlackBerry -Original Message- From: ccieid1ot ccieid...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com Date: Wed, 14 Jul 2010 15:20:27 To: ccie_voice@onlinestudylist.comCCIE_Voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME SRST Mode Hi gangs, I can not seem to get the ephones to register with their assigned DN's. Tried all types of srst mode auto-pro commands. Someone please give me a sanity check. ! telephony-service sdspfarm units 5 sdspfarm tag 1 SC-CFB no privacy conference hardware srst mode auto-provision none srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 ip source-address 192.102.66.254 port 2000 system message Your phones are in fallback max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult create cnf-files version-stamp 7960 Apr 05 2009 12:47:04 ! ! ephone-dn-template 1 call-forward busy 91658200 call-forward noan 91658200 timeout 20 ! ! ephone-template 1 privacy off
Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue
I still think the easiest way to do this is to have all dial-peers with a 9. As you are configuring, you may run into issues with TEHO, but if you do a debug voip dialpeer you can see the incoming number and add any extra dial-peers you may need. I have noticed for some reason when I configure TEHO, sometimes CUCM doesn't add the 9 even though it is in the prefix box of the RL. In this event, as I said, it would be easier to do the debug and figure out what is being sent. This simplifies adding the SRST functionality, which you can pretty much guarantee you will have to configure in some fashion. As you do these configurations over and over in this method, you will start to just think naturally about sending the 9. It is easier to have the 1 dial-peer than have to create 2 for each type of dialing pattern. Just my opinion, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Monday, July 12, 2010 6:23 PM To: Graham Hopkins Cc: CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue I proceeded to use the method where all my H323 dial-peers start with 9 in the destination-pattern. I imagine it's more work to have UCM keep the 9 on the dialed number because of TEHO to multiple gateways, it gets very busy to know when to prepend and not prepend in UCM route lists. Assuming 9 is stripped on UCM and the H323 gateway is adding 9 before sending the call to a POTS dial peer, is a VoIP dial-peer being created to match any incoming call and then it is sent through a translation-profile so it can match a POTS dial peer? On Jul 9, 2010, at 12:22 PM, Graham Hopkins wrote: With the two sets of dial-peers you do need to take care that overlapping patterns don't cause problems in SRST for example I hit issues with [2-9].. and 91[2-9]..[2-9].. I decided to go with the translation pattern to put the 9 back on to the digits sent by CUCM, but this 9 will still show up on the phone unless you use voice service voip no supplementary-service h225-notify cid-update Regards Graham Hopkins On 9 Jul 2010, at 19:21, Mark Holloway wrote: Sounds like you have the PSTN to CUCM part working ok. This is what I have been doing. On the H323 router create the following dial-peer dial-peer voice 10 pots destination-pattern [2-9]..$ port 0/0/0:23 On CUCM have a Route Pattern that handles \+1414.[2-9]XX for calls originated by BR1 phones and strip the predot. This way you can assign the call type as Subscriber within the Route Pattern and if local calls are supposed to send a 7 digit calling number you can set the calling party transformation mask within the Route Pattern to XXX. You could have a second dial-peer on your H323 router for SRST dial-peer voice 910 pots destination-pattern 9[2-9]..$ port 0/0/0:23 translation-profile outgoing LOCAL There are really two different ways to handle H323 gateway dial-peers. You can strip the 9 in CUCM then add it back on the H323 gateway through a translation-profile and only have one set of dial-peers. Or, build your dial-peers for local, LD, international, and 911 without the 9, copy/paste in notepad and put a 9 in front of the dial-peer number and the destination-pattern then paste it into your router. You will have two sets of dial-peers for SRST and normal operation. On Jul 9, 2010, at 10:28 AM, Joaquim Fernandes wrote: HI Team, I have an issue with this question. Question === when pstn number 414363 call phones at site b they should display 7 digits on the phone display. For example when pstn calling ph 1 or ph 2 at branch B it should display 363 on the screen. My solution = I have added +1 in Device pool of Branch B to make it globalised when the call comes in the H323 Branch B router. I have created \+1414.363 calling party transformation mask. I have created \+1414.363 route pattern with Branch B as the gateway. (branch b is the H323 gateway). So on the Route pattern i have just done predot and in the branch b route list i have done NANP-Predot and prefix 9. I have done vice versa as well but things doesnt work. IN the branch B router i have a dial-peer for the local calls. dial-peer voice 1 pots destination-pattern 9[2-9].. port 0/0/0:23 translation-profile outgoing local translation-rule 1 rule 1 /^8.../ /363\0/ translation-rule 2 rule 1 // // type any sub plan any isdn translation-profile lcoal translate called 2 translate calling 1 Note: If i make a dial-peer without 9 i.e (...) Then the display is perfect. but i dont feel this would be the solution. because in srst this would be an issue. Issue = The issue is when PSTN phone 414363 calls Brach B ph1 or ph2 the caller id is 363 and
Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue
The easiest way to accomplish this is to create a dial-peer for the number w/o the 9. Also, to effect the display on the phone, I've also seen that you would have to strip predot on the Route Pattern. So you will need the following: - A route pattern \+1414.XXX with DDI Predot - To route the call and display correctly on the phone - A calling party transformation incoming \+1414.XXX DDI Predot- to strip the number down on the display when calling from the PSTN - A called party transformation outgoing \+1414.XXX DDI Predot- to strip the number down when the call is going out - A dial-peer for the number 363 - to route the call from the GW I have done this plenty of times without issue with SRST. Your users would still use 9 to dial outside, therefore they technically wouldn't be aware of that dial-peer and wouldn't dial the number 363. Also if you do it right, you can still add redundancy at the RL to allow failover to the backup gateways. Also, it's worth noting that you should probably apply the incoming calling number prefix on the gateway instead of the Device Pool. Although I suppose it could be personal preference rather than any real issue. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Joaquim Fernandes Sent: Friday, July 09, 2010 10:28 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Globalisation/Localisation Issue HI Team, I have an issue with this question. Question === when pstn number 414363 call phones at site b they should display 7 digits on the phone display. For example when pstn calling ph 1 or ph 2 at branch B it should display 363 on the screen. My solution = I have added +1 in Device pool of Branch B to make it globalised when the call comes in the H323 Branch B router. I have created \+1414.363 calling party transformation mask. I have created \+1414.363 route pattern with Branch B as the gateway. (branch b is the H323 gateway). So on the Route pattern i have just done predot and in the branch b route list i have done NANP-Predot and prefix 9. I have done vice versa as well but things doesnt work. IN the branch B router i have a dial-peer for the local calls. dial-peer voice 1 pots destination-pattern 9[2-9].. port 0/0/0:23 translation-profile outgoing local translation-rule 1 rule 1 /^8.../ /363\0/ translation-rule 2 rule 1 // // type any sub plan any isdn translation-profile lcoal translate called 2 translate calling 1 Note: If i make a dial-peer without 9 i.e (...) Then the display is perfect. but i dont feel this would be the solution. because in srst this would be an issue. Issue = The issue is when PSTN phone 414363 calls Brach B ph1 or ph2 the caller id is 363 and in the missed call its globalized number +1414363 as per the question. But when i do redial using missed calls from Branch B ph1 or ph2 the calling number on the ip phones is displayed as 9363 (9 is the secondary dial tone) and the call goes through. Evrything works fine except for the display on ph1 or ph2, there is 9. How do i get rid of it 9. I hope i have made my point very clear of what issue i am facing. The question state the display on the phone should be only 363 and not 9363. Regards, JF ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PSTN Gatekeeper
On the PSTN gatekeeper, do you have an incoming dial-peer and a
translation-pattern to strip the number down?
Also, do a debug gatekeeper main 10 on HQ router to make sure that the
GK is even routing the call. Then move to the PSTN router and do a
debug h225 asn1 to see the H225 messages and then a debug dialpeer voip
to see whether you dial-peer is matching.
Hope this will help you get started in figuring this out,
Jeff
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mouhammad
Nasser
Sent: Wednesday, July 07, 2010 11:09 AM
To: ipexpert
Subject: [OSL | CCIE_Voice] PSTN Gatekeeper
Hi Experts,
I am trying to configure PSTN gatekeeper to test Gatekeeper VIA routing,
so I decided to start with a simple scenario where no VIA zone is
introduced, but I am not able to get it working, the dialed string
011916745738932 arrives to PSTN gatekeeper but with no hope to match,
and the phone never rings
I hope anyone can help
My configuration and debug are below
HQ router
gatekeeper
zone local GK ccie.com 142.1.64.254
zone remote BBGK cisco.com 10.1.5.5 1719
zone prefix BBGK 011*
no shutdown
!
++
on PSTN router it is:
interface GigabitEthernet0/0.5
encapsulation dot1Q 5 native
ip address 10.1.5.5 255.255.255.0
h323-gateway voip interface
h323-gateway voip id BBGK ipaddr 10.1.5.5 1719
h323-gateway voip h323-id PSTN
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr 10.1.5.5
!
!
gatekeeper
zone local BBGK cisco.com 10.1.5.5
zone remote GK ccie.com 142.1.64.254 1719
zone prefix BBGK 011*
gw-type-prefix 1#* default-technology
no shutdown
!
!
the dialed string is registered to gatekeeper:
PSTN-RTR#sh gatekeeper endpoint
GATEKEEPER ENDPOINT REGISTRATION
CallSignalAddr Port RASSignalAddr Port Zone Name Type
Flags
--- - --- - -
-
10.1.5.51720 10.1.5.556645 BBGK VOIP-GW
E164-ID: 011916745738932
H323-ID: PSTN
Voice Capacity Max.= Avail.= Current.= 0
Total number of active registrations = 1
the output of debug h225 asn1 is below
Jul 7 18:08:51.891:
Jul 7 18:08:51.891: RAS INCOMING PDU ::=
value RasMessage ::= locationRequest :
{
requestSeqNum 2054
destinationInfo
{
dialedDigits : 011916745738932
}
nonStandardData
{
nonStandardIdentifier h221NonStandard :
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
data '828B903011EDB5EBEBA941C3070007018ECA...'H
}
replyAddress ipAddress :
{
ip '8E0140FE'H
port 1719
}
sourceInfo
{
h323-ID : {GK}
}
canMapAlias TRUE
hopCount 6
}
Jul 7 18:08:51.891: H225 NONSTD INCOMING ENCODE BUFFER::=
828B903011EDB5EBEBA941C3070007018ECA401D05010180833407000A01C815820A
211E0034003800330044003800320044004300300030003000300030003000300032
Jul 7 18:08:51.891:
Jul 7 18:08:51.891: H225 NONSTD INCOMING PDU ::=
value LRQnonStandardInfo ::=
{
ttl 6
nonstd-callIdentifier
{
guid '00EDB5EBEBA941C3070007018ECA401D'H
}
gatewaySrcInfo
{
e164 : 5001
}
h225NonStdSrcCallSignalAddress h225NonStdIpAddress :
{
ip '0A01C815'H
port 33290
}
h225NonStdSrcendpointIdentifier {483D82DC0002}
}
Jul 7 18:08:51.895: H225 NONSTD OUTGOING PDU ::=
value LCFnonStandardInfo ::=
{
termAlias
{
e164 : 011916745738932,
h323-ID : {PSTN}
}
gkID {BBGK}
gateways
{
{
gwType voip : NULL
gwAlias
{
e164 : 011916745738932,
h323-ID : {PSTN}
}
sigAddress
{
ip '0A010505'H
port 1720
}
resources
{
maxDSPs 0
inUseDSPs 0
maxBChannels 0
inUseBChannels 0
activeCalls 0
bandwidth 0
inuseBandwidth 0
}
}
}
}
Jul 7 18:08:51.895: H225 NONSTD OUTGOING ENCODE BUFFER::=
00020700344C49A78A6BC65403005000530054004E06004200420047004B011002070034
4C49A78A6BC65403005000530054004E000A01050506B8
Jul 7 18:08:51.895:
Jul 7 18:08:51.895: RAS OUTGOING PDU ::=
value RasMessage ::= locationConfirm :
{
requestSeqNum 2054
callSignalAddress ipAddress :
{
ip '0A010505'H
port 1720
}
rasAddress ipAddress :
{
ip '0A010505'H
port 56645
}
nonStandardData
{
nonStandardIdentifier h221NonStandard :
{
t35CountryCode 181
t35Extension 0
Re: [OSL | CCIE_Voice] Request pending... when trying to access IPPAservice
I've seen that a restart of the IPPA service and UCCX Engine under the Control Center will fix most problems such as this, however I can't say that I have ever seen that exact status. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co Sent: Wednesday, July 07, 2010 11:04 AM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: [OSL | CCIE_Voice] Request pending... when trying to access IPPAservice Hi Guys, I get Request pending... when trying to access IPPA service URL I used.: http://100.0.0.11:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp Also all of the names have been changed to IP addresses in Enterprise param. Any idea ? Cheers, Jeremy ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PSTN Gatekeeper
My apologies, I didn't read the whole thing to see the H225 messages. I
would still make sure that you have the incoming dial-peer configured.
If the location request/confirm is sent, that would mean the GK routing
is there, but the actual call routing isn't. Mainly you need to strip
the tech-prefix that will be in front of the dialed number. Depending
on how you have the DN configured on the PSTN router, you may need to
strip the string down further, but this would be based upon your
specific implementation.
Jeff
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mouhammad
Nasser
Sent: Wednesday, July 07, 2010 11:09 AM
To: ipexpert
Subject: [OSL | CCIE_Voice] PSTN Gatekeeper
Hi Experts,
I am trying to configure PSTN gatekeeper to test Gatekeeper VIA routing,
so I decided to start with a simple scenario where no VIA zone is
introduced, but I am not able to get it working, the dialed string
011916745738932 arrives to PSTN gatekeeper but with no hope to match,
and the phone never rings
I hope anyone can help
My configuration and debug are below
HQ router
gatekeeper
zone local GK ccie.com 142.1.64.254
zone remote BBGK cisco.com 10.1.5.5 1719
zone prefix BBGK 011*
no shutdown
!
++
on PSTN router it is:
interface GigabitEthernet0/0.5
encapsulation dot1Q 5 native
ip address 10.1.5.5 255.255.255.0
h323-gateway voip interface
h323-gateway voip id BBGK ipaddr 10.1.5.5 1719
h323-gateway voip h323-id PSTN
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr 10.1.5.5
!
!
gatekeeper
zone local BBGK cisco.com 10.1.5.5
zone remote GK ccie.com 142.1.64.254 1719
zone prefix BBGK 011*
gw-type-prefix 1#* default-technology
no shutdown
!
!
the dialed string is registered to gatekeeper:
PSTN-RTR#sh gatekeeper endpoint
GATEKEEPER ENDPOINT REGISTRATION
CallSignalAddr Port RASSignalAddr Port Zone Name Type
Flags
--- - --- - -
-
10.1.5.51720 10.1.5.556645 BBGK VOIP-GW
E164-ID: 011916745738932
H323-ID: PSTN
Voice Capacity Max.= Avail.= Current.= 0
Total number of active registrations = 1
the output of debug h225 asn1 is below
Jul 7 18:08:51.891:
Jul 7 18:08:51.891: RAS INCOMING PDU ::=
value RasMessage ::= locationRequest :
{
requestSeqNum 2054
destinationInfo
{
dialedDigits : 011916745738932
}
nonStandardData
{
nonStandardIdentifier h221NonStandard :
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
data '828B903011EDB5EBEBA941C3070007018ECA...'H
}
replyAddress ipAddress :
{
ip '8E0140FE'H
port 1719
}
sourceInfo
{
h323-ID : {GK}
}
canMapAlias TRUE
hopCount 6
}
Jul 7 18:08:51.891: H225 NONSTD INCOMING ENCODE BUFFER::=
828B903011EDB5EBEBA941C3070007018ECA401D05010180833407000A01C815820A
211E0034003800330044003800320044004300300030003000300030003000300032
Jul 7 18:08:51.891:
Jul 7 18:08:51.891: H225 NONSTD INCOMING PDU ::=
value LRQnonStandardInfo ::=
{
ttl 6
nonstd-callIdentifier
{
guid '00EDB5EBEBA941C3070007018ECA401D'H
}
gatewaySrcInfo
{
e164 : 5001
}
h225NonStdSrcCallSignalAddress h225NonStdIpAddress :
{
ip '0A01C815'H
port 33290
}
h225NonStdSrcendpointIdentifier {483D82DC0002}
}
Jul 7 18:08:51.895: H225 NONSTD OUTGOING PDU ::=
value LCFnonStandardInfo ::=
{
termAlias
{
e164 : 011916745738932,
h323-ID : {PSTN}
}
gkID {BBGK}
gateways
{
{
gwType voip : NULL
gwAlias
{
e164 : 011916745738932,
h323-ID : {PSTN}
}
sigAddress
{
ip '0A010505'H
port 1720
}
resources
{
maxDSPs 0
inUseDSPs 0
maxBChannels 0
inUseBChannels 0
activeCalls 0
bandwidth 0
inuseBandwidth 0
}
}
}
}
Jul 7 18:08:51.895: H225 NONSTD OUTGOING ENCODE BUFFER::=
00020700344C49A78A6BC65403005000530054004E06004200420047004B011002070034
4C49A78A6BC65403005000530054004E000A01050506B8
Jul 7 18:08:51.895:
Jul 7 18:08:51.895: RAS OUTGOING PDU ::=
value RasMessage ::= locationConfirm :
{
requestSeqNum 2054
callSignalAddress ipAddress :
{
ip '0A010505'H
port 1720
}
rasAddress ipAddress :
{
ip '0A010505'H
port 56645
}
nonStandardData
{
nonStandardIdentifier h221NonStandard :
Re: [OSL | CCIE_Voice] Request pending... when trying to access IPPAservice
What port number are you using? The URL provide in the CAD installation guide uses 8080, but the real port should be 6293. Try that if you have not. Jeff From: jeremy co [mailto:jeremy.coo...@gmail.com] Sent: Wednesday, July 07, 2010 11:29 AM To: Jeff Price (jeffpric); avholloway+cisco-v...@gmail.com Cc: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: Re: [OSL | CCIE_Voice] Request pending... when trying to access IPPAservice Hi guys, I reloaded both CUCM and UCCX servers and I cannot get anything in browser with this URL. Cheers, Jeremy On Thu, Jul 8, 2010 at 4:22 AM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: I've seen that a restart of the IPPA service and UCCX Engine under the Control Center will fix most problems such as this, however I can't say that I have ever seen that exact status. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeremy co Sent: Wednesday, July 07, 2010 11:04 AM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: [OSL | CCIE_Voice] Request pending... when trying to access IPPAservice Hi Guys, I get Request pending... when trying to access IPPA service URL I used.: http://100.0.0.11:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp Also all of the names have been changed to IP addresses in Enterprise param. Any idea ? Cheers, Jeremy ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PSTN Gatekeeper
The incoming called number would actually be this:
dial-peer voice 200 voip
incoming called 1#
translation-profile in FROM_GK
voice translation-rule 1
rule 1 /1#\(.*\)/ /\1/
voice translation-profile FROM_GK
translate called 1
The called number will come in from the GK with the tech prefix that you
configured. You will need to strip this in order to route the call
correctly. Right now you are receiving an unallocated number b/c the
number would come in 1#01191674563892 and the GW doesn't have this
number for any endpoints.
If you aren't even getting any output from the debug voip dialpeer
command, that would mean that you have a GK routing issue. However, I
suspect that b/c your dialpeer sent the call back to RAS, you can
probably fix with what I am telling you. Keep me posted on your
progress.
Hope this helps,
Jeff
From: Mouhammad Nasser [mailto:engnasse...@hotmail.com]
Sent: Wednesday, July 07, 2010 11:36 AM
To: Jeff Price (jeffpric)
Cc: ipexpert
Subject: RE: [OSL | CCIE_Voice] PSTN Gatekeeper
Hi Jeff,
Thank you for your reply
In order to make things simplist, I altered the extension number
configured at CME to be exactly like the arrived dialed string
011916745738932, so no translation would be required
Anyway, I wasn't having an incoming dial-peer as you mentioned, so I
added the below:
dial-peer voice 200 voip
incoming called-number 01191T
session target ras
!
But also with no hope, the call does never reach the gateway (gateway
part of the router), because I receive no output at all for both debug
!
debug voip dialpeer all
debug voip ccapi inout
!
Sir, I have been trying for few days on it with no hope :(
I shall appreciate any help
BR,
Subject: RE: [OSL | CCIE_Voice] PSTN Gatekeeper
Date: Wed, 7 Jul 2010 13:26:29 -0500
From: jeffp...@cisco.com
To: engnasse...@hotmail.com; ccie_voice@onlinestudylist.com
My apologies, I didn't read the whole thing to see the H225 messages. I
would still make sure that you have the incoming dial-peer configured.
If the location request/confirm is sent, that would mean the GK routing
is there, but the actual call routing isn't. Mainly you need to strip
the tech-prefix that will be in front of the dialed number. Depending
on how you have the DN configured on the PSTN router, you may need to
strip the string down further, but this would be based upon your
specific implementation.
Jeff
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mouhammad
Nasser
Sent: Wednesday, July 07, 2010 11:09 AM
To: ipexpert
Subject: [OSL | CCIE_Voice] PSTN Gatekeeper
Hi Experts,
I am trying to configure PSTN gatekeeper to test Gatekeeper VIA routing,
so I decided to start with a simple scenario where no VIA zone is
introduced, but I am not able to get it working, the dialed string
011916745738932 arrives to PSTN gatekeeper but with no hope to match,
and the phone never rings
I hope anyone can help
My configuration and debug are below
HQ router
gatekeeper
zone local GK ccie.com 142.1.64.254
zone remote BBGK cisco.com 10.1.5.5 1719
zone prefix BBGK 011*
no shutdown
!
++
on PSTN router it is:
interface GigabitEthernet0/0.5
encapsulation dot1Q 5 native
ip address 10.1.5.5 255.255.255.0
h323-gateway voip interface
h323-gateway voip id BBGK ipaddr 10.1.5.5 1719
h323-gateway voip h323-id PSTN
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr 10.1.5.5
!
!
gatekeeper
zone local BBGK cisco.com 10.1.5.5
zone remote GK ccie.com 142.1.64.254 1719
zone prefix BBGK 011*
gw-type-prefix 1#* default-technology
no shutdown
!
!
the dialed string is registered to gatekeeper:
PSTN-RTR#sh gatekeeper endpoint
GATEKEEPER ENDPOINT REGISTRATION
CallSignalAddr Port RASSignalAddr Port Zone Name Type
Flags
--- - --- - - nb! sp;
-
10.1.5.51720 10.1.5.556645 BBGK VOIP-GW
E164-ID: 011916745738932
H323-ID: PSTN
Voice Capacity Max.= Avail.= Current.= 0
Total number of active registrations = 1
the output of debug h225 asn1 is below
Jul 7 18:08:51.891:
Jul 7 18:08:51.891: RAS INCOMING PDU ::=
value RasMessage ::= locationRequest :
{
requestSeqNum 2054
destinationInfo
{
dialedDigits : 011916745738932
}
nonStandardData
{
nonStandardIdentifier h221NonStandard :
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
data '828B903011EDB5EBEBA941C3070007018ECA...'H
}
replyAddress ipAddress :
{
ip '8E0140FE'H
port 1719
}
sourceInfo
{
h323-ID : {GK}
}
canMapAlias TRUE
! nbsp;nb sp; hopCount 6
[OSL | CCIE_Voice] Intercom Label
Hey everyone, I have noticed this behavior when configuration IPMA in proxy mode. When I first configure the Intercoms, they are labeled as I want as Manager Intercom on the Assistant's phone, and Assistant Intercom on the Manager's phone. Seemingly at random times, the label will switch to the name of the respective username that the Intercom destination is associated with. For instance, one my HQ Phone 2 (the Manager's phone) will say Assistant Intercom and then randomly switch to hquser1. Does anyone know why this behavior occurs? Is there any way to fix it? I have reset the phones and one has reverted back to the configured label, whereas the other has stayed with the username. Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com mailto:jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html http://www.cisco.com/web/about/doing_business/legal/cri/index.html image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Transcoder/Conf MRG
I would say better to put them in MRGs and then MRGLs. Although both would work, its better have control over who can access them. For example - HQ_R1_CONF_MRG, BR1_R2_XCODE_MRG Then create separate MRGLs with the same MRGs in them: HQ_MRGL - HQ_R1_CONF_MRG, BR1_R2_XCODE_MRG BR1_MRGL - HQ_R1_CONF_MRG, BR1_R2_XCODE_MRG However, for the exam purposes, it may just be easier to leave out for time J Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bo Gao Sent: Friday, June 25, 2010 6:06 PM To: OSL Subject: [OSL | CCIE_Voice] Transcoder/Conf MRG If I want HQ, BR1, and BR2 all share one HD conference bridge and one HD transcoder, will it be better if I just leave these resources in the default null MRG, or assign them into the HQ_MRG, BR1_MRG, and BR2_MRG? Thanks, Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Streaming MoH from Router Flash
Hey Mark, You will need under ccm-manager-fallback the following: max-ephones 1 max-dn 1 ip source ip address of voice vlan/loop Also, under global config add the following command: ccm-manager music-on-hold bind source ip address configured under ccm-manager-fallback Hope this helps, Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Sent: Wednesday, June 23, 2010 10:40 AM To: Amy Ryan Cc: OSL osl Subject: Re: [OSL | CCIE_Voice] Streaming MoH from Router Flash Thanks guys, in this case I am not using CME. BR2 is part of UCM. The 'problem' is when BR2 puts a call on hold there is silence. BR1 puts a call on hold and there is music. I am essentially trying to accomplish the same thing with both BR1 and BR2 by streaming from router flash yet things appear to be the same but not working for BR2. Amy, I created a dedicated Device Pool (DP_MoH) and Region (REG_MoH). Hop count is 1 and I have 1 audio source. Since BR1 is working I think my UCM configuration is good. Should the call-manager-fallback config on BR2 use 239.1.1.1 which is what I am using on BR1 or would I need to increment the IP? BR1 call-manager-fallback max-conferences 8 gain -6 transfer-system full-consult moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 10.20.10.1 10.220.10.254 interface Vlan302 description VOICE ip pim dense-mode BR2 call-manager-fallback max-conferences 8 gain -6 transfer-system full-consult moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 10.30.10.1 10.230.10.254 .1 = Voice VLAN .254 = Loopback My Device Pools have the appropriate MRGL assigned. MRGL_BR1 and MRGL_BR2 have the same MRG called MRG_PUB_MCAST_MoH. That MRG contains MOH_2(MOH)[Multicast] Music on Hold Server belong to DP_MoH, Enable Multicast Audio Sources, Multicast IP 239.1.1.1, port 16384, increment IP address, Max Hops = 1. For REG_MoH I have it set where it lists REG_BR1 G729, REG_BR2 G729. As I mentioned before, BR1 does in fact work and I cannot isolate the problem with BR2. Thanks for the assistance.. Mark On Jun 23, 2010, at 10:15 AM, Amy Ryan wrote: Mark, A couple quick suggestions -put CUCM PUB in a Device pool with a region that is set to g711-only -ensure you hop count is set to 1 -ensure in CUCM you only have one audio source configured HTH, Amy --- Amy Ryan - CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Mark m...@markholloway.com Date: Wed, 23 Jun 2010 10:04:29 -0700 To: OSL osl ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Streaming MoH from Router Flash If PUB is configured for multicast MoH, 239.1.1.1, port 16384, increment by IP, and I need to stream MoH from router flash on both BR1 and BR2, I'm having difficulty getting it to work on BR2. I have an MRG called PUB_MCAST_MoH and I've assigned it to both MRGL's MRGL_BR1 and MRGL_BR2 which are assigned to their respective Device Pools. On the BR1 router, under call-manager-fallback, I have set 'moh multicast 239.1.1.1 port 16384 route voice vlan ip loopback ip' and it's working. If I repeat the same process on BR2 it doesn't work. I've verified ip multicasting is set and dense mode is set. Any suggestions? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Live Record
I'm not sure for that specific problem. But I would make sure you have the following: - correct pilot number configured under telephony-service - a dial-peer that forwards all to cue - a working dial-peer for cue - conferencing configured - the cue pilot number configured in cue I'm leaning towards the cue pilot number not being configured. I believe its on a page called VM Configuration Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot Sent: Wednesday, June 23, 2010 11:22 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Live Record What would be the problem when trying to use Live Record and I get a, this mailbox is inactive? Is this a license issue? ___ 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] Live Record
Do you have conferencing resources configured? Jeff -Original Message- From: ccieid1ot [mailto:ccieid...@gmail.com] Sent: Wednesday, June 23, 2010 7:10 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Live Record Oh, and I have set the Live record pilot in CUE GUI, verified that voicemail live-record pilot set to 4250. On Wed, Jun 23, 2010 at 9:09 PM, ccieid1ot ccieid...@gmail.com wrote: Amy, SC Ph1 does have an active VM box since I can leave a VM and retrieved it. Jeff, Here's my config for Live Record Telephony-system live-record 4250 ephone-dn 20 oct number 4250 call-forward all 4220 dial-peer voice 4220 voip destination-pattern 42.. blah blah blah On Wed, Jun 23, 2010 at 2:55 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: I'm not sure for that specific problem. But I would make sure you have the following: - correct pilot number configured under telephony-service - a dial-peer that forwards all to cue - a working dial-peer for cue - conferencing configured - the cue pilot number configured in cue I'm leaning towards the cue pilot number not being configured. I believe its on a page called VM Configuration Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot Sent: Wednesday, June 23, 2010 11:22 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Live Record What would be the problem when trying to use Live Record and I get a, this mailbox is inactive? Is this a license issue? ___ 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] + for Calling Number
You could probably just create a Called Transformation Pattern with \+011 and not strip anything. Because it is more specific it would override the less-specific \+.! Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Sent: Monday, June 21, 2010 11:43 AM To: OSL osl Subject: [OSL | CCIE_Voice] + for Calling Number Ok, I'm stuck.. Could use some confirmation about my approach to + on the Calling Number when dialing to the PSTN. :) My phone displays the full E.164 in the upper right corner. If calls to the PSTN (conveniently) require + to be stripped from Calling Number in all cases, this is easy because I can assign a CSS Transformation on the gateway that strips the + from the Calling Number anytime a call egresses that gateway. However, if some call types don't require + in the Calling Number (local, LD) and others do (international) I cannot use the CSS on the gateway. So, assuming I do not have the CSS on the gateway to make the modification, would it be preferable to either create a Route Pattern or Translation Pattern (however you're building your dialing plan) and use the Calling Party Transformation within the RP or TP? For example, lets say at HQ, Local requires a 7 digit calling party number, LD requires 10 digits, and International requires +1 and 10 digits for Calling Party. Therefore my Route Pattern (or Translation Pattern) would look something like this. Patterns for all 3 call types are 9.[2-9]XX, 9.1[2-9]XX[2-9]XX, and 9.011! Partition = PT_HQ Local Calling Partying Transformation Mask = XXX LD Calling Partying Transformation Mask = XX International Calling Partying Transformation Mask = I would simply not modify the International RP/TP because my external phone mask is already +1XX. If BR1 is has the same requirements for Local and LD but also International does NOT require +1 in the Calling Number.. 9.[2-9]XX, 9.1[2-9]XX[2-9]XX, and 9.011! Partition = PT_BR1 Local Calling Partying Transformation Mask = XXX LD Calling Partying Transformation Mask = 1XX International Calling Partying Transformation Mask = 1XX ___ 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] sdspfarm
sdspfarm is how CME can register dsp resources. For SRST if you had configured the router's dsp resources to register with SUB then PUB, add 3rd statement that would then try the CME address. Then add sdspfarm statements under telephony-service to register the resources. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kalyan iyer Sent: Monday, June 21, 2010 1:03 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] sdspfarm Hi everyone, I am not clear what sdspfarm is used for? is this different from dspfarm? Is this what is needed when the site goes into SRST? Does this dip into the same MIP count that is being used for the dspfarm? Thanks Kalyan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] SD-BLF across WAN
Hi everyone, I am having trouble finding any information regarding how to configure this scenario (if it is even possible). There is an HQ Phone that needs to have a SD-BLF for a phone that is located at BR2 with CME. I know that CME offers a way to configure external access to phones using the watcher all command and a server ip address under presence. Is anyone aware of any good documentation for how to accomplish such a scenario? Do you know how to configure CUCM to allow this? Thanks in advance for your help, Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com mailto:jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html http://www.cisco.com/web/about/doing_business/legal/cri/index.html image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE Voice #26244
Congrats Ashar! From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Friday, June 18, 2010 11:46 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CCIE Voice #26244 Hello all, I went to Brussels yesterday and just an hour before learned that I am now officially CCIE Voice. It was my 2nd attempt but it was worth it. I learned a lot from my first attempt and it helped me build a better strategy for the 2nd. I am thankful to this wonderful list and IPExpert material which I used. Special thanks to Amy Ryan for her help whenever I needed. I am also grateful to my Study Partner Iwan Hoogendoorn, a triple CCIE and I was so lucky to have him as Study partner. I will never forget the way he use to make daily schedules and strictly made me follow those otherwise I am a lazy man..this number is for you Iwan! Few take home points for all those who will be making an attempt in coming days: 1 - Read the lab CAREFULLY (I made it Caps for a reason)..every word in a question is there for a reason! 2 - Do not rush! the mistakes you will make in first one hour will haunt you in the entire lab (unless you are lucky to figure out what went wrong) 3 - Do not spend too much time if something is not working - you can always come back to it. 4 - Note down sections and task which you are working and cross them as soon as you have completed it 5 - Call routing - This is how I did it, not necessarily helpful for you, I did call routing on a page first as what I am going to do at RL level, Pattern level etc..I configured everything first and then tested it one by one..took me 30 minutes to finish call routing 6 - Test everything you have done at least twice and as if it was configured by someone else and you are the proctor..I found one mistake while doing my 2nd check 7 - Save your config often, make sure before you leave that all gateways are up and registered to CUCM. I joined this list for my CCIE studies when I started my CCIE journey back in December 2009 but now I have decided to stick with it as I won't find such a nice bunch of people anywhere.. N.B: Above all, I loved my number..Digit '4' is my lucky number and Cisco made sure that I have enough of them.. :) Thank you all. It's party time now ;) Ashar Siddiqui CCIE#26244 (Voice) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SD-BLF across WAN
Thank you Amy! I am in the process of configuring a mock lab and will attempt this later. From: Amy Ryan [mailto:ar...@ipexpert.com] Sent: Friday, June 18, 2010 12:24 PM To: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SD-BLF across WAN Jeff, I am unsure if it is well documented anywhere, however in order to get this to work below is an overview. CUCM -SIP Trunk Security Profile that has Accept Presence Subscription enabled -Assign that to a SIP trunk using BR2 IP as the Destination Address and ensure proper SUBSCRIBE CSS is applied (will need to see RP to BR2 Phone) -Add RP for BR2 phone using SIP Trunk BR2-RTR Enable Presence and add allow watch to the DN HTH, Amy --- Amy Ryan - CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Jeff Price (jeffpric) jeffp...@cisco.com Date: Fri, 18 Jun 2010 13:47:46 -0500 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] SD-BLF across WAN Hi everyone, I am having trouble finding any information regarding how to configure this scenario (if it is even possible). There is an HQ Phone that needs to have a SD-BLF for a phone that is located at BR2 with CME. I know that CME offers a way to configure external access to phones using the watcher all command and a server ip address under presence. Is anyone aware of any good documentation for how to accomplish such a scenario? Do you know how to configure CUCM to allow this? Thanks in advance for your help, Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com mailto:jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html http://www.cisco.com/web/about/doing_business/legal/cri/index.html ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Better Voice Lab Locations
I believe it depends on your location, but normally they walk you to a local Cisco cafeteria with a voucher for your lunch (up to a certain price). -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jon1992 Sent: Friday, June 11, 2010 4:10 AM To: Amp; ccie voice Cc: ccie_voice@onlinestudylist.com; Mouhammad Nasser Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations During lunch are we stuck in the lab area or can we go and buy? -- From: Amp amccar...@cciequest.com Sent: Thursday, June 10, 2010 11:01 PM To: ccie voice cci...@gmail.com Cc: ccie_voice@onlinestudylist.com; Mouhammad Nasser engnasse...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations No not based on lunch. With the longer lunch time I will be able to have some time to think about what I have completed, what I need to complete, and if I need to change anything that I have done. Quoting ccie voice cci...@gmail.com: @Amp So you choose a lab location based on lunch? On Thu, Jun 10, 2010 at 1:14 PM, Amp amccar...@cciequest.com wrote: I live here in the RTP area but have decided to take the lab in San Jose. Here are my reasons: 1. Later Start Time 2. Longer Lunch 3. Better Weather 4. Just have a gut feeling about SJC Amp Quoting Jeff Garvas j...@cia.net: I heard that the West coast facility starts later, so someone east of that location would gain the time zone benefits as well as the late start. RTP supposedly starts first thing in the morning bright and early. 2010/6/9 Mouhammad Nasser engnasse...@hotmail.com Hi, I think it is better to take one that is closest to one's timezone! this will eliminate the factor of travel sickness, and one may go to exam awake enough! Regards, -- Hotmail: Trusted email with powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Gatekeeper Issue
Hi everyone, I am having trouble with my GK. I have made Bold what is the problem, but I can't seem to understand why I'm having this issue. I configured a tech-prefix of 1# under the Trunk configuration page. Here is the config - gatekeeper zone local ZONE_1 asccie.com 10.5.200.1 zone prefix ZONE_1 1* gw-priority 10 CUCM_GK_TRUNK_2 zone prefix ZONE_1 1* gw-priority 9 CUCM_GK_TRUNK_1 zone prefix ZONE_1 1* gw-priority 0 BR2_R3_GW BR1_R2_GW zone prefix ZONE_1 44* gw-priority 10 BR2_R3_GW zone prefix ZONE_1 44* gw-priority 0 BR1_R2_GW CUCM_GK_TRUNK_2 CUCM_GK_TRUNK_1 gw-type-prefix 1#* default-technology no shutdown Here is the debug gatekeeper main 10 output: May 25 23:55:58.011: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup May 25 23:55:58.187: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup R1(config-gk)# May 25 23:56:00.115: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup May 25 23:56:00.115: ////GK/gk_rassrv_arq: arqp=0x4AE0FB04,crv=0x19, answerCall=0 May 25 23:56:00.115: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/gk_dns_query: No Name servers May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo: (1#17752011001) Matched tech-prefix 1# May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo: (1#17752011001) Matched zone prefix 1 and remainder 7752011001 May 25 23:56:00.115: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4AE06200 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: matched zone is ZONE_1, and z_invian R1(config-gk)#amelen=0 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4AE06200 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: matched zone is ZONE_1, and z_outvianamelen=0 May 25 23:56:00.115: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo: (1#17752011001) tech-prefix gateway selection failed. May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x103) Here is the show gatekeeper call 10 output: May 26 00:02:15.899: ////GK/gk_call_new: src_endptp=0x4AE0F9F0, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160, crv=31, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x4925F3F8 May 26 00:02:15.899: ////GK/gk_call_find_endpts: NOT_FOUND May 26 00:02:15.899: ////GK/gk_call_new: checking for default (CLI) carrier for sep endpt 0x4AE0F9F0 May 26 00:02:15.899: //C6CEF7C380D2/C6CEF7C380D4/GK/gk_call_delete: callp=4AB57F54 May 26 00:02:15.899: //C6CEF7C380D2/C6CEF7C380D4/GK/gk_call_delete: c_callstate 0x0, c_resbw1 0, resbw2 0, c_reszp1 0x0, c_reszp2 0x0 Here is the show gatekeeper endpoints output: R1(config-gk)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 10.5.201.1 1720 10.5.201.1 61751 ZONE_1VOIP-GW H323-ID: BR1_R2_GW Voice Capacity Max.= Avail.= Current.= 0 10.5.202.1 1720 10.5.202.1 52635 ZONE_1VOIP-GW H323-ID: BR2_R3_GW E164-ID: 3001 E164-ID: 3002 Voice Capacity Max.= Avail.= Current.= 0 172.21.51.204 37257 172.21.51.204 32858 ZONE_1TERM H323-ID: CUCM_GK_TRUNK_1 172.21.51.205 34279 172.21.51.205 32814 ZONE_1TERM H323-ID: CUCM_GK_TRUNK_2 Total number of active registrations = 4 (The reason why 3001 and 3002 are registering with GK is the fact that I am using the secondary command on CME. For some reason that is still letting 3001/3002 register with the GK). Thanks in advance for your help! Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com mailto:jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142
Hi Ken, I actually tried doing that on CME, but it didn't work. After your email I tried again and I got the same result. I just set the Significant Digits to 4 and that didn't work either. I have the CSS on the trunk configured with a Translation Pattern that will strip the incoming 1#1775201.1001 to 1001. The CSS of the Translation Pattern has a CSS that can reach the phones. When I check the trunk using DNA, the call should route correctly. Therefore, I believe it's not even reaching CUCM. I think that something is going wrong on the GK and the call is never even making it to the CUCM routing logic. Thanks for the response Ken. Any other suggestions? Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Beck, Ken Sent: Tuesday, May 25, 2010 4:03 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142 Jeff on your CME ephone-dn's you'll need to put no-reg both after the number assignment Did you set the DDI on the GK Trunk to 4 or is it set to all. Try setting it to 4. Also please send a show gatek gw Regards, Ken -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Tuesday, May 25, 2010 3:54 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 51, Issue 142 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: I Passed CCIE#26088!!! (kerboute kerboute) 2. Re: I Passed CCIE#26088!!! (kerboute kerboute) 3. Re: I Passed CCIE#26088!!! (Ashar Siddiqui) 4. Gatekeeper Issue (Jeff Price (jeffpric)) -- Message: 1 Date: Tue, 25 May 2010 23:21:09 +0100 From: kerboute kerboute naoufal.kerbo...@cbi.ma Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!! To: ccie_voice@onlinestudylist.com Message-ID: 4bfc4d55.1060...@cbi.ma Content-Type: text/plain; charset=iso-8859-1 Congratulations brother On 05/25/2010 11:18 PM, Ehab Salem wrote: Dear Group, I Passed from the first shot Really thanks a lot for all your help...I really learned a lot from this kind study list J All what I want to say about my experience: the exam is easier than what we have in Volume 2...so it's all about Time Management, Strategy and Plan. I finished the lab in almost 6 hours. And spent the rest of time revising my configuration. I spent the week before the exam practicing on time management and putting a strategy and plan for each part in the exam that may comeand before the exam you should sleep well to start the exam with your full performance and energy. Anyway, it's over now for me...and wish u all the best J Thanks and best regards, * * *E**HAB **S**ALEM* Cisco Instructor | Sigma IT -- Egypt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100525/6c4 d1847/attachment-0001.htm -- Message: 2 Date: Tue, 25 May 2010 23:21:25 +0100 From: kerboute kerboute naoufal.kerbo...@cbi.ma Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!! To: esa...@sigma-it.net Cc: ccie_voice@onlinestudylist.com Message-ID: 4bfc4d65.8060...@cbi.ma Content-Type: text/plain; charset=iso-8859-1 Congratulations brother On 05/25/2010 11:18 PM, Ehab Salem wrote: Dear Group, I Passed from the first shot Really thanks a lot for all your help...I really learned a lot from this kind study list J All what I want to say about my experience: the exam is easier than what we have in Volume 2...so it's all about Time Management, Strategy and Plan. I finished the lab in almost 6 hours. And spent the rest of time revising my configuration. I spent the week before the exam practicing on time management and putting a strategy and plan for each part in the exam that may comeand before the exam you should sleep well to start the exam with your full performance and energy. Anyway, it's over now for me...and wish u all the best J Thanks and best regards, * * *E**HAB **S**ALEM* Cisco Instructor | Sigma IT -- Egypt ___ For more
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142
I figured out what I was doing wrong on CME at least. The command should be: number 3001 secondary 442321313001 no-reg both -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price (jeffpric) Sent: Tuesday, May 25, 2010 4:26 PM To: Beck, Ken; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142 Hi Ken, I actually tried doing that on CME, but it didn't work. After your email I tried again and I got the same result. I just set the Significant Digits to 4 and that didn't work either. I have the CSS on the trunk configured with a Translation Pattern that will strip the incoming 1#1775201.1001 to 1001. The CSS of the Translation Pattern has a CSS that can reach the phones. When I check the trunk using DNA, the call should route correctly. Therefore, I believe it's not even reaching CUCM. I think that something is going wrong on the GK and the call is never even making it to the CUCM routing logic. Thanks for the response Ken. Any other suggestions? Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Beck, Ken Sent: Tuesday, May 25, 2010 4:03 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142 Jeff on your CME ephone-dn's you'll need to put no-reg both after the number assignment Did you set the DDI on the GK Trunk to 4 or is it set to all. Try setting it to 4. Also please send a show gatek gw Regards, Ken -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Tuesday, May 25, 2010 3:54 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 51, Issue 142 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: I Passed CCIE#26088!!! (kerboute kerboute) 2. Re: I Passed CCIE#26088!!! (kerboute kerboute) 3. Re: I Passed CCIE#26088!!! (Ashar Siddiqui) 4. Gatekeeper Issue (Jeff Price (jeffpric)) -- Message: 1 Date: Tue, 25 May 2010 23:21:09 +0100 From: kerboute kerboute naoufal.kerbo...@cbi.ma Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!! To: ccie_voice@onlinestudylist.com Message-ID: 4bfc4d55.1060...@cbi.ma Content-Type: text/plain; charset=iso-8859-1 Congratulations brother On 05/25/2010 11:18 PM, Ehab Salem wrote: Dear Group, I Passed from the first shot Really thanks a lot for all your help...I really learned a lot from this kind study list J All what I want to say about my experience: the exam is easier than what we have in Volume 2...so it's all about Time Management, Strategy and Plan. I finished the lab in almost 6 hours. And spent the rest of time revising my configuration. I spent the week before the exam practicing on time management and putting a strategy and plan for each part in the exam that may comeand before the exam you should sleep well to start the exam with your full performance and energy. Anyway, it's over now for me...and wish u all the best J Thanks and best regards, * * *E**HAB **S**ALEM* Cisco Instructor | Sigma IT -- Egypt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100525/6c4 d1847/attachment-0001.htm -- Message: 2 Date: Tue, 25 May 2010 23:21:25 +0100 From: kerboute kerboute naoufal.kerbo...@cbi.ma Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!! To: esa...@sigma-it.net Cc: ccie_voice@onlinestudylist.com Message-ID: 4bfc4d65.8060...@cbi.ma Content-Type: text/plain; charset=iso-8859-1 Congratulations brother On 05/25/2010 11:18 PM, Ehab Salem wrote: Dear Group, I Passed from the first shot Really thanks a lot for all your help...I really learned a lot from this kind study list J All what I want to say about my experience: the exam is easier than what we have in Volume 2...so it's all about Time Management, Strategy and Plan. I finished the lab in almost 6 hours. And spent the rest of time revising my configuration. I spent the week before
Re: [OSL | CCIE_Voice] Gatekeeper Issue
Hi everyone, Does anyone have any other ideas? Have you seen this before? It's something with the Tech prefix and/or gateway selection in GK, but I'm not figuring this one out. The correct digits are sent, CUCM (according to DNA) should process the call right, and the right codecs should be negotiated. I'm lost. Google isn't helping much either. Please help! Thanks, Jeff -Original Message- From: Jeff Price (jeffpric) Sent: Tuesday, May 25, 2010 4:33 PM To: Jeff Price (jeffpric); Beck, Ken; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142 I figured out what I was doing wrong on CME at least. The command should be: number 3001 secondary 442321313001 no-reg both -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price (jeffpric) Sent: Tuesday, May 25, 2010 4:26 PM To: Beck, Ken; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142 Hi Ken, I actually tried doing that on CME, but it didn't work. After your email I tried again and I got the same result. I just set the Significant Digits to 4 and that didn't work either. I have the CSS on the trunk configured with a Translation Pattern that will strip the incoming 1#1775201.1001 to 1001. The CSS of the Translation Pattern has a CSS that can reach the phones. When I check the trunk using DNA, the call should route correctly. Therefore, I believe it's not even reaching CUCM. I think that something is going wrong on the GK and the call is never even making it to the CUCM routing logic. Thanks for the response Ken. Any other suggestions? Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Beck, Ken Sent: Tuesday, May 25, 2010 4:03 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 51, Issue 142 Jeff on your CME ephone-dn's you'll need to put no-reg both after the number assignment Did you set the DDI on the GK Trunk to 4 or is it set to all. Try setting it to 4. Also please send a show gatek gw Regards, Ken -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Tuesday, May 25, 2010 3:54 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 51, Issue 142 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: I Passed CCIE#26088!!! (kerboute kerboute) 2. Re: I Passed CCIE#26088!!! (kerboute kerboute) 3. Re: I Passed CCIE#26088!!! (Ashar Siddiqui) 4. Gatekeeper Issue (Jeff Price (jeffpric)) -- Message: 1 Date: Tue, 25 May 2010 23:21:09 +0100 From: kerboute kerboute naoufal.kerbo...@cbi.ma Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!! To: ccie_voice@onlinestudylist.com Message-ID: 4bfc4d55.1060...@cbi.ma Content-Type: text/plain; charset=iso-8859-1 Congratulations brother On 05/25/2010 11:18 PM, Ehab Salem wrote: Dear Group, I Passed from the first shot Really thanks a lot for all your help...I really learned a lot from this kind study list J All what I want to say about my experience: the exam is easier than what we have in Volume 2...so it's all about Time Management, Strategy and Plan. I finished the lab in almost 6 hours. And spent the rest of time revising my configuration. I spent the week before the exam practicing on time management and putting a strategy and plan for each part in the exam that may comeand before the exam you should sleep well to start the exam with your full performance and energy. Anyway, it's over now for me...and wish u all the best J Thanks and best regards, * * *E**HAB **S**ALEM* Cisco Instructor | Sigma IT -- Egypt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100525/6c4 d1847/attachment-0001.htm -- Message: 2 Date: Tue, 25 May 2010 23:21:25 +0100 From: kerboute kerboute naoufal.kerbo...@cbi.ma Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!! To: esa...@sigma
Re: [OSL | CCIE_Voice] Unable to Bind L3 to CCM
Thanks guys! (Slaps forehead) From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] Sent: Friday, May 21, 2010 4:01 PM To: Jeff Price (jeffpric); CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Unable to Bind L3 to CCM You forgot service mgcp at the end of your pri-group command in controller config. Sent from my Verizon Wireless Phone - Reply message - From: Jeff Price (jeffpric) jeffp...@cisco.com Date: Fri, May 21, 2010 6:55 pm Subject: [OSL | CCIE_Voice] Unable to Bind L3 to CCM To: CCIE Voice Maillist ccie_voice@onlinestudylist.com Hey everyone, Have you ever seen a situation where you can register a MGCP GW to CUCM but you are unable to bind L3 to CCM in IOS? Here's what I see: R1(config-if)#isdn bind-l3 ? q931 Select IOS Q.931 R1(config-if)# Here is my config: R1#show run Building configuration... Current configuration : 3127 bytes ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname R1 ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-24.T.bin boot-end-marker ! logging message-counter syslog enable password 7 110A1A0C12 ! no aaa new-model network-clock-participate wic 0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 10.5.200.1 ! ip dhcp pool HQ_PHONES network 10.5.200.0 255.255.255.0 option 150 ip 172.21.51.204 default-router 10.5.200.1 ! ! no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-ni ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! voice-card 0 ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller T1 0/0/0 pri-group timeslots 1-3,24 ! controller T1 0/0/1 ! ! ! ! ! interface GigabitEthernet0/0 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$ no ip address duplex auto speed auto ! interface GigabitEthernet0/0.102 encapsulation dot1Q 102 native ip address 172.21.51.196 255.255.255.224 ! interface GigabitEthernet0/0.150 encapsulation dot1Q 150 ip address 10.5.100.1 255.255.255.0 ! interface GigabitEthernet0/0.250 encapsulation dot1Q 250 ip address 10.5.200.1 255.255.255.0 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn outgoing display-ie no cdp enable ! interface Serial0/1/0 no ip address encapsulation frame-relay IETF tx-ring-limit 128 tx-queue-limit 128 serial restart-delay 0 frame-relay lmi-type ansi ! interface Serial0/1/0.1 point-to-point ip address 162.5.101.1 255.255.255.0 ip ospf mtu-ignore frame-relay interface-dlci 201 ! interface Serial0/1/0.2 point-to-point ip address 162.5.102.1 255.255.255.0 ip ospf mtu-ignore frame-relay interface-dlci 202 ! router ospf 1 log-adjacency-changes network 10.5.100.0 0.0.0.255 area 0 network 10.5.200.0 0.0.0.255 area 0 network 162.5.101.0 0.0.0.255 area 0 network 162.5.102.0 0.0.0.255 area 0 network 172.5.100.0 0.0.0.255 area 0 network 172.21.51.0 0.0.0.255 area 0 ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 172.21.51.193 ip http server no ip http secure-server ! ! ! nls resp-timeout 1 cpd cr-id 1 ! ! ! ! ! ! control-plane ! ! ! voice-port 0/0/0:23 ! ccm-manager redundant-host 172.21.51.204 ccm-manager mgcp ! mgcp mgcp call-agent 172.21.51.205 service-type mgcp version 0.1 mgcp fax t38 ecm mgcp bind control source-interface GigabitEthernet0/0.250 mgcp bind media source-interface GigabitEthernet0/0.250 ! mgcp profile default ! ! ! ! ! ! gatekeeper shutdown ! line con 0 exec-timeout 0 0 logging synchronous terminal-type mon line aux 0 line vty 0 4 exec-timeout 0 0 password 7 110A1A0C12 logging synchronous login terminal-type mon ! scheduler allocate 2 1000 end Here is the status of CCM registration: R1(config)#do show ccm MGCP Domain Name: R1 PriorityStatus Host Primary Registered 172.21.51.205 First BackupBackup Ready 172.21.51.204 Second Backup None Current active Call Manager:172.21.51.205 Backhaul/Redundant link port: 2428 Failover Interval: 30 seconds Keepalive Interval: 15 seconds Last keepalive sent:00:03:46 UTC May 22 2010 (elapsed time: 00:00:06) Last MGCP traffic time: 00:03:46 UTC May 22 2010 (elapsed time: 00:00:06) Last failover time: 23:59:51 UTC May 21 2010 from (172.21.51.205) Last switchback time: 00:00:21 UTC May 22 2010 from (172.21.51.204) Switchback mode:Graceful MGCP Fallback mode: Not Selected Last MGCP Fallback start time: None Last
Re: [OSL | CCIE_Voice] Problem ISDN Lab 5C Vol1
Hi Naoufal, Ensure that you have enough DSP resources to complete the call. Try decreasing the number of pri timeslots if you are unsure. Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kerboute kerboute Sent: Tuesday, May 11, 2010 3:51 PM To: CCIE Voice Maillist Subject: [OSL | CCIE_Voice] Problem ISDN Lab 5C Vol1 Hi, I'm working on lab 5C Vol1 and I'm having a troube with the HQ T1, when I make a call to 911 I've got the message below: May 12 02:46:26.355: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 May 12 02:46:26.355: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x2 0x1, Calling num 2123945003 May 12 02:46:26.359: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Called num 911 May 12 02:46:26.359: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0089 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x2181, '2123945003' Plan:ISDN, Type:National Called Party Number i = 0x80, '911' Plan:Unknown, Type:Unknown May 12 02:46:26.391: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8089 Channel ID i = 0xA98381 Exclusive, Channel 1 May 12 02:46:26.403: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8 callref = 0x8089 Progress HQ-RTR#Ind i = 0x8188 - In-band info or appropriate now available May 12 02:46:26.443: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x0089 Cause i = 0x80AC - Requested circuit/channel not available May 12 02:46:26.451: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x8089 May 12 02:46:26.455: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x0089 However I can see the all 3 channel are up and T1 are multiframe-established. Any Idea? Thank you Naoufal ___ 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] Vol 2 Lab 1 - ISDN Restart Errors
To fix this problem in the past, I simply restarted the PSTN router and it works. However, I do realize that this isn't ideal in the real world or the exam and I was never able to figure out the proper way to fix this. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve Denney (stdenney) Sent: Thursday, May 06, 2010 12:27 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors Hi, Seeing some errors today that I haven't encountered before in any other lab...wh! :) I'm working on Vol 2 Lab 1 Question 4.1, and trying to get calls from the PSTN working into HQ. Pretty straightforward stuff, except the calls never seem to get out of the PSTN router. When dialing the HQ phone from the PSTN phone (regardless of line selected), I get the following debug isdn q931 errors from the PSTN router: May 6 23:11:09.243: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Calling num 911 May 6 23:11:09.243: ISDN Se0/3/0:23 **ERROR**: CCPMSG_OutCall: fails with cause 0x22 And every 30 seconds, I see the same batch of 4 ISDN Restart messages, like this (also from the PSTN router): May 6 23:12:04.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 May 6 23:12:05.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 May 6 23:12:06.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 May 6 23:12:07.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 Show isdn status on the PSTN router looks normal for this interface: ISDN Serial0/3/0:23 interface *** Network side configuration *** dsl 1, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 1 CCBs = 0 The Free Channel Mask: 0x8000 Number of L2 Discards = 0, L2 Session ID = 0 Attaching show run and show isdn status as well for the HQ router (the other end) just for troubleshooting completeness, but there's no indication of anything amiss, nor any debug messages at all, on the HQ router. The call never gets that far. I started this morning on Voice Pod 11 and hit this. Ryan was kind enough to move me over to Voice Pod 16, but I'm hitting the same issue here. OSL archive and Google search turned up nothing concrete, other than a general theme of it sounds like your telco / carrier has issues. :) Any ideas? Cheers and TIA, sd ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors
This is true. I forgot about this. Normally it has to do with oversubscribing the DSP resources. Meaning, make sure you aren't using all of them and the PSTN has enough to use. Try decreasing the amount of channels you create under the pri-group timeslots command. Good point. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 Sent: Thursday, May 06, 2010 12:39 PM To: Steve Denney (stdenney) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors cause 0x22 - generaly means no channel availableplease checj you have right number of slots defined in pri-group timeslots statement and double check your PSTN router for same. On Thu, May 6, 2010 at 12:26 PM, Steve Denney (stdenney) stden...@cisco.com wrote: Hi, Seeing some errors today that I haven't encountered before in any other lab...wh! :) I'm working on Vol 2 Lab 1 Question 4.1, and trying to get calls from the PSTN working into HQ. Pretty straightforward stuff, except the calls never seem to get out of the PSTN router. When dialing the HQ phone from the PSTN phone (regardless of line selected), I get the following debug isdn q931 errors from the PSTN router: May 6 23:11:09.243: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Calling num 911 May 6 23:11:09.243: ISDN Se0/3/0:23 **ERROR**: CCPMSG_OutCall: fails with cause 0x22 And every 30 seconds, I see the same batch of 4 ISDN Restart messages, like this (also from the PSTN router): May 6 23:12:04.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 May 6 23:12:05.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 May 6 23:12:06.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 May 6 23:12:07.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 Show isdn status on the PSTN router looks normal for this interface: ISDN Serial0/3/0:23 interface *** Network side configuration *** dsl 1, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 1 CCBs = 0 The Free Channel Mask: 0x8000 Number of L2 Discards = 0, L2 Session ID = 0 Attaching show run and show isdn status as well for the HQ router (the other end) just for troubleshooting completeness, but there's no indication of anything amiss, nor any debug messages at all, on the HQ router. The call never gets that far. I started this morning on Voice Pod 11 and hit this. Ryan was kind enough to move me over to Voice Pod 16, but I'm hitting the same issue here. OSL archive and Google search turned up nothing concrete, other than a general theme of it sounds like your telco / carrier has issues. :) Any ideas? Cheers and TIA, sd ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] cBarge on CME not working - says messageFailed to setup barge
You also need to configure some ad-hoc conference ephone-dn(s). Then this should work. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 Sent: Thursday, April 22, 2010 8:29 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] cBarge on CME not working - says messageFailed to setup barge cBarge on CME - - eph-dn octoline - conf hardware in telephony service - privacy off on ephone to be bacrged into - ephon-temp remote-in-use to have cBArge key - apply eph-temp to eph Error message says message Failed to setup barge ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VLAN port speed and duplex
For the exam, no. All routing is pre-configured and if there is any issues you need to get the proctor involved. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bo Gao Sent: Thursday, April 22, 2010 10:51 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] VLAN port speed and duplex Hi guys, I am just starting my lab prep, and I am at lab 1A. When a port is combined with both voice and data, do we need to manually set port speed and duplex(i.e., 10/half) for the exam? Thank you, Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] OSL CUCM DHCP Server
This may seem silly, but it slips the mind sometimes. Did you remember to configure the switchports with the proper vlans? -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of wilson.sam...@usc-bt.com Sent: Tuesday, April 13, 2010 12:12 PM To: bkvalent...@gmail.com; ccie_voice-boun...@onlinestudylist.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] OSL CUCM DHCP Server Thanks for your response. Yes, the ip helper address is correct, infact on the router I do see that phone is looking for a DHCP Server, however the Server seems to be silent. I'm really stuck on this and don't seem to get any clue to get out of this. It must be something really silly, that I'm missing it Regards -Original Message- From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] Sent: Tuesday, April 13, 2010 2:58 PM To: Samuel, Wilson; ccie_voice-boun...@onlinestudylist.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] OSL CUCM DHCP Server Have IP helper address pointing to the CUCM Pub IP? Make sure the default gateway offered is correct. I won't tell you how I learned that one. Brian Sent via BlackBerry from T-Mobile -Original Message- From: wilson.sam...@usc-bt.com Date: Tue, 13 Apr 2010 13:55:51 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] OSL CUCM DHCP Server Hi All, I'm trying to setup DHCP Server on CUCM (ver 7.0 ), however for some reason its not offering any ACK or DHCPOFFR for any DHCP Request on the network. I have already disabled CSA on the CUCM, anything else I missed? Appreciate any help on the aspect. Kind Regards Wilson Samuel ___ 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] Inter-site calls Not routing over GK trunk
Ashar - How is the number coming into the gateway? Are you stripping the tech-prefix so that the CUCME will recognize the number? You can use the debug h225 asn1 command (a lot of output) to figure this one how, however this assumes the GK config is correct. Otherwise the h225 debug will not show anything besides registration, because the GK will never attempt to route the call to CUCME. For example, the number would come in to CUCME as 1#3001 and you would need a translation rule to strip the 1# so that CUCME will recognize. Also, if you are supporting abbreviated dialing from BR2 to HQ and BR1 you will need to prepend the tech-prefix prior to sending and then strip when CUCM receives it using a translation pattern. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Monday, April 05, 2010 9:37 AM To: CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: [OSL | CCIE_Voice] Inter-site calls Not routing over GK trunk Importance: High Hello guys, I am trying to route calls between HQ and BR1 to a CME site BR2 using 4-digit extension over the GK trunk. Gatekeeper is on HQ and its a MGCP gateway. The solution below is working fine as expected if we don't use no-reg primary at ephone-dn. This means phones are registering with their E.164 id to gatekeeper. I don't think this should happen. Phones should not register with Gatekeeper isn't it? I checked show gatekeeper endpoints and it was showing me E164 id 3001 and E164 id 3002 of BR2. Everything was working fine, calls can be routed over the GK trunk to CUCME site when a user dial 3001/3002 from HQ or BR1. But as soon as I entered no-reg primary command to the ephone-dn and did a reset, the GK trunk is not working and calls from HQ and BR1 are routed thru backup local GW with full E164 number. I am confiused a bit because CUCME is regsitered with 1# and GK knows 3* numbers sits at CUCME site. I also changed 3* to 3... on Gatekeeper but it isn't working. My question is what is an expected behaviour? do we need that no-reg primary command on ephone-dn? if not then the solution is working fine but if yes we need it on ephone-dn then please let me know where I am wrong as calls are not routed thru the GK trunk. Do I need to prefix anything at CUCM on Route List details- GK trunk like 1# or something? Calls from SC to HQ and SB are working fine as normal over the GK trunk. I have made a Region GK-729 with 729 codec within all sites. That region is in the DP of GK-TRUNK. Here is HQ config: (My subscriber is down at the moment so you will only see one VOIP-GW in outputs) gatekeeper zone local GK ccievoice.com 10.10.110.1 zone local CUCME ccievoice.com zone prefix GK 5... gw-priority 10 gk-trunk_2 zone prefix GK 5... gw-priority 9 gk-trunk_1 zone prefix GK 1... gw-priority 10 gk-trunk_2 zone prefix GK 1... gw-priority 9 gk-trunk_1 zone prefix CUCME 3* gw-type-prefix 1#* default-technology no shutdown ! HQ#sh gatek endpoints GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 10.10.110.3 1720 10.10.110.3 60441 GKH323-GW H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 10.10.210.101720 10.10.210.1044782 GKVOIP-GW H323-ID: gk-trunk_1 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 2 HQ# HQ#sh gatek zone prefix ZONE PREFIX TABLE = GK-NAME E164-PREFIX --- --- GK5... GK1... CUCME3* HQ# BR2#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 GK ipaddr 10.10.110.1 1719 h323-gateway voip h323-id CUCME h323-gateway voip tech-prefix 1# BR2#sh ephone ephone-1[0] Mac:0017.9497.1F89 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 12/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:1 IP:192.168.10.55 50512 7961 keepalive 42 max_line 6 button 1: dn 1 number 3001 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 3 number 3003 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared privacy button is enabled Preferred Codec: g711ulaw Username: scph1 ephone-2[1] Mac:0017.E089.7382 TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 12/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:1 IP:192.168.10.56 51267 7961 keepalive 179
Re: [OSL | CCIE_Voice] Converting CUE to Integrate with UCM
To my knowledge, it is that simple. Use this command on CUE software install clean url ftp://ip-address/license-filename.pkg user user pass pass and have the license file available on an FTP server. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Monday, March 22, 2010 4:12 PM To: osl osl Subject: [OSL | CCIE_Voice] Converting CUE to Integrate with UCM Is the only requirement to go from CME integration to UCM to load the proper license file? This is my companies equipment not proctor labs. I would like to be able to move back and forth similar to proctor labs but am unsure it is as easy as just loading the proper license file. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1
Are you by any chance running a VPN from routers to PSTN? I've noticed this causes distortion some times. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Granat Sent: Thursday, March 11, 2010 9:46 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1 Perhaps this is something simple that I am overlooking but I have the generic setup running in my home lab with 3 gateways and one PSTN router. 2 of the gateways are T1 and one is E1. The PSTN router is also running CME with a 7960 to simulate PSTN destinations. Calls from any site to the PSTN phone are fine. Calls between T1 sites are fine. Calls between T1 and E1 sites are distorted, like the gain is way too high. I tried playing with the gain on the voice-port but no luck. I'm not finding much online or in Cisco docs. Any suggestions? Thanks, Jason http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Reset CUCM 7.x to initial config on VM Ware
Wael, After you do a fresh install, take a snapshot in VMWare. Then you can always revert back to the snapshot without having to un-configure everything you have done. In VMWare you can normally only have 1 snapshot, unless you have one of the more advanced versions, so for all other backups your would have to use tftp, ftp, and DRS on CUCM in order to save configs and states. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wael Agina Sent: Wednesday, March 10, 2010 10:35 AM To: OSL Group Subject: [OSL | CCIE_Voice] Reset CUCM 7.x to initial config on VM Ware Dear All, I have CUCM 7, CU7, CUCCX 7 running on VM ware. My qeustions is how to reset all its setting to initial setup without having any config ? Also if possible how to make a backup from a certain config on each server. Sorry for simple qeustions but i am new to vmware world and a friend give me the images. -- Thanks and Best Regards, Wael Agina ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CM to GK BRQ behavior
Hi Scott, Go to Enterprise parameters and change the Advertise G722 to false. Hope this helps.\ Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo Sent: Monday, March 08, 2010 12:37 PM To: Scott ODonnell Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] CM to GK BRQ behavior Hello Scott, What is BRQ?? Am having siimilar issue how did you combat it. thanks On Wed, Jul 8, 2009 at 9:03 PM, Scott ODonnell scott.odonn...@gmail.com wrote: I'm seeing something strange in making calls from CM to CME via GK. I've enabled the BRQ service parameter in CM. I've included bandwidth total default 16 in my gk config and did a shut/no shut When I make calls from CM to CME the deb h225 asn1 shows (I think) that 128k is being requested. Am I missing something obivous here ? Currently all my calls get rejected from the GK and go via the HQ GW. If I remove the bandwidth command from the Gatekeeper config, the call works using g729. - Scott ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Voice Vol 1 Lab 5c task 5.8
Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo: (9011916745738932) Tech-prefix match failed. Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo: (9011916745738932) unresolved zone prefix, using source zone US You need to strip the 9011 before sending this to the gatekeeper. zone prefix SPAIN 34* zone prefix PSTN-WAN 91* Hope this helps. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIETalk.com Sent: Tuesday, February 23, 2010 10:10 AM To: Vik Malhi Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] Voice Vol 1 Lab 5c task 5.8 Thanks Vik. Here is the debug output Calling India HQ-RTR# Feb 23 17:45:35.916: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Feb 23 17:45:35.916: ////GK/gk_rassrv_arq: arqp=0x67399F7C,crv=0x4, answerCall=0 Feb 23 17:45:35.920: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/gk_dns_query: No Name servers Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo: (9011916745738932) Tech-prefix match failed. Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo: (9011916745738932) unresolved zone prefix, using source zone US Feb 23 17:45:35.920: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x67A6DFFC Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_arq_select_viazone: matched zone is US, and HQ-RTR#z_invianamelen=0 Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x67A6DFFC Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_arq_select_viazone: matched zone is US, and z_outvianamelen=0 Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo: No tech prefix Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo: Alias not found Feb 23 17:45:35.920: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo: (9011916745738932) default-tech gateway selection failed, status = 0x805 Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/rassrv_get_addrinfo: (9011916745738932) unknown address and no default technology defined. Feb 23 17:45:35.920: //004F983D0400/004F983D0400/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x107) Calling Spain Feb 23 17:45:08.956: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Feb 23 17:45:08.956: ////GK/gk_rassrv_arq: arqp=0x6816755C,crv=0x3, answerCall=0 Feb 23 17:45:08.956: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Feb 23 17:45:08.956: //806F802D0300/806F802D0300/GK/gk_dns_query: No Name servers Feb 23 17:45:08.956: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo: (90113432141891) Tech-prefix match failed. Feb 23 17:45:08.956: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo: (90113432141891) unresolved zone prefix, using source zone US Feb 23 17:45:08.956: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Feb 23 17:45:08.956: //806F802D0300/806F802D0300/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x67A6DFFC Feb 23 17:45:08.956: //806F802D0300/806F802D0300/GK/rassrv_arq_select_viazone: matched zone is US, and z_in HQ-RTR#vianamelen=0 Feb 23 17:45:08.956: //806F802D0300/806F802D0300/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x67A6DFFC Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/rassrv_arq_select_viazone: matched zone is US, and z_outvianamelen=0 Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo: No tech prefix Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo: Alias not found Feb 23 17:45:08.960: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo: (90113432141891) default-tech gateway selection failed, status = 0x805 Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/rassrv_get_addrinfo: (90113432141891) unknown address and no default technology defined. Feb 23 17:45:08.960: //806F802D0300/806F802D0300/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x107) Gatekeeper config HQ-RTR#sh run | begin gatekeeper gatekeeper zone local SPAIN ipexpert.com 10.10.110.1 zone local US ipexpert.com zone remote PSTN-WAN ipexpert.com 10.10.100.2 1719 zone prefix SPAIN 34* zone prefix
[OSL | CCIE_Voice] Phone boot problem
Hi everyone, I have a pair of 7945 phones that will not startup for me. I have been using these phones for the past 3 weeks or so and had no problems. I have configured DHCP and their switchports as follows: On the 3750 switch that the phones are directly connected to: interface GigabitEthernet1/0/13 description HQ PHONE 1 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast ! interface GigabitEthernet1/0/14 description HQ PHONE 2 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast ! interface GigabitEthernet1/0/15 description HQ PHONE 3 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast On the router that supplies the DHCP: ip dhcp excluded-address 10.5.200.1 ip dhcp pool HQ_PHONES network 10.5.200.0 255.255.255.0 option 150 ip 172.21.51.204 default-router 10.5.200.1 On port g1/0/13 of the switch I have a 7960 that has successfully registered with CUCM, but the two 7945 keep loading up to a Upgrading screen and then re-cycling. The CUCM PUB is running the TFTP service and the CM Group (for registration time only) is configured to use PUB then SUB so that auto-registration will work correctly. Any ideas why this may be happening? Thanks, Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com mailto:jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html http://www.cisco.com/web/about/doing_business/legal/cri/index.html image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Phone boot problem
Hi everyone, I hooked up a spare 7960 to this port and it has registered successfully.So what would be up with the 7945s that it can't register? Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price (jeffpric) Sent: Tuesday, February 23, 2010 10:32 AM To: OSL Group Subject: [OSL | CCIE_Voice] Phone boot problem Hi everyone, I have a pair of 7945 phones that will not startup for me. I have been using these phones for the past 3 weeks or so and had no problems. I have configured DHCP and their switchports as follows: On the 3750 switch that the phones are directly connected to: interface GigabitEthernet1/0/13 description HQ PHONE 1 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast ! interface GigabitEthernet1/0/14 description HQ PHONE 2 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast ! interface GigabitEthernet1/0/15 description HQ PHONE 3 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast On the router that supplies the DHCP: ip dhcp excluded-address 10.5.200.1 ip dhcp pool HQ_PHONES network 10.5.200.0 255.255.255.0 option 150 ip 172.21.51.204 default-router 10.5.200.1 On port g1/0/13 of the switch I have a 7960 that has successfully registered with CUCM, but the two 7945 keep loading up to a Upgrading screen and then re-cycling. The CUCM PUB is running the TFTP service and the CM Group (for registration time only) is configured to use PUB then SUB so that auto-registration will work correctly. Any ideas why this may be happening? Thanks, Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Phone boot problem
Amy, I actually have already factory reset the phones and manually added to the CUCM database. I will try again though, because I may have made some sort of mistake while doing so. Thanks for your help. Jeff From: Amy Ryan [mailto:ar...@ipexpert.com] Sent: Tuesday, February 23, 2010 12:06 PM To: Jeff Price (jeffpric); OSL Group Subject: Re: [OSL | CCIE_Voice] Phone boot problem Jeff, Can you try to manually add a 7945 to the UCM database with whatever the existing protocol the phone was using in the previous weeks? It sounds like the phone is trying to switch firmware and is having problems doing so. In the event that this fails, you could try to restore factory defaults by using the following steps: 1. unplug the Ethernet connection (an external power source if required) 2. hold down the # key and while holding, plug in the Ethernet connection (and power as needed) 3. When the Line Buttons to the right of the LCD display begin to flash amber, release the # key and press the numbers on the dialpad in the following sequence: 123456789*0# The phone will then initiate the factory reset and should download intended firmware, etc. This process can take 4-8 minutes roughly. HTH, Amy --- Amy Ryan - CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com . From: Jeff Price (jeffpric) jeffp...@cisco.com Date: Tue, 23 Feb 2010 12:41:29 -0600 To: Jeff Price (jeffpric) jeffp...@cisco.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Phone boot problem Hi everyone, I hooked up a spare 7960 to this port and it has registered successfully.So what would be up with the 7945s that it can't register? Jeff From: ccie_voice-boun...@onlinestudylist.com [ mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price (jeffpric) Sent: Tuesday, February 23, 2010 10:32 AM To: OSL Group Subject: [OSL | CCIE_Voice] Phone boot problem Hi everyone, I have a pair of 7945 phones that will not startup for me. I have been using these phones for the past 3 weeks or so and had no problems. I have configured DHCP and their switchports as follows: On the 3750 switch that the phones are directly connected to: interface GigabitEthernet1/0/13 description HQ PHONE 1 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast ! interface GigabitEthernet1/0/14 description HQ PHONE 2 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast ! interface GigabitEthernet1/0/15 description HQ PHONE 3 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast On the router that supplies the DHCP: ip dhcp excluded-address 10.5.200.1 ip dhcp pool HQ_PHONES network 10.5.200.0 255.255.255.0 option 150 ip 172.21.51.204 default-router 10.5.200.1 On port g1/0/13 of the switch I have a 7960 that has successfully registered with CUCM, but the two 7945 keep loading up to a Upgrading screen and then re-cycling. The CUCM PUB is running the TFTP service and the CM Group (for registration time only) is configured to use PUB then SUB so that auto-registration will work correctly. Any ideas why this may be happening? Thanks, Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab
Re: [OSL | CCIE_Voice] Phone boot problem
Hi Chris, SCCP45.8-4-1S is the phone load for the 7945. I am unable to get to the settings page as the phone doesn't boot up far enough. Thanks, Jeff From: Christopher Clouse [mailto:christopherc_56...@hotmail.com] Sent: Tuesday, February 23, 2010 11:33 AM To: Jeff Price (jeffpric) Subject: Re: [OSL | CCIE_Voice] Phone boot problem Check in CM under the device defaults for the phone loads and see what the load it is attempting to grab. If you can get as far as the setttings page on the phone you should be able to see it under device settings. From: Jeff Price (jeffpric) mailto:jeffp...@cisco.com Sent: Tuesday, February 23, 2010 1:31 PM To: Christopher Clouse mailto:christopherc_56...@hotmail.com Subject: RE: [OSL | CCIE_Voice] Phone boot problem CUCM is 7. How can I find out the software version of the phone? I don't believe the boot-up process gets far enough for me to access this information. It's worth noting that I factory reset the phones in an attempt to get them to work. Jeff From: Christopher Clouse [mailto:christopherc_56...@hotmail.com] Sent: Tuesday, February 23, 2010 11:29 AM To: Jeff Price (jeffpric) Subject: Re: [OSL | CCIE_Voice] Phone boot problem What version of software is on the phone and what is on the CM? ~Chris From: Jeff Price (jeffpric) mailto:jeffp...@cisco.com Sent: Tuesday, February 23, 2010 12:41 PM To: Jeff Price (jeffpric) mailto:jeffp...@cisco.com ; OSL Group mailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Phone boot problem Hi everyone, I hooked up a spare 7960 to this port and it has registered successfully.So what would be up with the 7945s that it can't register? Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price (jeffpric) Sent: Tuesday, February 23, 2010 10:32 AM To: OSL Group Subject: [OSL | CCIE_Voice] Phone boot problem Hi everyone, I have a pair of 7945 phones that will not startup for me. I have been using these phones for the past 3 weeks or so and had no problems. I have configured DHCP and their switchports as follows: On the 3750 switch that the phones are directly connected to: interface GigabitEthernet1/0/13 description HQ PHONE 1 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast ! interface GigabitEthernet1/0/14 description HQ PHONE 2 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast ! interface GigabitEthernet1/0/15 description HQ PHONE 3 switchport access vlan 150 switchport mode access switchport voice vlan 250 spanning-tree portfast On the router that supplies the DHCP: ip dhcp excluded-address 10.5.200.1 ip dhcp pool HQ_PHONES network 10.5.200.0 255.255.255.0 option 150 ip 172.21.51.204 default-router 10.5.200.1 On port g1/0/13 of the switch I have a 7960 that has successfully registered with CUCM, but the two 7945 keep loading up to a Upgrading screen and then re-cycling. The CUCM PUB is running the TFTP service and the CM Group (for registration time only) is configured to use PUB then SUB so that auto-registration will work correctly. Any ideas why this may be happening? Thanks, Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper Issue
Hi, I have restored and begun another lab already so I am unable to supply this debug. However, I realized I had the H225 Trunk was assigned to the HQ_DP, which used the HQ_REG, which requested G711. Eventually G729 was negotiated, because that's all that the BR2 would allow. I assigned the trunk to the correct DP, and the correct bandwidth was then requested. Jeff From: Angel Perez [mailto:gorr...@hotmail.com] Sent: Monday, February 22, 2010 2:17 AM To: afat...@verizon.net; Jeff Price (jeffpric) Cc: osl osl Subject: RE: [OSL | CCIE_Voice] Gatekeeper Issue Hello: If the gk trunk has inbound fast start checked, the gk will ask for 128k of bandwith then with a BRQ message it will reduce the bw to 16k, a debug h225 asn1 would be clarifaing. Jeff: Can you provide the following debug: deb h225 asn1 thx Date: Fri, 19 Feb 2010 18:48:47 -0800 From: afat...@verizon.net To: jeffp...@cisco.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue whats the direction of call? Previous debug has bandwidth 160, which is 16k for g729 and this one is showing bandwidth 1280 which is 128k for g711. So this call leg debug is showing is g711 is being requested somwhere. -- Mustafa Jeff Price (jeffpric) wrote: Here is the output of command debug gatekeeper call 10: *Feb 20 03:38:14.913: ////GK/gk_call_new: src_endptp=0x4ABCA7A8, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160, crv=61, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78 *Feb 20 03:38:14.913: ////GK/gk_call_find_endpts: NOT_FOUND *Feb 20 03:38:14.913: ////GK/gk_call_new: checking for default (CLI) carrier for sep endpt 0x4ABCA7A8 *Feb 20 03:38:14.937: ////GK/gk_call_find_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:14.937: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is DEP R1# *Feb 20 03:38:15.005: ////GK/gk_call_find_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:15.005: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is DEP R1# *Feb 20 03:38:24.397: ////GK/gk_call_find_crv: endptp=0x4ABCA7A8, crv=61: *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is SEP *Feb 20 03:38:24.397: ////GK/gk_call_clear_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is DEP *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x801, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A35B91C R1# *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x7C01, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:24.425: ////GK/gk_call_find_dstendpt: NOT_FOUND *Feb 20 03:38:24.425: ////GK/gk_call_new: src_endptp=0x47621FB0, dst_endptp=0x4ABFA848, src_pxp=0x0, dst_pxp=0x0, bw=1280, crv=32830, whichcrv=0x2, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78 *Feb 20 03:38:24.425: ////GK/gk_call_find_endpts: NOT_FOUND *Feb 20 03:38:24.429: ////GK/gk_call_new: checking for default (CLI) carrier for dep endpt 0x4ABFA848 R1# *Feb 20 03:38:29.449: ////GK/gk_call_clear_crv: endptp=0x4ABFA848, crv=32830: *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is DEP *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x800, c_resbw1 0, resbw2 1280, c_reszp1 0x4761C1A4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A57164C *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x6C00, c_resbw1 0, resbw2 0, c_reszp1 0x4761C1A4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.453: ////GK/gk_call_clear_crv: endptp=0x4ABCA7A8, crv=61: R1# *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is SEP *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A35B91C *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 The gk_call_find_endpts: NOT_FOUND line is leading me to believe something that I was noticed on the debug gatekeeper main 10 command: //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001) Matched zone prefix 1 and remainder 7752011001
Re: [OSL | CCIE_Voice] Gatekeeper Issue
I'm looking through this as I'm sending. I also wanted to note that I set the Service Parameter BRQ Enable to True, because I noticed that the DRQs were coming shortly after the BRQ were sent by the CME. *Feb 20 03:18:10.457: ////GK/gk_process: got a TIMER event *Feb 20 03:18:10.457: ////GK/gk_handle_timers *Feb 20 03:18:10.457: ////GK/gk_handle_timers: managed timer expired 0x47620C08 *Feb 20 03:18:10.733: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup R1# *Feb 20 03:18:13.393: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:13.393: ////GK/gk_rassrv_arq: arqp=0x4A5F7A4C,crv=0x37, answerCall=0 *Feb 20 03:18:13.393: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/gk_dns_query: No Name servers *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001) Matched tech-prefix 1# *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001) Matched zone prefix 1 and remainder 7752011001 *Feb 20 03:18:13.393: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4A696CA4 *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: matched zone is ZONE_01, an R1#d z_invianamelen=0 *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4A696CA4 *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: matched zone is ZONE_01, and z_outvianamelen=0 *Feb 20 03:18:13.397: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 *Feb 20 03:18:13.421: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:13.421: ////GK/gk_rassrv_arq: arqp=0x4AD8C8B8,crv=0x8037, answerCall=1 *Feb 20 03:18:13.421: //95E2E82B812F/95E2E82B8131/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC *Feb 20 03:18:13.465: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:13.469: ////GK/gk_rassrv_brq: state = 0xF *Feb 20 03:18:13.469: ////GK/gk_rassrv_brq: brqp=0x4A6F94A8, crv=0x8037, bandWidth=160 R1# R1# *Feb 20 03:18:22.781: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:22.781: ////GK/gk_rassrv_brq: state = 0xF *Feb 20 03:18:22.781: ////GK/gk_rassrv_brq: brqp=0x4A6F94A8, crv=0x37, bandWidth=0 *Feb 20 03:18:22.785: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:22.805: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:22.805: ////GK/gk_rassrv_arq: arqp=0x4AB97164,crv=0x8038, answerCall=1 *Feb 20 03:18:22.805: //95E2E82B812F/95E2E82B8131/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC R1# *Feb 20 03:18:25.457: ////GK/gk_process: got a TIMER event *Feb 20 03:18:25.457: ////GK/gk_handle_timers *Feb 20 03:18:25.457: ////GK/gk_handle_timers: managed timer expired 0x47620C08 R1# *Feb 20 03:18:27.825: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:27.829: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup R1# *Feb 20 03:18:33.965: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:34.381: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup R1# *Feb 20 03:18:40.457: ////GK/gk_process: got a TIMER event *Feb 20 03:18:40.457: ////GK/gk_handle_timers *Feb 20 03:18:40.457: ////GK/gk_handle_timers: managed timer expired 0x47620C08 Thanks, Jeff -Original Message- From: Mustafa [mailto:afat...@verizon.net] Sent: Friday, February 19, 2010 6:09 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue Jeff, Do a debug gatekeeper main 10 on the gatekeeper and check/send the output when you make a call. It will tell you whats happening and if there is a routing failure at the gatekeeper. Also, take a look at this documentation http://www.cisco.com/en/US/docs/ios/voice/cubegk/configuration/guide/ve_ book/ve_book.html , it details how to configure gatekeeper with CUBE and its a good read. -- Mustafa Jeff Price (jeffpric) wrote: Hi everyone, I'm having an issue with my gatekeeper. When calling from CUCM to GK to CME, the calls go through. I know this because I have shutdown the T1 controller
Re: [OSL | CCIE_Voice] Gatekeeper Issue
Mustafa, Do I need a XCODER on CME? Region is configured to use G729 and so is the dial-peer on CME. The call goes through successfully when calling from CUCM to CME site, so that leads me to believe the XCODER is necessary? The call never goes through. No ringing, no busy, nothing. As far as I can tell, the call goes through the GK logic of trying both TRUNKS (SUB then PUB) and then falls back to the PSTN dial-peer I've configured and the call is completed successfully through the PSTN. Jeff -Original Message- From: Mustafa [mailto:afat...@verizon.net] Sent: Friday, February 19, 2010 6:28 PM To: Jeff Price (jeffpric); CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue GK seems to be routing the call. Do you get a busy tone when the call is unsuccessful? Have you configured an xcoder anyway on CME? -- Mustafa Jeff Price (jeffpric) wrote: I'm looking through this as I'm sending. I also wanted to note that I set the Service Parameter BRQ Enable to True, because I noticed that the DRQs were coming shortly after the BRQ were sent by the CME. *Feb 20 03:18:10.457: ////GK/gk_process: got a TIMER event *Feb 20 03:18:10.457: ////GK/gk_handle_timers *Feb 20 03:18:10.457: ////GK/gk_handle_timers: managed timer expired 0x47620C08 *Feb 20 03:18:10.733: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup R1# *Feb 20 03:18:13.393: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:13.393: ////GK/gk_rassrv_arq: arqp=0x4A5F7A4C,crv=0x37, answerCall=0 *Feb 20 03:18:13.393: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/gk_dns_query: No Name servers *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001) Matched tech-prefix 1# *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001) Matched zone prefix 1 and remainder 7752011001 *Feb 20 03:18:13.393: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4A696CA4 *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: matched zone is ZONE_01, an R1#d z_invianamelen=0 *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4A696CA4 *Feb 20 03:18:13.393: //95E2E82B812F/95E2E82B8131/GK/rassrv_arq_select_viazone: matched zone is ZONE_01, and z_outvianamelen=0 *Feb 20 03:18:13.397: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 *Feb 20 03:18:13.421: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:13.421: ////GK/gk_rassrv_arq: arqp=0x4AD8C8B8,crv=0x8037, answerCall=1 *Feb 20 03:18:13.421: //95E2E82B812F/95E2E82B8131/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC *Feb 20 03:18:13.465: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:13.469: ////GK/gk_rassrv_brq: state = 0xF *Feb 20 03:18:13.469: ////GK/gk_rassrv_brq: brqp=0x4A6F94A8, crv=0x8037, bandWidth=160 R1# R1# *Feb 20 03:18:22.781: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:22.781: ////GK/gk_rassrv_brq: state = 0xF *Feb 20 03:18:22.781: ////GK/gk_rassrv_brq: brqp=0x4A6F94A8, crv=0x37, bandWidth=0 *Feb 20 03:18:22.785: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:22.805: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:22.805: ////GK/gk_rassrv_arq: arqp=0x4AB97164,crv=0x8038, answerCall=1 *Feb 20 03:18:22.805: //95E2E82B812F/95E2E82B8131/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC R1# *Feb 20 03:18:25.457: ////GK/gk_process: got a TIMER event *Feb 20 03:18:25.457: ////GK/gk_handle_timers *Feb 20 03:18:25.457: ////GK/gk_handle_timers: managed timer expired 0x47620C08 R1# *Feb 20 03:18:27.825: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:27.829: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup R1# *Feb 20 03:18:33.965: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:34.381: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup R1# *Feb 20 03:18:40.457: ////GK/gk_process: got a TIMER event *Feb 20 03:18:40.457: //
Re: [OSL | CCIE_Voice] Gatekeeper Issue
Here is the output of command debug gatekeeper call 10: *Feb 20 03:38:14.913: ////GK/gk_call_new: src_endptp=0x4ABCA7A8, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160, crv=61, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78 *Feb 20 03:38:14.913: ////GK/gk_call_find_endpts: NOT_FOUND *Feb 20 03:38:14.913: ////GK/gk_call_new: checking for default (CLI) carrier for sep endpt 0x4ABCA7A8 *Feb 20 03:38:14.937: ////GK/gk_call_find_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:14.937: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is DEP R1# *Feb 20 03:38:15.005: ////GK/gk_call_find_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:15.005: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is DEP R1# *Feb 20 03:38:24.397: ////GK/gk_call_find_crv: endptp=0x4ABCA7A8, crv=61: *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is SEP *Feb 20 03:38:24.397: ////GK/gk_call_clear_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is DEP *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x801, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A35B91C R1# *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x7C01, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:24.425: ////GK/gk_call_find_dstendpt: NOT_FOUND *Feb 20 03:38:24.425: ////GK/gk_call_new: src_endptp=0x47621FB0, dst_endptp=0x4ABFA848, src_pxp=0x0, dst_pxp=0x0, bw=1280, crv=32830, whichcrv=0x2, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78 *Feb 20 03:38:24.425: ////GK/gk_call_find_endpts: NOT_FOUND *Feb 20 03:38:24.429: ////GK/gk_call_new: checking for default (CLI) carrier for dep endpt 0x4ABFA848 R1# *Feb 20 03:38:29.449: ////GK/gk_call_clear_crv: endptp=0x4ABFA848, crv=32830: *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is DEP *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x800, c_resbw1 0, resbw2 1280, c_reszp1 0x4761C1A4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A57164C *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x6C00, c_resbw1 0, resbw2 0, c_reszp1 0x4761C1A4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.453: ////GK/gk_call_clear_crv: endptp=0x4ABCA7A8, crv=61: R1# *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is SEP *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A35B91C *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 The gk_call_find_endpts: NOT_FOUND line is leading me to believe something that I was noticed on the debug gatekeeper main 10 command: //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001) Matched zone prefix 1 and remainder 7752011001 The pattern I've configured on CUCM is expecting to receive in the form 17752011001. This output almost makes it seem like only 7752011001 is being sent, is this correct? I'm going to try to add another pattern without the 1 so that I can test this. Jeff -Original Message- From: Mustafa [mailto:afat...@verizon.net] Sent: Friday, February 19, 2010 6:28 PM To: Jeff Price (jeffpric); CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue GK seems to be routing the call. Do you get a busy tone when the call is unsuccessful? Have you configured an xcoder anyway on CME? -- Mustafa Jeff Price (jeffpric) wrote: I'm looking through this as I'm sending. I also wanted to note that I set the Service Parameter BRQ Enable to True, because I noticed that the DRQs were coming shortly after the BRQ were sent by the CME. *Feb 20 03:18:10.457: ////GK/gk_process: got a TIMER event *Feb 20 03:18:10.457: ////GK/gk_handle_timers *Feb 20 03:18:10.457: ////GK/gk_handle_timers: managed timer expired 0x47620C08 *Feb 20 03:18:10.733: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup R1# *Feb 20 03:18:13.393: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Feb 20 03:18:13.393: ////GK/gk_rassrv_arq: arqp=0x4A5F7A4C,crv=0x37, answerCall=0 *Feb 20 03:18:13.393
Re: [OSL | CCIE_Voice] Gatekeeper Issue
Mustafa, So I figured out why. CUCM was unable to find the number and that's why it couldn't route. The reason why CUCM could find the number is...a stupid mistake. I simply forgot to set the DDI to Predot. Now this is working. It always the little things... Thanks Mustafa for your help. Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price (jeffpric) Sent: Friday, February 19, 2010 6:38 PM To: CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue Here is the output of command debug gatekeeper call 10: *Feb 20 03:38:14.913: ////GK/gk_call_new: src_endptp=0x4ABCA7A8, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160, crv=61, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78 *Feb 20 03:38:14.913: ////GK/gk_call_find_endpts: NOT_FOUND *Feb 20 03:38:14.913: ////GK/gk_call_new: checking for default (CLI) carrier for sep endpt 0x4ABCA7A8 *Feb 20 03:38:14.937: ////GK/gk_call_find_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:14.937: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is DEP R1# *Feb 20 03:38:15.005: ////GK/gk_call_find_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:15.005: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is DEP R1# *Feb 20 03:38:24.397: ////GK/gk_call_find_crv: endptp=0x4ABCA7A8, crv=61: *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is SEP *Feb 20 03:38:24.397: ////GK/gk_call_clear_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is DEP *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x801, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A35B91C R1# *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x7C01, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:24.425: ////GK/gk_call_find_dstendpt: NOT_FOUND *Feb 20 03:38:24.425: ////GK/gk_call_new: src_endptp=0x47621FB0, dst_endptp=0x4ABFA848, src_pxp=0x0, dst_pxp=0x0, bw=1280, crv=32830, whichcrv=0x2, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78 *Feb 20 03:38:24.425: ////GK/gk_call_find_endpts: NOT_FOUND *Feb 20 03:38:24.429: ////GK/gk_call_new: checking for default (CLI) carrier for dep endpt 0x4ABFA848 R1# *Feb 20 03:38:29.449: ////GK/gk_call_clear_crv: endptp=0x4ABFA848, crv=32830: *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is DEP *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x800, c_resbw1 0, resbw2 1280, c_reszp1 0x4761C1A4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A57164C *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x6C00, c_resbw1 0, resbw2 0, c_reszp1 0x4761C1A4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.453: ////GK/gk_call_clear_crv: endptp=0x4ABCA7A8, crv=61: R1# *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is SEP *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A35B91C *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 The gk_call_find_endpts: NOT_FOUND line is leading me to believe something that I was noticed on the debug gatekeeper main 10 command: //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001) Matched zone prefix 1 and remainder 7752011001 The pattern I've configured on CUCM is expecting to receive in the form 17752011001. This output almost makes it seem like only 7752011001 is being sent, is this correct? I'm going to try to add another pattern without the 1 so that I can test this. Jeff -Original Message- From: Mustafa [mailto:afat...@verizon.net] Sent: Friday, February 19, 2010 6:28 PM To: Jeff Price (jeffpric); CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue GK seems to be routing the call. Do you get a busy tone when the call is unsuccessful? Have you configured an xcoder anyway on CME? -- Mustafa Jeff Price (jeffpric) wrote: I'm looking through this as I'm sending. I also wanted to note that I set the Service Parameter BRQ Enable to True, because I noticed that the DRQs were coming shortly after the BRQ were sent by the CME. *Feb 20 03:18:10.457
Re: [OSL | CCIE_Voice] Gatekeeper Issue
Hi Mustafa, The direction is from CME to CUCM. Both are configured for G.729 and I've confirmed a call goes through and uses G729 both ways. That's definitely odd for the bandwidth to say that. I don't really have an answer, but the call is using G729 as configured. Thanks again. Jeff -Original Message- From: Mustafa [mailto:afat...@verizon.net] Sent: Friday, February 19, 2010 6:49 PM To: Jeff Price (jeffpric) Cc: CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue whats the direction of call? Previous debug has bandwidth 160, which is 16k for g729 and this one is showing bandwidth 1280 which is 128k for g711. So this call leg debug is showing is g711 is being requested somwhere. -- Mustafa Jeff Price (jeffpric) wrote: Here is the output of command debug gatekeeper call 10: *Feb 20 03:38:14.913: ////GK/gk_call_new: src_endptp=0x4ABCA7A8, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160, crv=61, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78 *Feb 20 03:38:14.913: ////GK/gk_call_find_endpts: NOT_FOUND *Feb 20 03:38:14.913: ////GK/gk_call_new: checking for default (CLI) carrier for sep endpt 0x4ABCA7A8 *Feb 20 03:38:14.937: ////GK/gk_call_find_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:14.937: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is DEP R1# *Feb 20 03:38:15.005: ////GK/gk_call_find_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:15.005: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is DEP R1# *Feb 20 03:38:24.397: ////GK/gk_call_find_crv: endptp=0x4ABCA7A8, crv=61: *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_find_crv: crv is SEP *Feb 20 03:38:24.397: ////GK/gk_call_clear_crv: endptp=0x4A6F20B0, crv=32829: *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is DEP *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x801, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A35B91C R1# *Feb 20 03:38:24.397: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x7C01, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:24.425: ////GK/gk_call_find_dstendpt: NOT_FOUND *Feb 20 03:38:24.425: ////GK/gk_call_new: src_endptp=0x47621FB0, dst_endptp=0x4ABFA848, src_pxp=0x0, dst_pxp=0x0, bw=1280, crv=32830, whichcrv=0x2, circuit=0x0, capacity=0x0, ret_callpp=0x492D8D78 *Feb 20 03:38:24.425: ////GK/gk_call_find_endpts: NOT_FOUND *Feb 20 03:38:24.429: ////GK/gk_call_new: checking for default (CLI) carrier for dep endpt 0x4ABFA848 R1# *Feb 20 03:38:29.449: ////GK/gk_call_clear_crv: endptp=0x4ABFA848, crv=32830: *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is DEP *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x800, c_resbw1 0, resbw2 1280, c_reszp1 0x4761C1A4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A57164C *Feb 20 03:38:29.449: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x6C00, c_resbw1 0, resbw2 0, c_reszp1 0x4761C1A4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.453: ////GK/gk_call_clear_crv: endptp=0x4ABCA7A8, crv=61: R1# *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: crv is SEP *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_clear_crv: c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: callp=4A35B91C *Feb 20 03:38:29.453: //620E1FC68141/620EBBEE8143/GK/gk_call_delete: c_callstate 0x7C00, c_resbw1 0, resbw2 0, c_reszp1 0x4A696CA4, c_reszp2 0x4A696CA4 The gk_call_find_endpts: NOT_FOUND line is leading me to believe something that I was noticed on the debug gatekeeper main 10 command: //95E2E82B812F/95E2E82B8131/GK/rassrv_get_addrinfo: (1#17752011001) Matched zone prefix 1 and remainder 7752011001 The pattern I've configured on CUCM is expecting to receive in the form 17752011001. This output almost makes it seem like only 7752011001 is being sent, is this correct? I'm going to try to add another pattern without the 1 so that I can test this. Jeff -Original Message- From: Mustafa [mailto:afat...@verizon.net] Sent: Friday, February 19, 2010 6:28 PM To: Jeff Price (jeffpric); CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Gatekeeper Issue GK seems to be routing the call. Do you get a busy tone when the call is unsuccessful? Have you configured an xcoder
Re: [OSL | CCIE_Voice] Gatekeeper Issue
Hi Kavi, Thanks. I already had this configured. I appreciate your help. Now the GK functionality is working like a charm. I just forgot the Predot DDI in the CUCM Translation Pattern. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kavi ten Sent: Friday, February 19, 2010 8:12 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper Issue Hi Jeff, Please chk the Tech prefix on the CUCM , If its set to 1# , then u need to send 1 # along with the no from CME remove it at the GW with incoming allowed nos or use TP. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Dialing problem
Hey Bas, I got everything working for the most part. It seems like some things just take a while to start working for this lab. I did create some SIP dial rules, because I know its best practice to offload that processing. I have since restored all of my lab back to a clean config to run through the lab from the beginning, so if I run into any problems I will let you all know. Thanks for getting back to me. Jeff From: Bas Janssen [mailto:basmj...@msn.com] Sent: Friday, February 12, 2010 4:15 AM To: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Dialing problem Hi Jeff, Any luck so far? One last remark, if you use SIP phones, don't forget to make dial rules, even if your SIP phones supports KPML Otherwise calling back from the missed call list might not work because SIP phones use overlap sending from the missed call list without SIP dial rules. regards, Bas Subject: RE: [OSL | CCIE_Voice] Dialing problem Date: Tue, 9 Feb 2010 12:32:46 -0600 From: jeffp...@cisco.com To: basmj...@msn.com; ccie_voice@onlinestudylist.com Hey Bas, Thanks for the input. I will look into all of this later today. Jeff From: Bas Janssen [mailto:basmj...@msn.com] Sent: Tuesday, February 09, 2010 6:03 AM To: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Dialing problem Hi Jeff, I run into the same issue today, it was introduced on configuring + dialing. So, call flow is for example 93942123---trans. pattern--- +12123942123route pattern \+!-Route List ---GW---called party transform-3942123 type subscriber Objective is to use only one route pattern to PSTN. Had play second dial tone ON on TP and RP. You have to check box urgent priority on the \+! dial pattern, otherwise you have the second dial tone while waiting for inter digit timeout. You can remove the check for playing second dial tone, but than you stiil have an inter digit timeout. (without urgent priority) Watch out for international numbers, these will fail with urgent priority on the \+! so if you have a trans pattern with \+!, remove the urgent priority here to get international calls to work. regards, Bas Date: Mon, 8 Feb 2010 20:10:08 -0600 From: jeffp...@cisco.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Dialing problem Jonathan, CUCM and SCCP. Jeff From: Jonathan Charles [mailto:jonv...@gmail.com] Sent: Monday, February 08, 2010 4:32 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Dialing problem CME or CUCM? SIP or SCCP? If SIP, did you create a SIP Dial Plan? Jonathan On Mon, Feb 8, 2010 at 5:28 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: Hi everyone, I am having an issue with the dialing for my phones. All of my phones are able to call across the PSTN, however it only works when I dial the digits and then press the Dial softkey. If I pick up the headset and dial that way I get a steady tone. Anyone familiar with this issue? Thanks. Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Free, trusted and rich email service. Get it now. https://signup.live.com/signup.aspx?id=60969 Hotmail: Trusted email with powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 image001.jpgimage002.jpgimage003.jpg___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] PSTN/ISDN issue
Hi everyone, I am having an issue with my PSTN router. The PSTN router keeps sending a Restart message on channel 1 every 30 seconds or so to my HQ's router (R1). Does anyone know what could cause this behavior? I have tried reconfiguring my isdn on both sides, but I am still seeing the same issue even after reconfiguring this. Here is the error as seen from R1: Feb 13 06:29:00.402: ISDN Se0/0/0:23 Q931: RX - RESTART pd = 8 callref = 0x Channel ID i = 0xA98381 Exclusive, Channel 1 Restart Indicator i = 0x80 Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com mailto:jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html http://www.cisco.com/web/about/doing_business/legal/cri/index.html image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Dialing problem
Hey Bas, Thanks for the input. I will look into all of this later today. Jeff From: Bas Janssen [mailto:basmj...@msn.com] Sent: Tuesday, February 09, 2010 6:03 AM To: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Dialing problem Hi Jeff, I run into the same issue today, it was introduced on configuring + dialing. So, call flow is for example 93942123---trans. pattern--- +12123942123route pattern \+!-Route List ---GW---called party transform-3942123 type subscriber Objective is to use only one route pattern to PSTN. Had play second dial tone ON on TP and RP. You have to check box urgent priority on the \+! dial pattern, otherwise you have the second dial tone while waiting for inter digit timeout. You can remove the check for playing second dial tone, but than you stiil have an inter digit timeout. (without urgent priority) Watch out for international numbers, these will fail with urgent priority on the \+! so if you have a trans pattern with \+!, remove the urgent priority here to get international calls to work. regards, Bas Date: Mon, 8 Feb 2010 20:10:08 -0600 From: jeffp...@cisco.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Dialing problem Jonathan, CUCM and SCCP. Jeff From: Jonathan Charles [mailto:jonv...@gmail.com] Sent: Monday, February 08, 2010 4:32 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Dialing problem CME or CUCM? SIP or SCCP? If SIP, did you create a SIP Dial Plan? Jonathan On Mon, Feb 8, 2010 at 5:28 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: Hi everyone, I am having an issue with the dialing for my phones. All of my phones are able to call across the PSTN, however it only works when I dial the digits and then press the Dial softkey. If I pick up the headset and dial that way I get a steady tone. Anyone familiar with this issue? Thanks. Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Free, trusted and rich email service. Get it now. https://signup.live.com/signup.aspx?id=60969 image001.jpgimage002.jpgimage003.jpg___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MGCP Gateway Problem
Hi everyone, So after trying all of your suggestions, I guess the old Windows philosophy worked. After I restarted the PUB/SUB I am not able to make the calls. Thanks for all of your help. Jeff From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas Sent: Monday, February 08, 2010 2:09 PM To: Jeff Price (jeffpric) Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem I'm sorry Jeff. I -always- do that but if I go to the CUCM interface I do it right. The Route List can be reset -- try that and see if you can suddenly start dialing. If not, try resetting the gateway too.We've had cases in our production environment where a location can't dial anywhere, but if you reset the route list its fixes it. -jeff On Mon, Feb 8, 2010 at 4:58 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: Jeff, I am not sure what you mean by reset the route group. I don't see a reset option under Route Group. However, I have reset the GW, and Route List numerous times. Is this what you mean? If not, could you provide me some more information? Thanks for your help. Jeff From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas Sent: Monday, February 08, 2010 12:46 PM To: Jeff Price (jeffpric) Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Jeff - did you try to reset the route group? If not, navigate to your route group, reset it and see if your call attempt shows up in debug isdn q931. If you're not using the console don't forget to turn on console logging. About 99% of the time in our production environment when I get a fast busy signal immediately after dialing (and no presence of the call attempting to exit the gateway) its because the route group needs to be reset. -jeff On Mon, Feb 8, 2010 at 2:50 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: Hi all, I'm fairly new to the RTMT tool. I'm looking around trying to find the correct area to be and what to select to monitor. Can someone point me in the right direction for this problem? Thanks. Jeff -Original Message- From: Roger Källberg [mailto:roger.kallb...@cygate.se] Sent: Monday, February 08, 2010 4:08 AM To: Scott Totaro (stotaro); Jeff Price (jeffpric); afatsum Cc: ccie_voice@onlinestudylist.com Subject: SV: [OSL | CCIE_Voice] MGCP Gateway Problem Hi Scott, Even if you use MGCP, ie backhauled D-channel, you can see the output of deb isdn q931 on the gateway. It will be locally echoed. Brgds, Roger Källberg Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se Från: Scott Totaro (stotaro) [stot...@cisco.com] Skickat: den 8 februari 2010 01:15 Till: Jeff Price (jeffpric); afatsum Kopia: ccie_voice@onlinestudylist.com Ämne: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Because the D-channel is backhauled to CUCM for MGCP gateways, you will not see Q931 output on the router. Instead, you'll need to look at the detailed trace files on the subscriber that the MGCP gateway is registered too. In my experience, you won't be able to use any IOS commands that depend on the router terminating the D-channel (e.g. show isdn history.) Hope this helps, Scott -Original Message- From: Jeff Price (jeffpric) Sent: Sunday, February 07, 2010 3:41 PM To: afatsum Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem It appears that my transformation is working. When dialing 91408425, the display on the phone says To 408425. And the DNA analysis shows what the CUCM is going through process-wise. Yet I am still not receiving any ISDN Q931 debug output on R1 and the phones still receive a fast busy. As I had said in a previous email, the PSTN router that the phones are calling to is already pre-configured and I don't have access to it. Even if it was the PSTN router causing the problem, wouldn't I still see the Q931 output on R1? Thanks for the help. Jeff -Original Message- From: Jeff Price (jeffpric) Sent: Sunday, February 07, 2010 12:35 PM To: 'afatsum' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem I have deactivated all of the services on the SUB and let everything register with the PUB. Jeff -Original Message- From: afatsum [mailto:afat...@verizon.net] Sent: Sunday, February 07, 2010 12:05 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Hi Jeff, Can you shutdown the sub and let everything register to pub and then do the testing? This way atleast we can eliminate the sub for troubleshooting purposes. -- Mustafa Jeff Price (jeffpric) wrote: Hi again, I was able to access the DNA on SUB, but not the PUB even though both servers are running the service. Here is the output of DNA. Everything seems to be okay. *Cisco Unified Communications
Re: [OSL | CCIE_Voice] MGCP Gateway Problem
Now I am able to. I ran the utils dbreplication status command as well, which didn't report any issues. However, maybe this command had an issue as well. I'm just glad it's working now. From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas Sent: Monday, February 08, 2010 3:11 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Jeff - you ARE able to make the calls? Sounds like you may have had some kind of replication issue going on between the pub/sub. -jeff On Mon, Feb 8, 2010 at 5:47 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: Hi everyone, So after trying all of your suggestions, I guess the old Windows philosophy worked. After I restarted the PUB/SUB I am not able to make the calls. Thanks for all of your help. Jeff From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas Sent: Monday, February 08, 2010 2:09 PM To: Jeff Price (jeffpric) Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem I'm sorry Jeff. I -always- do that but if I go to the CUCM interface I do it right. The Route List can be reset -- try that and see if you can suddenly start dialing. If not, try resetting the gateway too.We've had cases in our production environment where a location can't dial anywhere, but if you reset the route list its fixes it. -jeff On Mon, Feb 8, 2010 at 4:58 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: Jeff, I am not sure what you mean by reset the route group. I don't see a reset option under Route Group. However, I have reset the GW, and Route List numerous times. Is this what you mean? If not, could you provide me some more information? Thanks for your help. Jeff From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas Sent: Monday, February 08, 2010 12:46 PM To: Jeff Price (jeffpric) Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Jeff - did you try to reset the route group? If not, navigate to your route group, reset it and see if your call attempt shows up in debug isdn q931. If you're not using the console don't forget to turn on console logging. About 99% of the time in our production environment when I get a fast busy signal immediately after dialing (and no presence of the call attempting to exit the gateway) its because the route group needs to be reset. -jeff On Mon, Feb 8, 2010 at 2:50 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: Hi all, I'm fairly new to the RTMT tool. I'm looking around trying to find the correct area to be and what to select to monitor. Can someone point me in the right direction for this problem? Thanks. Jeff -Original Message- From: Roger Källberg [mailto:roger.kallb...@cygate.se] Sent: Monday, February 08, 2010 4:08 AM To: Scott Totaro (stotaro); Jeff Price (jeffpric); afatsum Cc: ccie_voice@onlinestudylist.com Subject: SV: [OSL | CCIE_Voice] MGCP Gateway Problem Hi Scott, Even if you use MGCP, ie backhauled D-channel, you can see the output of deb isdn q931 on the gateway. It will be locally echoed. Brgds, Roger Källberg Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se Från: Scott Totaro (stotaro) [stot...@cisco.com] Skickat: den 8 februari 2010 01:15 Till: Jeff Price (jeffpric); afatsum Kopia: ccie_voice@onlinestudylist.com Ämne: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Because the D-channel is backhauled to CUCM for MGCP gateways, you will not see Q931 output on the router. Instead, you'll need to look at the detailed trace files on the subscriber that the MGCP gateway is registered too. In my experience, you won't be able to use any IOS commands that depend on the router terminating the D-channel (e.g. show isdn history.) Hope this helps, Scott -Original Message- From: Jeff Price (jeffpric) Sent: Sunday, February 07, 2010 3:41 PM To: afatsum Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem It appears that my transformation is working. When dialing 91408425, the display on the phone says To 408425. And the DNA analysis shows what the CUCM is going through process-wise. Yet I am still not receiving any ISDN Q931 debug output on R1 and the phones still receive a fast busy. As I had said in a previous email, the PSTN router that the phones are calling to is already pre-configured and I don't have access to it. Even if it was the PSTN router causing the problem, wouldn't I still see the Q931 output on R1? Thanks for the help. Jeff -Original Message- From: Jeff Price (jeffpric) Sent: Sunday, February 07, 2010 12:35 PM To: 'afatsum' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem I have deactivated all of the services on the SUB and let everything register with the PUB. Jeff
[OSL | CCIE_Voice] Dialing problem
Hi everyone, I am having an issue with the dialing for my phones. All of my phones are able to call across the PSTN, however it only works when I dial the digits and then press the Dial softkey. If I pick up the headset and dial that way I get a steady tone. Anyone familiar with this issue? Thanks. Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com mailto:jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html http://www.cisco.com/web/about/doing_business/legal/cri/index.html image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Dialing problem
Jonathan, CUCM and SCCP. Jeff From: Jonathan Charles [mailto:jonv...@gmail.com] Sent: Monday, February 08, 2010 4:32 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Dialing problem CME or CUCM? SIP or SCCP? If SIP, did you create a SIP Dial Plan? Jonathan On Mon, Feb 8, 2010 at 5:28 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: Hi everyone, I am having an issue with the dialing for my phones. All of my phones are able to call across the PSTN, however it only works when I dial the digits and then press the Dial softkey. If I pick up the headset and dial that way I get a steady tone. Anyone familiar with this issue? Thanks. Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MGCP Gateway Problem
Good morning everyone, Thanks Jeff for the response. I'm going to be looking into all of this. However, my configuration is as follow: HQ Phones - CSS_HQ_DEVICES (DEVICE, no LINE CSS as of yet) Translation Patterns - PT_USA, CSS_GLOBAL Route Pattern - PT_GLOBAL CSSs CSS_HQ_DEVICES: PT_HQ_DEVICES, PT_USA CSS_GLOBAL: PT_GLOBAL This brings me to the next question, why would DNA not allow me to use it? Every time I open up the DNA, I get a DNA Service is still initializing. Please refresh after some time or something like that. Which if you look at the user guide from Cisco, that's normal. But it stays like that for over an hour. I can't actually use the DNA. I've dis-enabled and then re-enabled the Database synchronization and then restarted the service through the Control Center - Feature Services. I've also gone to Service Activation and then Deactivated and then Activated, and still the same message for long after it should have stopped. Has anyone seen this issue before? Any suggestions as to how to get this to work? Jeff From: jgar...@gmail.com [mailto:jgar...@gmail.com] On Behalf Of Jeff Garvas Sent: Saturday, February 06, 2010 2:07 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Jeff, If debug q931 isn't producing output and everything appears to be registered right I'd assume that you have an issue with your phone/line CSS, route patterns, route list, route group, etc. Have you tried resetting the Route Group? Have you tried the dialed number analyzer to see where CUCM thinks the calls should be going for a DN in the particular CSS in use, if anywhere? DNA will tell you if you have a failure in your design, but it won't tell you to go reset the Route Group / Gateway etc. -Jeff On Sat, Feb 6, 2010 at 3:44 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: Hey everyone, I wanted see if anyone had any ideas as to what may be wrong here. I have set up the MGCP gateway correctly as far as I can tell. It is registered with CUCM. Attached is the GW's config as well as show ccm-manager, show isdn status and the call routing elements. Everything configuration wise seems to be working. I have a registered status in CUCM and on the GW. Everything seems pretty standard with the routing elements. I have repeatedly done a no mgcp, mgcp to restart the mgcp process. I have physically unplugged the t1 cable to restart it. I have deleted and reconfigured the GW in CUCM as well as on the GW itself. I changed the Stop Routing on Unallocated Number Service Parameter to False (which I know probably isn't effecting anything related to this). I've restarted and/or activated/deactivated all of the various related feature services. I'm sure there's more I've done as I've been trying to figure this out for almost two days. Also, the debug isdn q931 isn't producing any output, which is odd to me. I keep getting a fast busy whenever I dial any numbers. I don't have access to the PSTN router's configuration, however I know it was working as I was able to make phone calls to it when I first got here, but now that I've done my own configuration it is not working anymore. Also, one of the engineers who set it up verified the configuration for me. Please help! It is very much appreciated. J Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.jpgimage002.jpgimage003.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MGCP Gateway Problem
I have deactivated all of the services on the SUB and let everything register with the PUB. Jeff -Original Message- From: afatsum [mailto:afat...@verizon.net] Sent: Sunday, February 07, 2010 12:05 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Hi Jeff, Can you shutdown the sub and let everything register to pub and then do the testing? This way atleast we can eliminate the sub for troubleshooting purposes. -- Mustafa Jeff Price (jeffpric) wrote: Hi again, I was able to access the DNA on SUB, but not the PUB even though both servers are running the service. Here is the output of DNA. Everything seems to be okay. *Cisco Unified Communications Manager Dialed Number Analyzer Results * * *Results Summary* o *Calling Party Information* + *Calling Party* = +14085252001 + *Partition* = PT_HQ_DEVICES + *Device CSS* = CSS_HQ_DEVICES + *Line CSS* = + *AAR Group Name* = + *AAR CSS* = o *Dialed Digits* = 91408425 o *Match Result* = RouteThisPattern o *Matched Pattern Information* + *Pattern* = \+! + *Partition* = PT_GLOBAL + *Time Schedule* = o *Called Party Number* = +1408425 o *Time Zone* = Pacific Standard/Daylight Time o *End Device* = RL_LOCAL o *Call Classification* = OffNet o *InterDigit Timeout* = NO o *Device Override* = Disabled o *Outside Dial Tone* = NO * *Call Flow* o *TranslationPattern* :*Pattern*= 9.1[2-9]XX[2-9]XX + *Positional Match List* = +1408425 + *Calling Party Number* = +14085252001 + *PreTransform Calling Party Number* = 2001 + *PreTransform Called Party Number* = 91408425 + *Calling Party Transformations* # *External Phone Number Mask* = YES # *Calling Party Mask* = # *Prefix* = # *CallingLineId Presentation* = Default # *CallingName Presentation* = Default # *Calling Party Number* = +14085252001 + *ConnectedParty Transformations* # *ConnectedLineId Presentation* = Default # *ConnectedName Presentation* = Default + *Called Party Transformations* # *Called Party Mask* = # *Discard Digits Instruction* = PreDot # *Prefix* = + # *Called Number* = +1408425 o *Route Pattern* :*Pattern*= \+! + *Positional Match List* = +1408425 + *DialPlan* = + *Route Filter* # *Filter Name* = # *Filter Clause* = + *Require Forced Authorization Code* = No + *Authorization Level* = 0 + *Require Client Matter Code* = No + *Call Classification* = + *PreTransform Calling Party Number* = +14085252001 + *PreTransform Called Party Number* = +1408425 + *Calling Party Transformations* # *External Phone Number Mask* = YES # *Calling Party Mask* = # *Prefix* = # *CallingLineId Presentation* = Default # *CallingName Presentation* = Default # *Calling Party Number* = +14085252001 + *ConnectedParty Transformations* # *ConnectedLineId Presentation* = Default # *ConnectedName Presentation* = Default + *Called Party Transformations* # *Called Party Mask* = # *Discard Digits Instruction* = None # *Prefix* = # *Called Number* = +1408425 o *Route List* :*Route List Name*= RL_LOCAL + *RouteGroup* :*RouteGroup Name*= Standard Local Route Group # *PreTransform Calling Party Number* = +14085252001 # *PreTransform Called Party Number* = +1408425 # *Calling Party Transformations* * *External Phone Number Mask* = Default * *Calling Party Mask* = * *Prefix* = * *Calling Party Number* = +14085252001 # *Called Party Transformations* * *Called Party Mask* = * *Discard Digits Instructions
Re: [OSL | CCIE_Voice] MGCP Gateway Problem
It appears that my transformation is working. When dialing 91408425, the display on the phone says To 408425. And the DNA analysis shows what the CUCM is going through process-wise. Yet I am still not receiving any ISDN Q931 debug output on R1 and the phones still receive a fast busy. As I had said in a previous email, the PSTN router that the phones are calling to is already pre-configured and I don't have access to it. Even if it was the PSTN router causing the problem, wouldn't I still see the Q931 output on R1? Thanks for the help. Jeff -Original Message- From: Jeff Price (jeffpric) Sent: Sunday, February 07, 2010 12:35 PM To: 'afatsum' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem I have deactivated all of the services on the SUB and let everything register with the PUB. Jeff -Original Message- From: afatsum [mailto:afat...@verizon.net] Sent: Sunday, February 07, 2010 12:05 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Hi Jeff, Can you shutdown the sub and let everything register to pub and then do the testing? This way atleast we can eliminate the sub for troubleshooting purposes. -- Mustafa Jeff Price (jeffpric) wrote: Hi again, I was able to access the DNA on SUB, but not the PUB even though both servers are running the service. Here is the output of DNA. Everything seems to be okay. *Cisco Unified Communications Manager Dialed Number Analyzer Results * * *Results Summary* o *Calling Party Information* + *Calling Party* = +14085252001 + *Partition* = PT_HQ_DEVICES + *Device CSS* = CSS_HQ_DEVICES + *Line CSS* = + *AAR Group Name* = + *AAR CSS* = o *Dialed Digits* = 91408425 o *Match Result* = RouteThisPattern o *Matched Pattern Information* + *Pattern* = \+! + *Partition* = PT_GLOBAL + *Time Schedule* = o *Called Party Number* = +1408425 o *Time Zone* = Pacific Standard/Daylight Time o *End Device* = RL_LOCAL o *Call Classification* = OffNet o *InterDigit Timeout* = NO o *Device Override* = Disabled o *Outside Dial Tone* = NO * *Call Flow* o *TranslationPattern* :*Pattern*= 9.1[2-9]XX[2-9]XX + *Positional Match List* = +1408425 + *Calling Party Number* = +14085252001 + *PreTransform Calling Party Number* = 2001 + *PreTransform Called Party Number* = 91408425 + *Calling Party Transformations* # *External Phone Number Mask* = YES # *Calling Party Mask* = # *Prefix* = # *CallingLineId Presentation* = Default # *CallingName Presentation* = Default # *Calling Party Number* = +14085252001 + *ConnectedParty Transformations* # *ConnectedLineId Presentation* = Default # *ConnectedName Presentation* = Default + *Called Party Transformations* # *Called Party Mask* = # *Discard Digits Instruction* = PreDot # *Prefix* = + # *Called Number* = +1408425 o *Route Pattern* :*Pattern*= \+! + *Positional Match List* = +1408425 + *DialPlan* = + *Route Filter* # *Filter Name* = # *Filter Clause* = + *Require Forced Authorization Code* = No + *Authorization Level* = 0 + *Require Client Matter Code* = No + *Call Classification* = + *PreTransform Calling Party Number* = +14085252001 + *PreTransform Called Party Number* = +1408425 + *Calling Party Transformations* # *External Phone Number Mask* = YES # *Calling Party Mask* = # *Prefix* = # *CallingLineId Presentation* = Default # *CallingName Presentation* = Default # *Calling Party Number* = +14085252001 + *ConnectedParty Transformations* # *ConnectedLineId Presentation* = Default # *ConnectedName Presentation* = Default + *Called Party Transformations* # *Called Party Mask* = # *Discard Digits Instruction* = None # *Prefix* = # *Called Number* = +1408425
Re: [OSL | CCIE_Voice] MGCP Gateway Problem
Mustafa - I have configured Locations, however all of the HQ devices (GW and phones) should all be in the HQ_LOC location. However, I will verify this tomorrow when I am back in the lab. Scott - Thanks for the advice. I plan on trying the RTMT tomorrow while back in the lab. Hopefully, this will help reveal the problem. Thanks everyone for the help and I will provide more information tomorrow as to my findings. Jeff -Original Message- From: Scott Totaro (stotaro) Sent: Sunday, February 07, 2010 8:41 PM To: afatsum Cc: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem I stand corrected...sorry for the misdirection. I guess it's the show commands that haven't worked whereas the debugs are still useful. I'd still suggest investigating the detailed trace files for the source of the failure as Mustafa points out that the lack of debug likely indicates that the call is never making it to the gateway. Scott -Original Message- From: afatsum [mailto:afat...@verizon.net] Sent: Sunday, February 07, 2010 7:46 PM To: Scott Totaro (stotaro) Cc: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem If you do a Debug isdn q931, you will see the layer 3 activity debug on the gateway. The layer 3 backhaul makes all the processing happen on the CUCM, thats why no cli commands, other than mgcp and ccm-manager are useful on the gateway. The debug isdn q931 output should show if the call is ever sent over the isdn interface. If you don't see the output, then its never making it to the gateway. Another issue that can cause fast busy is if the gateway and phones are in two different locations and CAC has been implemented. Do you have configured CAC? Hub_none is still a different location. -- Mustafa Scott Totaro (stotaro) wrote: Because the D-channel is backhauled to CUCM for MGCP gateways, you will not see Q931 output on the router. Instead, you'll need to look at the detailed trace files on the subscriber that the MGCP gateway is registered too. In my experience, you won't be able to use any IOS commands that depend on the router terminating the D-channel (e.g. show isdn history.) Hope this helps, Scott -Original Message- From: Jeff Price (jeffpric) Sent: Sunday, February 07, 2010 3:41 PM To: afatsum Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem It appears that my transformation is working. When dialing 91408425, the display on the phone says To 408425. And the DNA analysis shows what the CUCM is going through process-wise. Yet I am still not receiving any ISDN Q931 debug output on R1 and the phones still receive a fast busy. As I had said in a previous email, the PSTN router that the phones are calling to is already pre-configured and I don't have access to it. Even if it was the PSTN router causing the problem, wouldn't I still see the Q931 output on R1? Thanks for the help. Jeff -Original Message- From: Jeff Price (jeffpric) Sent: Sunday, February 07, 2010 12:35 PM To: 'afatsum' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem I have deactivated all of the services on the SUB and let everything register with the PUB. Jeff -Original Message- From: afatsum [mailto:afat...@verizon.net] Sent: Sunday, February 07, 2010 12:05 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Hi Jeff, Can you shutdown the sub and let everything register to pub and then do the testing? This way atleast we can eliminate the sub for troubleshooting purposes. -- Mustafa Jeff Price (jeffpric) wrote: Hi again, I was able to access the DNA on SUB, but not the PUB even though both servers are running the service. Here is the output of DNA. Everything seems to be okay. *Cisco Unified Communications Manager Dialed Number Analyzer Results * * *Results Summary* o *Calling Party Information* + *Calling Party* = +14085252001 + *Partition* = PT_HQ_DEVICES + *Device CSS* = CSS_HQ_DEVICES + *Line CSS* = + *AAR Group Name* = + *AAR CSS* = o *Dialed Digits* = 91408425 o *Match Result* = RouteThisPattern o *Matched Pattern Information* + *Pattern* = \+! + *Partition* = PT_GLOBAL + *Time Schedule* = o *Called Party Number* = +1408425 o *Time Zone* = Pacific Standard/Daylight Time o *End Device* = RL_LOCAL o *Call Classification* = OffNet o *InterDigit Timeout* = NO o *Device Override* = Disabled o *Outside Dial Tone* = NO * *Call Flow* o *TranslationPattern
[OSL | CCIE_Voice] MGCP Gateway Problem
Hey everyone, I wanted see if anyone had any ideas as to what may be wrong here. I have set up the MGCP gateway correctly as far as I can tell. It is registered with CUCM. Attached is the GW's config as well as show ccm-manager, show isdn status and the call routing elements. Everything configuration wise seems to be working. I have a registered status in CUCM and on the GW. Everything seems pretty standard with the routing elements. I have repeatedly done a no mgcp, mgcp to restart the mgcp process. I have physically unplugged the t1 cable to restart it. I have deleted and reconfigured the GW in CUCM as well as on the GW itself. I changed the Stop Routing on Unallocated Number Service Parameter to False (which I know probably isn't effecting anything related to this). I've restarted and/or activated/deactivated all of the various related feature services. I'm sure there's more I've done as I've been trying to figure this out for almost two days. Also, the debug isdn q931 isn't producing any output, which is odd to me. I keep getting a fast busy whenever I dial any numbers. I don't have access to the PSTN router's configuration, however I know it was working as I was able to make phone calls to it when I first got here, but now that I've done my own configuration it is not working anymore. Also, one of the engineers who set it up verified the configuration for me. Please help! It is very much appreciated. J Jeff Price Network Consulting Engineer - Unified Communications Practice jeffp...@cisco.com mailto:jeffp...@cisco.com Phone: 408-525-8293 Mobile: 408-204-4510 Cisco Systems, Inc. 170 West Tasman Drive, San Jose, CA 95134-1706 USA Cisco home page http://www.cisco.com/ Think before you print. This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html http://www.cisco.com/web/about/doing_business/legal/cri/index.html image001.jpgimage002.jpgimage003.gifversion 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname R1 ! boot-start-marker boot system flash:c2800nm-adventerprisek9-mz.124-24.T.bin boot-end-marker ! logging message-counter syslog enable password 7 110A1A0C12 ! no aaa new-model network-clock-participate wic 0 ! dot11 syslog ip source-route ! ! ip cef ! ! no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-ni ! ! voice-card 0 ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller T1 0/0/0 pri-group timeslots 1-6,24 service mgcp ! controller T1 0/0/1 ! ! ! ! ! interface GigabitEthernet0/0 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$ no ip address duplex auto speed auto ! interface GigabitEthernet0/0.102 encapsulation dot1Q 102 native ip address 172.21.51.196 255.255.255.224 ! interface GigabitEthernet0/0.150 encapsulation dot1Q 150 ip address 10.5.100.1 255.255.255.0 ! interface GigabitEthernet0/0.250 encapsulation dot1Q 250 ip address 10.5.200.1 255.255.255.0 ip helper-address 172.21.51.204 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! interface Serial0/1/0 no ip address encapsulation frame-relay IETF tx-ring-limit 128 tx-queue-limit 128 serial restart-delay 0 frame-relay lmi-type ansi ! interface Serial0/1/0.1 point-to-point ip address 162.5.101.1 255.255.255.0 ip ospf mtu-ignore frame-relay interface-dlci 201 ! interface Serial0/1/0.2 point-to-point ip address 162.5.102.1 255.255.255.0 ip ospf mtu-ignore frame-relay interface-dlci 202 ! router ospf 1 log-adjacency-changes network 10.5.100.0 0.0.0.255 area 0 network 10.5.200.0 0.0.0.255 area 0 network 162.5.101.0 0.0.0.255 area 0 network 162.5.102.0 0.0.0.255 area 0 network 172.5.100.0 0.0.0.255 area 0 network 172.21.51.193 0.0.0.0 area 0 network 172.21.51.0 0.0.0.255 area 0 ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 172.21.51.193 ip http server no ip http secure-server ! ! ! ! ! ! ! ! ! control-plane ! ! ! voice-port 0/0/0:23 ! ccm-manager redundant-host 172.21.51.204 ccm-manager mgcp no ccm-manager fax protocol cisco ccm-manager music-on-hold ccm-manager config server 172.21.51.205 ccm-manager config ! mgcp mgcp call-agent 172.21.51.205 2427 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp modem passthrough voip mode nse mgcp package-capability rtp-package mgcp package-capability
Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected
Otto, Thanks for getting back to me. After spending a lot of time on this, I determined it was an underlying routing issue. The lab I was using was mis-configured on the PUB/SUB with the incorrect Default GW. Therefore, the DHCP server would work at Layer 2, but then when the phones would try to get their configs, they couldn't get a response from the PUB/SUB. Also there may have been an inter-VLAN routing issue as well, but I had tried so many different things I can't recall if I had to fix that or not. Basically, if DB replication is working, it most likely will have to be an underlying routing problem for this error to occur is what I have found. Jeff From: Otto Sanchez [mailto:o...@ipexpert.com] Sent: Friday, February 05, 2010 5:43 AM To: Jeff Price (jeffpric); kavi ten Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected Hi Jeff, Kavi, A cause for this error is that the auto registration number range has been exhausted (i.e. at some point in time the dn numbers have been assigned), please increase that range and try again, also, make sure your pub and sub have different ranges, If still with problems, I would think that the sub server is not getting the auto-registration settings from cisco unified cm configuration section, so as Roger mentioned this might be a replication issue, Then try to force a database replication from the pub server's cli, if it still doesn't work, please send us the sub call manager service traces when the phone is auto registering, Thanks, On Thu, Feb 4, 2010 at 3:04 PM, Jeff Price (jeffpric) jeffp...@cisco.com wrote: Anyone have any ideas? I'm still stuck here. I can't find anything on google relating to this error, other than IP communicator stuff. I don't believe its DB replication, because the Phones have replicated to the SUB when I am logged into the CM Administration page of the SUB. However, I am in a lab that has restricted access to CLI, OS Admin Page, and DRS page, so I'm unable to verify other than logging into the SUB CM Admin page. Thanks, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price (jeffpric) Sent: Thursday, February 04, 2010 9:55 AM To: Roger Källberg; kavi ten; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected Kavi, I am having the same issue. I will let you know if I have any success in finding a solution. I'm asking if you don't mind doing the same. Thank you all for your help. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg Sent: Thursday, February 04, 2010 9:39 AM To: kavi ten; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected Sound like a db replication issue or possibly, but less likely, the order of CPE in the call manager group. Roger Källberg Unified Communication Consultant Cygate AB From: kavi ten [mailto:kaucc...@gmail.com] Sent: den 4 februari 2010 14:23 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected Hi Guys, I have my DHCP server as Publisher DHCP Server: PUB Primary TFTP : 10.10.210.11 Secondary TFTP : 10.10.210.10 Auto reguistration enabed for SUB with range specified. Now the phone shown in the Devices-- Phones page but Status Rejected On the Phone it shows Rejectration Rejected: Security Error When I auto register with PUB it registers properly. What could be the problem when Auto regiosteration is enabled in the SUB. Thanks, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected
Anyone have any ideas? I'm still stuck here. I can't find anything on google relating to this error, other than IP communicator stuff. I don't believe its DB replication, because the Phones have replicated to the SUB when I am logged into the CM Administration page of the SUB. However, I am in a lab that has restricted access to CLI, OS Admin Page, and DRS page, so I'm unable to verify other than logging into the SUB CM Admin page. Thanks, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price (jeffpric) Sent: Thursday, February 04, 2010 9:55 AM To: Roger Källberg; kavi ten; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected Kavi, I am having the same issue. I will let you know if I have any success in finding a solution. I'm asking if you don't mind doing the same. Thank you all for your help. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg Sent: Thursday, February 04, 2010 9:39 AM To: kavi ten; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected Sound like a db replication issue or possibly, but less likely, the order of CPE in the call manager group. Roger Källberg Unified Communication Consultant Cygate AB From: kavi ten [mailto:kaucc...@gmail.com] Sent: den 4 februari 2010 14:23 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected Hi Guys, I have my DHCP server as Publisher DHCP Server: PUB Primary TFTP : 10.10.210.11 Secondary TFTP : 10.10.210.10 Auto reguistration enabed for SUB with range specified. Now the phone shown in the Devices-- Phones page but Status Rejected On the Phone it shows Rejectration Rejected: Security Error When I auto register with PUB it registers properly. What could be the problem when Auto regiosteration is enabled in the SUB. Thanks, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
