Re: [OSL | CCIE_Voice] ISDN Channel not available :D
Maybe you configured more channels for the controller as available on the PSTN Router Inbound calls are taking the first channel, outgoing the last channel. As far as I remember the HQ router on proctorlabs has only 6 channels, site B and C have 4 channels. Am Donnerstag, 10. Januar 2013 schrieb Abel ... : Try inbound from the 911 line and check the channel used. On Wed, Jan 9, 2013 at 6:52 PM, Nicolas MICHEL mcl.nico...@gmail.comwrote: Inbound calls are working perfectly :( *Jan 9 08:59:43.227: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x0084 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Display i = 'Emergency Services' Calling Party Number i = 0x0080, '911' Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '2123945002' Plan:ISDN, Type:National *Jan 9 08:59:43.251: ISDN Se0/0/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x8084 Channel ID i = 0xA98381 Exclusive, Channel 1 HQ-RTR(config-if)# *Jan 9 08:59:43.267: ISDN Se0/0/0:23 Q931: TX - ALERTING pd = 8 callref = 0x8084 *Jan 9 08:59:44.191: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8 callref = 0x0084 Cause i = 0x8290 - Normal call clearing *Jan 9 08:59:44.195: ISDN Se0/0/0:23 Q931: TX - RELEASE pd = 8 callref = 0x8084 *Jan 9 08:59:44.207: ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x0084 HQ-RTR(config-if)# Le Wednesday, January 09, 2013 11:50:14 PM, Abel ... a écrit : Try an inbound call from the pstn and run your debugs. Check the channel used in that call. That test is to discard L1/L2 issues. On Jan 9, 2013 6:21 PM, Cory Gray corygray22...@hotmail.com mailto:corygray22842@hotmail.**com wrote: I will say this. This happened to me once using MGCP and could never figure it out. I just assumed something was messed up and could not find an answer in over 2 hours of troubleshooting. If you ever figure it out, please let me know. -Original Message- From: ccie_voice-bounces@**onlinestudylist.com mailto:ccie_voice-bounces@**onlinestudylist.com [mailto:ccie_voice-bounces@**onlinestudylist.com mailto:ccie_voice-bounces@**onlinestudylist.com] On Behalf Of Nicolas MICHEL Sent: Wednesday, January 09, 2013 12:49 PM To: Bill Cc: OSL Voice Subject: Re: [OSL | CCIE_Voice] ISDN Channel not available :D Hi Guys, no the HQ is H323 otherwise I would not change the ISDN bchan order in the IOS but rather in the CUCM interface. the ISDN status shows Multi Frame Establish, all is all right. I can have incoming calls to the HQ but not outgoing calls to the Local and LD route pattern. From what I remember, International calls are working great :( Next time I'll debug Q931 with detail enabled Thanks for the help :) Nic Le 1/9/2013 11:41 AM, Bill a écrit : So is you hq a mgcp? If so it looks like you are missing your isdn bind-l3 ccm command Is it h323 or maybe stand alone cme with just a plan pri? If h323 the does it show status in gateways? Does it show unknown unknown instead of unknown and ip? Try to give us more and clearer information because right now all we know is pri, but most likely hq is a gateway for CUCM and we don't know what type. We don't know if this is your lab, rack rental or r ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] VOL2 lab 1 - HQ Phone cannot register ? WTH
Hi Nicolas, I had this problem too on some pods, where only one of the HQ phones want to register, same for site B, dhcp addressing is working but no registration occurs. Because I had no physical access on the phones to check what the display is showing ( I guess something like registration rejected), I tried several things to avoid this issue for the next session. In most cases it has worked when I used completly different dhcp ranges in the subnet, e.g 200.70 - 200.80. When 7960s had problems to register it helps to configure them manually in the cucm. Afterwards they are shown as registered. In some circumstances it only had solved the issue when I configured a dhcp pool on the router (for siteB probs) or on the HQ Sw for the HQ phones. Am Dienstag, 8. Januar 2013 schrieb Nicolas MICHEL : Hey guys. I m in trouble with the phone connected to Fa1/0/23 of the 3750 in HQ. Vlan assignement are OK: HQ-3750#sh run int fa1/0/23 Building configuration... Current configuration : 113 bytes ! interface FastEthernet1/0/23 switchport access vlan 10 switchport voice vlan 20 spanning-tree portfast end HQ-3750#sh run int fa1/0/1 Building configuration... Current configuration : 153 bytes ! interface FastEthernet1/0/1 switchport trunk encapsulation dot1q switchport trunk native vlan 10 switchport mode trunk speed 100 duplex full end HQ-3750#sh int trunk PortMode Encapsulation StatusNative vlan Fa1/0/1 on 802.1q trunking 10 PortVlans allowed on trunk Fa1/0/1 1-4094 PortVlans allowed and active in management domain Fa1/0/1 1,10,20,30 PortVlans in spanning tree forwarding state and not pruned Fa1/0/1 1,10,20,30 IP HELPER on the L3 Device is OK interface FastEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 end IP Phone can get an IP from the PUB (10.10.210.10) HQ-3750#sh cdp neigh fa1/0/23 det - Device ID: SEP0021A086825D Entry address(es): IP address: 10.10.200.50 Platform: Cisco IP Phone 7962, Capabilities: Host Phone Interface: FastEthernet1/0/23, Port ID (outgoing port): Port 1 Holdtime : 178 sec Version : SCCP42.8-4-1S advertisement version: 2 Duplex: full Power drawn: 6.300 Watts Power request id: 33373, Power management id: 1 Power request levels are:6300 0 0 0 0 Management address(es): the DHCP subnet is identical (except the IP range of course :) ) to the others DHCP pool which other phone use and can register to CUCM... I have no idea what is the problem here since it is a very basic problem if anyone has experienced the same on POD 11 of PL ? Many thanks for your help on this __**_ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Remote access on lab.
Both is available on the desktop (in Brussels) Am Mittwoch, 9. Januar 2013 schrieb Abdullin Kamil : Hi men, On lab what program is used for remote access on UCCX? windows remote desktop, VNC or any other? Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Gatekeeper in Single zone : Call Processing engine prioritize
You can define a gw-priority for the trunks to each endpoint. Am Mittwoch, 9. Januar 2013 schrieb Ramcharan Arya : Hello, I have a small query about Gatekeeper in single zone configuration it is prefer PUB as call processing ( as per ARQ message). if lab requirement says make subscriber as call processing engine then it does not meet requirement. Does any one know the answer what is the solution of this tricky situation. Regards, Ramcharan Arya CCIE # 28926 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Connection Monitor Duration for SRST testing
I always put it down to 10s, not less because sometimes the wan connection is flapping/very low and so the phones flapping between cucm and srst. Am Mittwoch, 9. Januar 2013 schrieb Pixar Perfect : Hello, is it advisable to bring down the Connection Monitor Duration under Device Pool to a low value like 30 seconds to expedite the SRST testing in the lab? Does it affect grading if we happen to put it back to default? thx ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Hostname on the routers
No don't do that, also don't add or remove domain names. Am Sonntag, 6. Januar 2013 schrieb sanity insanity : hi Guys, I am just wondering if we are allowed to change host names on the routers . For example if the changing the router name on R1 to HQ and so too for the branches. -MJ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] QOS Best Pract
The problem with your example is that the frame-relay ip rtp header-compression is out-dated and the header compression should be in the policy-map in the rtp class. Also I don't see any less flexibilty in using auto qos, you can adjust all the values as with the manual way. Regards Steffen Am Samstag, 5. Januar 2013 schrieb Pixar Perfect : What is the best practice for the real lab QOS after running the AutoQos - use the AutoQoS nomenclature and make changes to the classmaps, policymaps and map-class OR copy the AutoQOS output on a notepad and rename policymaps, classmaps and mapclass? I typically follow the second route as it give me more flexibility to finalize the configs on the notepad and then just copy paste instead of changing the AutoQOS configs on the router itself. Any inputs , comments or suggestions especially from those who had success with the second approach? Example: ! class-map match-any RTP match ip dscp ef class-map match-any CONTROL match ip dscp cs3 ! policy-map VOIP class RTP priority 24 class CONTROL bandwidth 19 class class-default fair-queue ! ! interface Serial0/1/1:0.1 point-to-point bandwidth 384 frame-relay interface-dlci 201 class FRVOIP frame-relay ip rtp header-compression ! ! ! map-class frame-relay FRVOIP frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 frame-relay fragment 480 service-policy output VOIP ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] QOS Best Pract
No grading is not affected, because it is allowed to use manual qos config
and so you can use whatever you want for the names.
Am Samstag, 5. Januar 2013 schrieb Pixar Perfect :
Ignore the RTP header compressions part. My question is not technical but
from strategy point. Do you think the grading could be affected if we
remove the names built by the QOS script and use our custom names?
Thanks
--
Date: Sat, 5 Jan 2013 08:55:57 +0100
Subject: Re: [OSL | CCIE_Voice] QOS Best Pract
From: stbruen...@gmail.com javascript:_e({}, 'cvml',
'stbruen...@gmail.com');
To: pixarperf...@live.com javascript:_e({}, 'cvml',
'pixarperf...@live.com');
CC: ccie_voice@onlinestudylist.com javascript:_e({}, 'cvml',
'ccie_voice@onlinestudylist.com');
The problem with your example is that the frame-relay ip rtp
header-compression is out-dated and the header compression should be in the
policy-map in the rtp class.
Also I don't see any less flexibilty in using auto qos, you can adjust all
the values as with the manual way.
Regards
Steffen
Am Samstag, 5. Januar 2013 schrieb Pixar Perfect :
What is the best practice for the real lab QOS after running the AutoQos
- use the AutoQoS nomenclature and make changes to the classmaps,
policymaps and map-class OR copy the AutoQOS output on a notepad and rename
policymaps, classmaps and mapclass?
I typically follow the second route as it give me more flexibility to
finalize the configs on the notepad and then just copy paste instead of
changing the AutoQOS configs on the router itself. Any inputs , comments or
suggestions especially from those who had success with the second approach?
Example:
!
class-map match-any RTP
match ip dscp ef
class-map match-any CONTROL
match ip dscp cs3
!
policy-map VOIP
class RTP
priority 24
class CONTROL
bandwidth 19
class class-default
fair-queue
!
!
interface Serial0/1/1:0.1 point-to-point
bandwidth 384
frame-relay interface-dlci 201
class FRVOIP
frame-relay ip rtp header-compression
!
!
!
map-class frame-relay FRVOIP
frame-relay cir 364800
frame-relay bc 3648
frame-relay be 0
frame-relay mincir 364800
frame-relay fragment 480
service-policy output VOIP
!
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[OSL | CCIE_Voice] Passed
I went on Thursday to Brussels for my 2nd attempt and I got my results suprisingly today. I passed. CCIE 37891. I like to say thank you to all on you on the list which shares your questions and answers about the blueprint task and ipexpert labs. I also wanna say thank you to IPexpert for their really really good learning materials. Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Question for test class
Its not allowed to bring any electronical device into the exam room. Am Samstag, 5. Januar 2013 schrieb Marko Milivojevic : I would guess not. If you're concerned about it, you could open a case with the certifications support. -- Marko Milivojevic - CCIE #18427 (SP RS) Senior CCIE Instructor - IPexpert On Sat, Jan 5, 2013 at 9:31 AM, Chrysostomos Christofi ch.christ...@logicom.net javascript:; wrote: Hi Marko I mean into the testing class for the ccie lab -Original Message- From: Marko Milivojevic [mailto:mar...@ipexpert.com javascript:;] Sent: Σάββατο, 5 Ιανουαρίου 2013 7:27 μμ To: Chrysostomos Christofi Cc: Online Study (ccie_voice@onlinestudylist.com javascript:;) Subject: Re: [OSL | CCIE_Voice] Question for test class Into our class or the testing room? On Sat, Jan 5, 2013 at 5:07 AM, Chrysostomos Christofi ch.christ...@logicom.net javascript:; wrote: For many peoples this question maybe it’s funny , but if you know I would like to know if I can bring an electronic cigarette into the class ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Any good blogs link on DB Replication Troubleshooting ?
The dbreplication issues which you have at ipexpert/proctorlabs session you
can fix with an easy utils dbreplication repair all, nothing more.
Am Samstag, 5. Januar 2013 schrieb Chrysostomos Christofi :
Hi Ram
** **
** **
First check in PUB
** **
utils network connectivity To verify that PUB and SUB have connectivity**
**
** **
** **
1. Sub: utils dbreplication stop (wait for it to complete before going to
step 2.)
2. Pub: utils dbreplication stop (wait for it to complete before going to
step 3.)
3. Pub: utils dbreplication reset all (monitor the dbstatus in RTMT for
all 2’s.)
4. Reboot the cluster and re-verify in RTMT.
** **
Final:
** **
utils dbreplication runtimestate
** **
*From:* ccie_voice-boun...@onlinestudylist.com javascript:_e({}, 'cvml',
'ccie_voice-boun...@onlinestudylist.com'); [mailto:
ccie_voice-boun...@onlinestudylist.com javascript:_e({}, 'cvml',
'ccie_voice-boun...@onlinestudylist.com');] *On Behalf Of *Ramcharan Arya
*Sent:* Σάββατο, 5 Ιανουαρίου 2013 8:19 μμ
*To:* ccie_voice@onlinestudylist.com javascript:_e({}, 'cvml',
'ccie_voice@onlinestudylist.com');
*Subject:* [OSL | CCIE_Voice] Any good blogs link on DB Replication
Troubleshooting ?
** **
Hello,
** **
I am running on db replication issues very often.
** **
PUB Sub status are in these combinations 2:0, 3:0, 3:3, 4:4,
** **
Is any these structured procedure to fix these different state of db
replications.
** **
I have search IPX blogs but could not find information I was looking for.*
***
** **
Can you someone please share detail procedures about how to fix DB
replication between in following state:
** **
Thanks Regards,
Ramcharan Arya
** **
___
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Re: [OSL | CCIE_Voice] CUE CLI configuration
You don't need to write a complete cue config, most is done with factory defaults, only focus on integration, so ccn subsystem jtapi ( for cucm) ccn subsystem sip (for cme and srst) ccn trigger for sip and jtapi and how to create users and voicemail boxes. Am Samstag, 5. Januar 2013 schrieb CCIEing : Hello Again guys, Return to CUE configuration using CLI . So practicing CLI Imposes on us to know all the parameters for all applications, but there are a lot of parameters like the autoattendant app, right? is there any way to remember them from inside the cli cue help, does any one have a difficulty to learn them . Appreciate your cooperation On Fri, Jan 4, 2013 at 2:12 PM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Sorry ** ** Apologize wrong type ** ** Correction: I heard from a lot guys that they had troubles to access the cue through gui J ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chrysostomos Christofi *Sent:* Παρασκευή, 4 Ιανουαρίου 2013 1:09 μμ *To:* Ahmad Taamneh; William Bell *Cc:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] CUE CLI configuration ** ** I heard from a lot guys that they had troubles to access the cue through cli So we have to learn both ways ** ** Also its more fast with cli either with cme or cucm integration ** ** Regards Chrysostomos ** ** ** ** ** ** *From:* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] GK to CUE Direct Call
Did you see the call incoming on the cme? debug voip dial-peer? Am Freitag, 4. Januar 2013 schrieb Cory Gray : All, ** ** I noticed during the following setup that I cannot call DIRECTLY into CUE but if SiteC-Phone1 is busy or does not answer, forwarding the voicemail works. I am not even sure you would lose points for this but I always like to make sure things like this work. ** ** Setup is as follows. ** ** CUCM Site A GK Site C Gateway CUE ** ** Does anyone know what extra configuration is need to accomplish this? ** ** Again, this Setup works fine ** ** CUCM Site A GK Site C Gateway SiteC Phone 1 CUE ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Terminal used for lab is putty or CRT or any other terminal
There is a trial version for 30 days.
Am Sonntag, 30. Dezember 2012 schrieb singh :
hi Guys,
Thanks for your inputs.
I generally use command prompt and putty to access my routers and servers
hence I am not sure how different is secure crt from these terminals.
Besides this I see it is a paid software ( secure crt) unlike putty (
which is free) .
1) How different is secure crt from other terminals in operation (
keyboard shortcuts , and other tabs)?
2)Do we need to practice with secure crt as well to prevent problems that
may arise during the exam ?
The expenses involved in studying for ccie program seem to just be going
higher and higher by the day : - )
-singh
-- Original message --
From:Steffen Bruening stbruen...@gmail.com javascript:_e({}, 'cvml',
'stbruen...@gmail.com');
Date: 30 Dec 12 00:05:55
Subject: Re: [OSL | CCIE_Voice] Terminal used for lab is putty or CRT or
any other terminal
To: Marko Milivojevic mar...@ipexpert.com javascript:_e({}, 'cvml',
'mar...@ipexpert.com');
Cc: ; OSL Voice **; ccie_voice-requ...@onlinestudylist.comjavascript:_e({},
'cvml', 'ccie_voice-requ...@onlinestudylist.com');
**
I can confirm it is SecureCRT, but an old version without tabs. I training
with desktop shortcuts to the device connections which opens in seperate
windows.
2012/12/29 Marko Milivojevic mar...@ipexpert.com javascript:_e({},
'cvml', 'mar...@ipexpert.com');
I *think* Voice lab is still using SecureCRT. Does it matter whether
it's SSH or Telnet? :-)
--
Marko Milivojevic - CCIE #18427 (SP RS)
Senior CCIE Instructor - IPexpert
On Sat, Dec 29, 2012 at 9:24 AM, singh singh8...@in.comjavascript:_e({},
'cvml', 'singh8...@in.com');
wrote:
hi Guys,
I am interesting in knowning the following ...
1) From the CCIE voice lab which is the terminal connnection used (
putty or
crt)?
2)Is it ssh or telnet?
-singh
Get Yourself a cool, short @in.com Email ID now!
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please
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___
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visit www.ipexpert.com
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Get Yourself a cool, short *@in.com* Email ID
now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing
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Re: [OSL | CCIE_Voice] Long distance calls should include 1 or not ?
When they ask for 10 digits you have to send only the last ten digits. your RP looks good, on your dial-peer you don't need the forward digits command, because 9 and 1 are the only digits which are clearly seen they will be stripped automatically Am Samstag, 29. Dezember 2012 schrieb sanity insanity : hi Guys , For long distance calls I use the following patterns... CUCM :- 91.[2-9]XX[2-9]XX predot strip H323 gateway :- dial-peer voice 10 pots description srst 10 digit LD destination-pattern 91[2-9]..[2-9].. port 0/0/0:23 forward-digits 10 Questions : 1) The requirement is that 10 digits need to be sent to PSTN. Does this require the 1 to be included when sent to pstn or is it just [2-9]..[2-9].. as per NANP? 2) Is my understanding correct . I see some srnds in which the 1 is also sent to PSTN. Regards, MJ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Terminal used for lab is putty or CRT or any other terminal
I can confirm it is SecureCRT, but an old version without tabs. I training with desktop shortcuts to the device connections which opens in seperate windows. 2012/12/29 Marko Milivojevic mar...@ipexpert.com I *think* Voice lab is still using SecureCRT. Does it matter whether it's SSH or Telnet? :-) -- Marko Milivojevic - CCIE #18427 (SP RS) Senior CCIE Instructor - IPexpert On Sat, Dec 29, 2012 at 9:24 AM, singh singh8...@in.com wrote: hi Guys, I am interesting in knowning the following ... 1) From the CCIE voice lab which is the terminal connnection used ( putty or crt)? 2)Is it ssh or telnet? -singh Get Yourself a cool, short @in.com Email ID now! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cisco Unity Connection - VM pilot and VM profile made default but not applied automatically on phones
Maybe it is a good idea to use just the default profile for all the phones which have CUCN mailbox and only to add a new profile to the users who get a CUE mailbox. 2012/12/29 CCIEing aboaz...@gmail.com Hi virajith, You may update all your phone settings using bulk edit. On Sat, Dec 29, 2012 at 8:36 PM, virajith vir...@rediffmail.com wrote: hi All, I am noticing that for my cisco unity connection integrated with CUCM - the VM pilot and profile that I have made as default VM pilot and profile is not getting automatically applied on my phones and therefore any call forward busy or no answer gets a busy tone. I have to manually update the VM profile on the DN level of the phone to indicate that the default is the unity connection. How do I correct this behaviour so that the phones automatically can get the default VM settings without me having to specify it? -Vir http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? Catch India as it happens with the *Rediff News App*. To download it for FREE, click herehttp://track.rediff.com/click?url=___http://www.rediff.com/newsapp___lnk=signaturenewservice=newsapp ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE correct procedure...
Hi singh, connet to the cue, check/install license then put the module offline (conf t, offline) when it is in offline state run restore factory default and wait till it ask to press any key for reload. After the reload it will go through the intial configuration wizard, when you completed that you can configure the integration you ask for. Am Mittwoch, 26. Dezember 2012 schrieb singh : Guys, How do I verify if a CUE module is in the factory default or not? -If the CUE module is factory default then I am thinking ... 1) to first check the license installed . If the license installed is not correct then will install the license 2) complete all config on cue via the cli 3) reload the CUE module ( please note the reload will be done only once) Is the steps (1-3) the correct way of doing it as reloading once would say time instead of having to do it again and again. Let me know if the points 1-3 is the correct procedure? -singh Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MGCP gateway : DDI for International Route Pattern options
Why is predot trailing needed for mgcp? I use always only predot and it works perfect. 2012/12/22 Chrysostomos Christofi ch.christ...@logicom.net Ramcharan , hi ** ** Both route patterns for the below are correct It depends what the question ask , or what is your requirement ** ** **1) **With # in the h323 you don’t have to use Predot trailing # you can use only Predot but for MGCP GW its need it. But its not any harm if you use Predot trailing # for h323 also Regards ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ramcharan Arya *Sent:* Σάββατο, 22 Δεκεμβρίου 2012 8:14 μμ *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] MGCP gateway : DDI for International Route Pattern options ** ** Hello, I have a requirement to use SLRG for International destination: First gateway : Site B ( H.323) Second gateway : Site A (MGCP) My route patterns are below: RP : 9011.! pt: pt-sa-int RP : 9011.i# pt: pt-sa-int Option : 1 RL: SLRG DDI: NANP: Predot Option : 2 RL: SLRG DDI: NANP: Predot trailing # Can you suggest which DDI is correct correct for the above route pattern. Thanks Regards, Ramcharan Arya CCIE # 28926 (RS) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA
What so you want to see? You are calling from a PSTN number into the XML Script. When you call a internal number through the mva, you will the number am when the destination hooks up and the call is connected you will also see the name. Am Sonntag, 23. Dezember 2012 schrieb Rrcrumm : Hello I have set up MVA and it seems to be working properly One question, Are we suppose to see name and number when we make a call to the MVA number and then press one to dial another number and we cal an internal extension? Thanks Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA
Only in the connect state, not in the ringing state
Am Sonntag, 23. Dezember 2012 schrieb Rrcrumm :
Sorry
I call from the local site B PSTN phone that is configured for a RDP and RD
I call the MVA number put in my pin, press 1 to dial a number and dial
2001.
When I look at the screen of hqph1 I just see the extension 3001 is
calling, no name.
Should I see a name and number?
Thanks
Randall
On Dec 22, 2012, at 10:36 PM, Steffen Bruening
stbruen...@gmail.comjavascript:_e({}, 'cvml', 'stbruen...@gmail.com');
wrote:
What so you want to see? You are calling from a PSTN number into the XML
Script. When you call a internal number through the mva, you will the
number am when the destination hooks up and the call is connected you will
also see the name.
Am Sonntag, 23. Dezember 2012 schrieb Rrcrumm :
Hello
I have set up MVA and it seems to be working properly
One question,
Are we suppose to see name and number when we make a call to the MVA
number and then press one to dial another number and we cal an internal
extension?
Thanks
Randall
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Re: [OSL | CCIE_Voice] ProctorLab Phones 7960/7962
7962s are not guranteed. HQ SW is connected with one 7960 and one 7962,
site B also. Only site C has two 7962s at most of the PODs. The problem is
that for SiteB and C the phones not always connected directly, sometime
they are remotly connect via vpn, therefore the port is up, but cdp shows
nothing.
I am happy that I can mix up with my own phones, otherwise some sessions
are not very funny with only two physical phones and I don't like this ip
blue and x-lite stuff.
Am Freitag, 21. Dezember 2012 schrieb Mohamed Gazzaz :
Sometimes the phones are connected to different ports. Try to un-shut the
other ports and see if they show up.
Regards,
Mohamed Gazzaz
--
Date: Fri, 21 Dec 2012 14:18:34 +
From: sp1...@yahoo.co.uk javascript:_e({}, 'cvml',
'sp1...@yahoo.co.uk');
To: ccie_voice@onlinestudylist.com javascript:_e({}, 'cvml',
'ccie_voice@onlinestudylist.com');
Subject: [OSL | CCIE_Voice] ProctorLab Phones 7960/7962
Q. Do all the PL voice racks actually have 7962 phones connected to
the sw ports ?.
I have been using the racks for some and know about the various
'mis-configs' inserted for troubleshooting - but when you apply the same
phone config for 7960 phones ports and these phones appear in sh cdp and
the 7962 don't - it makes you wonder if the 7962 phone is actually
connected.
Normally I have my hardware phones on via a remote vpn link , but on
this session i don't and and trying to use pview. How do you do the OWLE
Labs using + dialling when you only see 7960 phones on the PL racks ?
Unfortunately this is not the first time I have come across missing 7962
and have also heard this mentioned by some attendees on a recent IP expert
bootcamp.
So what do you do to verify whether the rack has 7962's ?
I guess IPEx/PL don't guarentee every rack has 7962 phones as when on a
bootcamp they had to tweak the configs [using ILMED ?] to ensure the the
candidates actually had the 7962 phones on their PODS they could do the
labs.
I have emailed the support team to verify but they have not/do not reply.
Seems i'm screwed on this session.
Any opinions ?
.Sanjay
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Re: [OSL | CCIE_Voice] Documents and other details
Hi, you need one identy document, passport for example. Lunch is included in the exam fee, you will get a coupon for one meal, salat, drink and dessert. In Brussels waa also a machine near the exam room were you get water and tea for free during the exam. Regards Steffen Am Donnerstag, 20. Dezember 2012 schrieb virajith : hi Guys, I have a question regarding if we need to carry any identity documents when we go to the exam center such as - driver's license or passport photocopy . - Also do we need to carry lunch and water or is it provided? - any other details or documents we need to carry with us when we go to the exam center for taking the exam? - Is it a desktop they give us to operate? Does it have 2 monitors? Thanks, Vir http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? Catch India as it happens with the *Rediff News App*. To download it for FREE, click herehttp://track.rediff.com/click?url=___http://www.rediff.com/newsapp___lnk=signaturenewservice=newsapp ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] VOL1 : Lab 9 Task 2: BACD Problem
Hi, I would always go with a pots dial-peer. Your external BACD dial in number is truncated to +343500 and therefore it could not find a matching dial-peer, check your voice translation. Am Montag, 17. Dezember 2012 schrieb Ramcharan Arya : Hello, I have configured BACD on Brach2 site CME When I dial pilot number 3500 from PSTN Line 4 : 3214-1891 It generate reorder tone and call is coming on router and matching outbound dial-peer but script is not working. Below is my configuration and debug output. service queue flash:bacdprompts/app-b-acd-2.1.2.2.tcl param queue-len 15 param aa-hunt10 3006 param queue-manager-debugs 1 param aa-hunt2 3210 param number-of-hunt-grps 2 ! service aa flash:bacdprompts/app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 param number-of-hunt-grps 2 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 3500 paramspace english location flash:bacdprompts/ param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param voice-mail 3001 param max-time-call-retry 90 param service-name queue ! dial-peer voice 222 voip service aa destination-pattern 3500 session target ipv4:10.10.110.3 incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad --- Dec 17 16:27:23.328: ISDN Se0/0/0:15 Q931: RX - SETUP pd = 8 callref = 0x00D3 Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Calling Party Number i = 0x2180, '32141891' Plan:ISDN, Type:National Called Party Number i = 0x80, '+3432143500' Plan:Unknown, Type:Unknown Dec 17 16:27:23.328: ISDN Se0/0/0:15 Q931: Received SETUP callref = 0x80D3 callID = 0x0032 switch = primary-net5 interface = User Dec 17 16:27:23.332: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=32141891, Called Number=+343500, Voice-Interface=0x49FFDE10, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Dec 17 16:27:23.332: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1 Dec 17 16:27:23.332: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=32141891, Called Number=+343500, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_FAX Dec 17 16:27:23.332: //-1//DPM/dpAssociateIncomingPeerCore: Result=NO_MATCH(-1) After All Match Rules Attempt Dec 17 16:27:23.332: //-1//DPM/dpMatchPeersCore: Calling Number=, Called Number=+343500, Peer Info Type=DIALPEER_INFO_SPEECH Dec 17 16:27:23.332: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=+343500 Dec 17 16:27:23.332: //-1//DPM/dpMatchPeersCore: No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1) Dec 17 16:27:23.332: //-1//DPM/dpMatchPeers: Result=NO_MATCH(-1) Dec 17 16:27:23.340: //-1/78994D148033/DPM/dpMatchPeersCore: Calling Number=, Called Number=+343500, Peer Info Type=DIALPEER_INFO_SPEECH Dec 17 16:27:23.340: //-1/78994D148033/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=+343500 Dec 17 16:27:23.340: //-1/78994D148033/DPM/dpMatchPeersCore: No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1) Dec 17 16:27:23.340: //-1/78994D148033/DPM/dpMatchPeersMoreArg: Result=NO_MATCH(-1) Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore: Calling Number=+343500, Called Number=+343500, Peer Info Type=DIALPEER_INFO_SPEECH Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=+343500 Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore: No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1) Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersMoreArg: Result=NO_MATCH(-1) Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore: Calling Number=+343500, Called Number=+343500, Peer Info Type=DIALPEER_INFO_SPEECH Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=+343500 Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore: No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1) Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersMoreArg: Result=NO_MATCH(-1) Dec 17 16:27:23.340: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=+343500, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP,
Re: [OSL | CCIE_Voice] CUCM TFTP service restart from CLI
Thats not allowed via CLI, so no syntax available. Am Montag, 3. Dezember 2012 schrieb Ramcharan Arya : Hello, Can you someone please tell me command syntax how to restart Cisco Tftp service from CUCM CLI.? Thanks Regards, Ramcharan Arya CCIE # 28926 (RS) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Interface for binding?
You don't have to change the routing, routing is done for you. You can the
reach the CUCM from any interface.
Am Sonntag, 2. Dezember 2012 schrieb virajith :
hi Steffen,
Thanks for your reply
If we can bind to any interface ( gig , loopback ) then is it necessary
for us to setup the the routing as well as traffic from loopback or
virtual interfaces needs to reach the callmanger and other devices also if
the routing needs to setup then this is going to eat into the allocated
exam time?
-Vir
From: Steffen Bruening stbruen...@gmail.com javascript:_e({}, 'cvml',
'stbruen...@gmail.com');
Sent: Sat, 01 Dec 2012 12:24:35
To: virajith vir...@rediffmail.com javascript:_e({}, 'cvml',
'vir...@rediffmail.com');
Cc: ccie_voice@onlinestudylist.com javascript:_e({}, 'cvml',
'ccie_voice@onlinestudylist.com');
ccie_voice@onlinestudylist.comjavascript:_e({}, 'cvml',
'ccie_voice@onlinestudylist.com');
Subject: Re: [OSL | CCIE_Voice] Interface for binding?
Generally it doesn't matter to which you bind your media resources. But
maybe the question ask you for a specific binding.
The only thing you never should do is to bind Cube and Gatekeeper to the
same Interface.
Regards
Steffen
Am Samstag, 1. Dezember 2012 schrieb virajith :
Hi All,
I am wondering which would be the best interface to bind for transcoders ,
gateways , cfbs and for sccp configurations I know loopback is the best for
binding but with like to clarify with the example below...
For example...
R3#sh ip int brief
Interface IP-Address OK? Method Status Protocol
Serial0/0/1:0 unassigned YES NVRAM up up
Serial0/0/1:0.101 142.1.67.2 YES NVRAM up up
Vlan1unassigned YES NVRAM up up
Vlan502 142.102.66.254 YES NVRAM up up
Vlan602 142.202.66.254 YES NVRAM up up
SSLVPN-VIF0 unassigned NO unset up up
Loopback0 142.1.66.254 YES NVRAM up up
In the above scenario interface vlan502 is the is in the voice vlan
interface with the ip address of 142.102.66.254 . However I have the
loopback ip address is 142.1.66.254.
My Servers ( CUCM , CUC, CUPS, CCX) are in 142.100.64.x
In this situation which is the best interface to bin with and why ?
-Vir
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Re: [OSL | CCIE_Voice] UCCX Prompt Question
Hi Bill, When you ask for callers ahead of you, you should go with ContactsWaiting+decrement step, because position in Queue is your position and when nobody else is in the queue you will hear a - 1 after the decrement, don't know whether there is also an increment step. At the end it is your choice how you get your script working. Not the way, only the result is graded. Regards Steffen Am Freitag, 30. November 2012 schrieb William Bell : I have a question that may really just come down to a matter of preference. However, I want to make sure there is something that I am not missing. For those who want to read along my question stems from IPexpert's One Week Lab Experience lab 2. I also think i have seen a similar question in the 5-lab workbook. For everyone else, the CCX requirement is to play a contact's position in queue while they are in the queued branch of Select Resource. Specifically, they want you to play a prompt that says The number of people ahead of you is one (or two, or three, etc.). The way I do this is as follows: step: Select Resource from CSQ - (Connected) - (Queued) label: queueLoop intPosInQ = Get Reporting Statistic PositionInQueue decrement intPosInQ playPrompt (P[YourPosinQ.wav] + intPosInQ) delay 30s goto label: queueLoop The way I have seen IPExpert handle this has a few more steps: step: Select Resource from CSQ - (Connected) - (Queued) label: queueLoop intPosInQ = Get Reporting Statistic PositionInQueue decrement intPosInQ promptNumInQ = Create Generated Prompt number (intPosInQ) promptQueue = Create Container Prompt Concatenation (P[YourPosinQ] + promptNumInQ) playPrompt (promptQueue) delay 30s goto label: queueLoop When I use my method, I get the desired result. My question is what (if any) advantage is there in generating the spoken prompt and packaging the two prompts instead of just doing the concatenation in-line with the Play Prompt step? Thanks in advance. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] UCCX Prompt Question
Hi Bill,
I had my first attempt in November, but failed because I don't read
some question corectly
in every detail.
I had this problem with -1 in some of my lab sessions (not proctor
labs) and it worked with contacts waiting for me. I tested it with 3
call-in users and all get an indivudually annoucement for the amount of
calls ahead.
But as I wrote, it doesn't matter how the script looks like as long as it
fit the question in your binder.
Regards
Steffen
Am Freitag, 30. November 2012 schrieb Bill Lake :
Sounds like either way works, so as long as it works, it really doesn't
matter how you get it, just as long as the proctor gets what he is supposed
to hear.
I know I have used the slightly longer way as it helps me stay organized
but next lab session I am going to try this if I can keep my ducks in a row
:)
I also know from my practice that it does not say -1 when I set this up
using Position in queue, it says your position in queue is zero or whatever
I record.
I also practice recording in both in CUC and UCCX just in case one does
not work. You know both methods so just practice because you never know :)
It seems like you might be getting close to your lab, do you have one
schedule?
Bill
On Fri, Nov 30, 2012 at 4:05 PM, William Bell
b...@ucguerrilla.comjavascript:_e({}, 'cvml', 'b...@ucguerrilla.com');
wrote:
I have a question that may really just come down to a matter of
preference. However, I want to make sure there is something that I am not
missing. For those who want to read along my question stems from
IPexpert's One Week Lab Experience lab 2. I also think i have seen a
similar question in the 5-lab workbook.
For everyone else, the CCX requirement is to play a contact's position in
queue while they are in the queued branch of Select Resource.
Specifically, they want you to play a prompt that says The number of
people ahead of you is one (or two, or three, etc.).
The way I do this is as follows:
step: Select Resource from CSQ
- (Connected)
- (Queued)
label: queueLoop
intPosInQ = Get Reporting Statistic PositionInQueue
decrement intPosInQ
playPrompt (P[YourPosinQ.wav] + intPosInQ)
delay 30s
goto label: queueLoop
The way I have seen IPExpert handle this has a few more steps:
step: Select Resource from CSQ
- (Connected)
- (Queued)
label: queueLoop
intPosInQ = Get Reporting Statistic PositionInQueue
decrement intPosInQ
promptNumInQ = Create Generated Prompt number (intPosInQ)
promptQueue = Create Container Prompt Concatenation (P[YourPosinQ] +
promptNumInQ)
playPrompt (promptQueue)
delay 30s
goto label: queueLoop
When I use my method, I get the desired result. My question is what (if
any) advantage is there in generating the spoken prompt and packaging the
two prompts instead of just doing the concatenation in-line with the Play
Prompt step?
Thanks in advance.
-Bill
--
William Bell
blog: http://ucguerrilla.com
twitter: @ucguerrilla
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Re: [OSL | CCIE_Voice] Interface for binding?
Generally it doesn't matter to which you bind your media resources. But maybe the question ask you for a specific binding. The only thing you never should do is to bind Cube and Gatekeeper to the same Interface. Regards Steffen Am Samstag, 1. Dezember 2012 schrieb virajith : Hi All, I am wondering which would be the best interface to bind for transcoders , gateways , cfbs and for sccp configurations I know loopback is the best for binding but with like to clarify with the example below... For example... R3#sh ip int brief Interface IP-Address OK? Method Status Protocol Serial0/0/1:0 unassigned YES NVRAM up up Serial0/0/1:0.101 142.1.67.2 YES NVRAM up up Vlan1unassigned YES NVRAM up up Vlan502 142.102.66.254 YES NVRAM up up Vlan602 142.202.66.254 YES NVRAM up up SSLVPN-VIF0 unassigned NO unset up up Loopback0 142.1.66.254 YES NVRAM up up In the above scenario interface vlan502 is the is in the voice vlan interface with the ip address of 142.102.66.254 . However I have the loopback ip address is 142.1.66.254. My Servers ( CUCM , CUC, CUPS, CCX) are in 142.100.64.x In this situation which is the best interface to bin with and why ? -Vir http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? Catch India as it happens with the *Rediff News App*. To download it for FREE, click herehttp://track.rediff.com/click?url=___http://www.rediff.com/newsapp___lnk=signaturenewservice=newsapp ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Learn and Motivation Strategy for the second attempt
Hi all, I was in Bruessels yesterday and I failed. I was suprised about my score report and about the amount of section were lost points. I only know one definitly mistake, about the other things I could only suppose. It seems that I am just to stupid to read the questions. I was finished with the configuration after 5 1/2 hours and I tested everything twice, solved the issues I found, but unfortunately it was not enough. I booked my second attempt for January, but to be honest I am really frustated and I don't know how should keep my motivation and focus. Sure I will make further remote labs to keep my speed, but what else could I do till my second attempt? So all second/more attempter what have you made during your till your next lab seat? Did you changed or strategy/learning focus or anything else? Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ESW and switch config
Recommed by whom? Ipexpert, I know, but the problem is that the Proctors/Scripts which marks your lab are not from IPExpert they are from Cisco. Therefore it should be better to follow the cisco guidelines: You should configure voice VLAN on switch access ports; voice VLAN is not supported on trunk ports. You can only configure a voice VLAN on Layer 2 ports. http://www.ciscosystems.com/en/US/docs/switches/metro/catalyst3750m/software/release/12.2_25_ey/configuration/guide/swvoip.html#wp1030825 2012/11/1 Cory Gray corygray22...@hotmail.com You should be fine without one just know that the recommended ESW config is Switchport mode trunk Switch trunk native vlan X (X equals data vlan) Switch voice vlan Y (y equals voice vlan) ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *virajith *Sent:* Wednesday, October 31, 2012 11:22 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] ESW and switch config ** ** hi Guys, I am just wondering if one can practice the labs without a ESW module . I have a setup in with I am using a switch (3750) for the vlan config. Is an ESW module necessary for the lab practice? How is the above config on 3750 different from using an ESW module? -Vir http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? Catch India as it happens with the *Rediff News App*. To download it for FREE, click herehttp://track.rediff.com/click?url=___http://www.rediff.com/newsapp___lnk=signaturenewservice=newsapp ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Uccx scripting making me mad!!
Here you can find some scripting guides: http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_programming_reference_guides_list.html 2012/11/1 sanity insanity networksanitytoinsan...@gmail.com hi Guys, I really need your help to understand UCCX scripts ...How they are made? and how they work? Please help guys! -Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ESW and switch config
Oh okay. I apologize for the confusion I made. 2012/11/1 Cory Gray corygray22...@hotmail.com IPexpert and myself have run into problems doing it the traditional way. But for the authoritative source you should be looking at the LAN Switching Guide for IOS 12.4T which confirms that recommendation. The guide you point to is for Metro 3750 switches. The ISR routers in the lab run 12.4T. http://www.cisco.com/en/US/docs/ios-xml/ios/lanswitch/configuration/12-4t/lsw-hwic-ethsw-ic.html#GUID-379450C0-5434-4AC3-9BED-396CB7D162C1 -- Date: Thu, 1 Nov 2012 08:08:36 + Subject: Re: [OSL | CCIE_Voice] ESW and switch config From: stbruen...@gmail.com To: corygray22...@hotmail.com CC: vir...@rediffmail.com; ccie_voice@onlinestudylist.com Recommed by whom? Ipexpert, I know, but the problem is that the Proctors/Scripts which marks your lab are not from IPExpert they are from Cisco. Therefore it should be better to follow the cisco guidelines: You should configure voice VLAN on switch access ports; voice VLAN is not supported on trunk ports. You can only configure a voice VLAN on Layer 2 ports. http://www.ciscosystems.com/en/US/docs/switches/metro/catalyst3750m/software/release/12.2_25_ey/configuration/guide/swvoip.html#wp1030825 2012/11/1 Cory Gray corygray22...@hotmail.com You should be fine without one just know that the recommended ESW config is Switchport mode trunk Switch trunk native vlan X (X equals data vlan) Switch voice vlan Y (y equals voice vlan) ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *virajith *Sent:* Wednesday, October 31, 2012 11:22 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] ESW and switch config ** ** hi Guys, I am just wondering if one can practice the labs without a ESW module . I have a setup in with I am using a switch (3750) for the vlan config. Is an ESW module necessary for the lab practice? How is the above config on 3750 different from using an ESW module? -Vir http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm%40Middle? Catch India as it happens with the *Rediff News App*. To download it for FREE, click herehttp://track.rediff.com/click?url=___http://www.rediff.com/newsapp___lnk=signaturenewservice=newsapp ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ESW and switch config
I still need the sw access mode, because there is no spanning-tree portfast trunk command for the ESW and so I would lose the marks for the vlan section because in most cases it is required to bypass all the spanning-tree states or just that the phones should boot up as fast as possible. Regards Steffen 2012/11/1 Steffen Bruening stbruen...@gmail.com Oh okay. I apologize for the confusion I made. 2012/11/1 Cory Gray corygray22...@hotmail.com IPexpert and myself have run into problems doing it the traditional way. But for the authoritative source you should be looking at the LAN Switching Guide for IOS 12.4T which confirms that recommendation. The guide you point to is for Metro 3750 switches. The ISR routers in the lab run 12.4T. http://www.cisco.com/en/US/docs/ios-xml/ios/lanswitch/configuration/12-4t/lsw-hwic-ethsw-ic.html#GUID-379450C0-5434-4AC3-9BED-396CB7D162C1 -- Date: Thu, 1 Nov 2012 08:08:36 + Subject: Re: [OSL | CCIE_Voice] ESW and switch config From: stbruen...@gmail.com To: corygray22...@hotmail.com CC: vir...@rediffmail.com; ccie_voice@onlinestudylist.com Recommed by whom? Ipexpert, I know, but the problem is that the Proctors/Scripts which marks your lab are not from IPExpert they are from Cisco. Therefore it should be better to follow the cisco guidelines: You should configure voice VLAN on switch access ports; voice VLAN is not supported on trunk ports. You can only configure a voice VLAN on Layer 2 ports. http://www.ciscosystems.com/en/US/docs/switches/metro/catalyst3750m/software/release/12.2_25_ey/configuration/guide/swvoip.html#wp1030825 2012/11/1 Cory Gray corygray22...@hotmail.com You should be fine without one just know that the recommended ESW config is Switchport mode trunk Switch trunk native vlan X (X equals data vlan) Switch voice vlan Y (y equals voice vlan) ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *virajith *Sent:* Wednesday, October 31, 2012 11:22 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] ESW and switch config ** ** hi Guys, I am just wondering if one can practice the labs without a ESW module . I have a setup in with I am using a switch (3750) for the vlan config. Is an ESW module necessary for the lab practice? How is the above config on 3750 different from using an ESW module? -Vir http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm%40Middle? Catch India as it happens with the *Rediff News App*. To download it for FREE, click herehttp://track.rediff.com/click?url=___http://www.rediff.com/newsapp___lnk=signaturenewservice=newsapp ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] did issue
Q931 doesnt lie. Something is missing in your config, maybe on the cucm, maybe on the gateway (for h323). It depends on where your stripping your DIDs to the internal number format. 2012/10/30 otunola Akerele otunola.aker...@gmail.com hi all, am new to the forum, ples am having some issues am using proctorlabs and following the 5lab workbook 1.i get a fast busy when i place a call to the pstn from siteA and again what could be the issue if i place a call from the pstn to site b and the debug says unallocated/unknown number whereas every of the configuration looks fine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Unable to call AVT from PSTN 911
Hi, you don't need such a bunch of dial-peers for CUE You are using MWI outcalling, therefore you only need the both DN's also you can send Voicemal and AVT on the same Dial-Peer to the CUE, use 400[45] as destination-pattern. If outcalling is not request in the question I would prefer unsolicited notification to the CME. Check your AVT sip trigger on the cue cli. Regards Steffen 2012/10/25 Ramcharan Arya ramcharan.a...@gmail.com Hello, I am working on a task trying to call AVT extension 4005 from PSTN 911. I am able to call voice mail and auto-attendent but when I call AVT number I get re-order tone. Has anyone try AVT setup on CUE. Please let me know. I got another problem with MWI lamp is not working when use left voice mail to extension. telephony-service no auto-reg-ephone max-ephones 52 max-dn 192 ip source-address 10.10.128.1 port 2000 voicemail 4004 max-conferences 8 gain -6 moh music-on-hold.au multicast moh 239.23.4.10 port 2000 web admin system name Admin password admin dn-webedit time-webedit transfer-system full-consult ! ephone-dn 98 number 4010 mwi off ! ! ephone-dn 99 number 4011 mwi on ! dial-peer voice 98 voip description MWI OFF incoming called-number 4010 codec g711ulaw no vad ! dial-peer voice 99 voip description MWI ON incoming called-number 4011 codec g711ulaw no vad ! dial-peer voice 4004 voip destination-pattern 4004 session protocol sipv2 session target ipv4:10.10.128.2 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 4005 voip ** Description AVT *** destination-pattern 4005 session protocol sipv2 session target ipv4:10.10.128.2 dtmf-relay sip-notify codec g711ulaw no vad Can you please suggest is something is missing in my configuration. Thanks Regards, Ramcharan Arya ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] RSVP brokes transcoding
Hi Cory, that's the trick. Thanks a lot. 2012/10/20 Cory Gray corygray22...@hotmail.com Do you have g729r8 on the transcoder? It is not their by default so you would have to add it to the list. I always add it to conferencing and transcoder just to get in the habit. Sent from my iPhone On Oct 20, 2012, at 4:32 PM, Steffen Bruening stbruen...@gmail.com wrote: Hi, I have 3 Sites, all of them configured on the CUCM, Site C has Voicemail with local CUE. When I am dialing from Site B to C codec g279 will be used and I can reach the voicemail, so I know When I am dialing from Site A to C through an RSVP CAC Location I get a fast busy when reaching the VM Pilot of Site C. When I take of the RSVP I can also reach VM Pilot of Site C with G729. Maybe somebody can explain to me why these RSVP Calls to VM are failing. My workaround for this to create a new location for VM Ports and Route Points which does not use RSVP. But this will brake the requirement of the question to allow only 4 calls between HQ und SC over the WAN. Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] LAN Qos questions
I have this seen this also, to be honest I think it shouldn't matter whether it is in threshold 1 or 3 as long as no other COS is in same Threshold of queue 1 of queset 2. When you leave in in threshold 3 I think you should be fine with: mls qos queue-set output *2* threshold 1 100* *100* 75 *100. Maybe I am completly wrong but thats they way I understood this. Regards Steffen 2012/10/20 Pixar Perfect pixarperf...@live.com The requirement is as follows on Lab 2 QOS section of the 5-Lab Handbook. For traffic being sent to the Site A gateway ensure that the traffic marked with COS 5 is dropped if the queue 1 is 75% full The Solution guide (page 408) has the following solution. mls qos queue-set output *2* threshold 1 *75 100 100 100* -- queset is preconfigured on the port to 2 mls qos srr-queue output cos-map queue 1 threshold *3* 5 .. My interpretation was to move the Cos 5 into Q1t1 but the command says threshold 3 .. is this just a typo or am I missing something obvious. Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] LAN Qos questions
Hi Krishna, sounds good so far. But as I said when COS 5 is the only COS in Queue 1, which Traffic/COS could fill T1 and T2 up to 100% when COS 5 is only mapped to T3? Regards Steffen 2012/10/21 Krishna vinayak_...@yahoo.com steffen, your approach is not right way of doing it because when u look the threshold values of the queues you have allocated max threshold is 100 and reserved threshold is 100, guess what both threshold i.e. t1 and t2 takes up to 100% value when desired and that being said after t1 and t2 were filled it comes to t3 which has 75% i.e. it is the last threshold where it will take/borrow the memory value from reserved threshold when desired. long story short... right way of doing it either assign it to t2 or t1 and assign threshold value of 75% for correct approach... thank you krishna. -- *From:* Steffen Bruening stbruen...@gmail.com *To:* Pixar Perfect pixarperf...@live.com *Cc:* ccie_voice@onlinestudylist.com *Sent:* Saturday, October 20, 2012 6:38 PM *Subject:* Re: [OSL | CCIE_Voice] LAN Qos questions I have this seen this also, to be honest I think it shouldn't matter whether it is in threshold 1 or 3 as long as no other COS is in same Threshold of queue 1 of queset 2. When you leave in in threshold 3 I think you should be fine with: mls qos queue-set output *2* threshold 1 100* *100* 75 *100. Maybe I am completly wrong but thats they way I understood this. Regards Steffen 2012/10/20 Pixar Perfect pixarperf...@live.com The requirement is as follows on Lab 2 QOS section of the 5-Lab Handbook. For traffic being sent to the Site A gateway ensure that the traffic marked with COS 5 is dropped if the queue 1 is 75% full The Solution guide (page 408) has the following solution. mls qos queue-set output *2* threshold 1 *75 100 100 100* -- queset is preconfigured on the port to 2 mls qos srr-queue output cos-map queue 1 threshold *3* 5 .. My interpretation was to move the Cos 5 into Q1t1 but the command says threshold 3 .. is this just a typo or am I missing something obvious. Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] lan qos- cos 5 threshold 1 in queue 1 doesn't show in running config
Hi Krishna, you will not see Cos 5 in Threshold 1 because this is the default Threshold for Cos 5. The running-config presents always only the non-default values, e.g if you put an prefererence command on your dial-peers you will this only as long as it is not the default value 0. Regards Steffen 2012/10/14 Krishna vinayak_...@yahoo.com hi guys, i was wondering whether i am doing right way of doing lan qos or not ?? the requirements are assign cos 5 to priority queue , cos 3 4 to queue 2 with 60% exceed of cos 4 should be dropped. so here is my configuration for that mls qos mls qos srr-queue output cos-map queue 1 threshold 1 5 mls qos srr-queue output cos-map queue 2 threshold 2 3 mls qos srr-queue output cos-map queue 2 threshold 1 4 mls qos queue-set output 2 threshold 3 60 100 100 272 when i issued show run | i mls commands, i see every mls qos command except the cos 5 which is assigned to q1 t1. Is my approach is correct in dealing this question correctly?? does it matter whether we assign cos values to t1 or t2 or t3 in the queues??? your input is much appreciated. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME Multicast MOH Port number
It's from multicast moh section of the cmeadmin guide, page 1252 of the current guide version. I think that has nothing to do with live-feed in this case. 2012/10/4 Dimuthu dim...@yahoo.com For me it seems document is correct (no typos). We can configure different moh live-feeds with unique media port numbers using voice moh-group number command. Document Author seems referring to the live-feed with media port 2000 inside telephony-service configuration. *From:* Jason Aarons (AM) jason.aar...@dimensiondata.com *To:* Steffen Bruening stbruen...@gmail.com; ccie_voice ccie_voice@onlinestudylist.com *Sent:* Wednesday, October 3, 2012 8:59 PM *Subject:* Re: [OSL | CCIE_Voice] CME Multicast MOH Port number I think that is a typo. Stick with 16384. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Steffen Bruening *Sent:* Tuesday, October 02, 2012 5:28 PM *To:* ccie_voice *Subject:* [OSL | CCIE_Voice] CME Multicast MOH Port number Hi all, I know that usually we would configure multicast moh for CME in this way multicast' moh 239.1.1.1 port 16384 route . I read through the CME admin guide and found the following sentence: *port port-number—Media port for multicast. Range is 2000 to 65535. We recommend port 2000 because it is already used for normal RTP media transmissions between IP phones and the router.* I use hardware vpn with Proctorlabs so I can't test multicast moh with my phones at home. Did somebody uses Port 2000 for multicast moh? Regards Steffen itevomcid ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] DSP Allocation
Hi all, I have a fractional PRI with MGCP (8 channels). When I look int sh voice dsp group all I could see that 8 channels of dsp 5 are used. Maybe someone can explain how the dsp allocation algorithm is working. I would like to know why not dsp 1 will be used. Is there maybe an option that I could decide which dsp should be used (a kind of preference)??? dsp 1: State: UP, firmware: 9.4.7 Max signal/voice channel: 8/8 Max credits: 160 num_of_sig_chnls_allocated: -2 Transcoding channels allocated: 0 Group: FLEX_GROUP_VOICE, complexity: FLEX Shared credits: 160, reserved credits: 0 Signaling channels allocated: 0 Voice channels allocated: 0 Credits used: 0 dsp 5: State: UP, firmware: 9.4.7 Max signal/voice channel: 16/16 Max credits: 240 num_of_sig_chnls_allocated: 8 Transcoding channels allocated: 0 Group: FLEX_GROUP_VOICE, complexity: FLEX Shared credits: 240, reserved credits: 0 Signaling channels allocated: 8 Voice channels allocated: 0 Credits used: 0 Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CME Multicast MOH Port number
Hi all, I know that usually we would configure multicast moh for CME in this way multicast' moh 239.1.1.1 port 16384 route . I read through the CME admin guide and found the following sentence: *port port-number—Media port for multicast. Range is 2000 to 65535. We recommend port 2000 because it is already used for normal RTP media transmissions between IP phones and the router.* I use hardware vpn with Proctorlabs so I can't test multicast moh with my phones at home. Did somebody uses Port 2000 for multicast moh? Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] sip gateway PSTN disconnect cause delayed send to cisco phone
Hi all, this question is not ccie lab related. I have 2911 integrated via sip to cucm 8.6. When a call terminates because of user busy/unallocated number or what ever, the cisco phone still rings for 30 seconds because the sip disconnect message was send 30 seconds after pstn disconnect. How could I reduce this timer? *Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x2 0x1, Calling num 30210035294 *Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x0087 callID = 0x8008 switch = primary-net5 interface = User *Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8 callref = 0x0087 Bearer Capability i = 0x9090A3 Standard = CCITT Transfer Capability = 3.1kHz Audio Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA9839F Exclusive, Channel 31 Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info Calling Party Number i = 0x2181, '30210035294' Plan:ISDN, Type:National Called Party Number i = 0x81, '0302' Plan:ISDN, Type:Unknown *Aug 30 09:50:55.631: //84/45469900/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac From: sip:8000294@10.0.242.58 ;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666 To: sip:00302@10.0.242.61 Date: Thu, 30 Aug 2012 09:50:55 GMT Call-ID: 45469900-3f137a3-1c0-3af2000a@10.0.242.58 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-15.2.4.M1 Content-Length: 0 *Aug 30 09:50:56.495: ISDN Se0/0/0:15 Q931: RX - SETUP_ACK pd = 8 callref = 0x8087 Channel ID i = 0xA9839F Exclusive, Channel 31 *Aug 30 09:50:57.135: ISDN Se0/0/0:15 Q931: RX - DISCONNECT pd = 8 callref = 0x8087 Cause i = 0x8283 - No route to destination Facility i = 0x91A10802011F0201228100 Protocol Profile = Remote Operations Protocol 0xA10802011F0201228100 Component = Invoke component Invoke Id = 31 Operation = AOCDChargingUnit Progress Ind i = 0x8288 - In-band info or appropriate now available *Aug 30 09:50:57.135: ISDN Se0/0/0:15 Q931: call_disc: PI received in disconnect; Postpone sending RELEASE for callid 0x8008 *Aug 30 09:50:57.139: //84/45469900/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac From: sip:8000294@10.0.242.58 ;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666 To: sip:00302@10.0.242.61;tag=CF900-858 Date: Thu, 30 Aug 2012 09:50:55 GMT Call-ID: 45469900-3f137a3-1c0-3af2000a@10.0.242.58 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: sip:0302@10.0.242.61 ;party=called;screen=no;privacy=off Contact: sip:00302@10.0.242.61:5060;transport=tcp Supported: sdp-anat Reason: Q.850;cause=3 Server: Cisco-SIPGateway/IOS-15.2.4.M1 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 7858 2021 IN IP4 10.0.242.61 s=SIP Call c=IN IP4 10.0.242.61 t=0 0 m=audio 16398 RTP/AVP 0 8 101 c=IN IP4 10.0.242.61 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 *Aug 30 09:51:37.203: ISDN Se0/0/0:15 Q931: RX - RELEASE pd = 8 callref = 0x8087 Cause i = 0x8283 - No route to destination Facility i = 0x91A10802011F0201228100 Protocol Profile = Remote Operations Protocol 0xA10802011F0201228100 Component = Invoke component Invoke Id = 31 Operation = AOCDChargingUnit *Aug 30 09:51:37.203: ISDN Se0/0/0:15 Q931: TX - RELEASE_COMP pd = 8 callref = 0x0087 *Aug 30 09:51:37.227: //84/45469900/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 404 Not Found Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac From: sip:8000294@10.0.242.58 ;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666 To: sip:00302@10.0.242.61;tag=CF900-858 Date: Thu, 30 Aug 2012 09:50:55 GMT Call-ID: 45469900-3f137a3-1c0-3af2000a@10.0.242.58 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-15.2.4.M1 Reason: Q.850;cause=3 Content-Length: 0 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] sip gateway PSTN disconnect cause delayed send to cisco phone
Hi Krishna, I fixed it with ios software mtp and mtp required on the sip trunk. 2012/8/30 Krishna vinayak_...@yahoo.com steffen, do you have other gw listed in the route list??? 404 sip response tries to hunt next gw if it is listed in route list that may causing this issue... thank you Krishna. -- *From:* Steffen Bruening stbruen...@gmail.com *To:* ccie_voice ccie_voice@onlinestudylist.com *Sent:* Thursday, August 30, 2012 4:56 AM *Subject:* [OSL | CCIE_Voice] sip gateway PSTN disconnect cause delayed send to cisco phone Hi all, this question is not ccie lab related. I have 2911 integrated via sip to cucm 8.6. When a call terminates because of user busy/unallocated number or what ever, the cisco phone still rings for 30 seconds because the sip disconnect message was send 30 seconds after pstn disconnect. How could I reduce this timer? *Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x2 0x1, Calling num 30210035294 *Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x0087 callID = 0x8008 switch = primary-net5 interface = User *Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8 callref = 0x0087 Bearer Capability i = 0x9090A3 Standard = CCITT Transfer Capability = 3.1kHz Audio Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA9839F Exclusive, Channel 31 Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info Calling Party Number i = 0x2181, '30210035294' Plan:ISDN, Type:National Called Party Number i = 0x81, '0302' Plan:ISDN, Type:Unknown *Aug 30 09:50:55.631: //84/45469900/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac From: sip:8000294@10.0.242.58 ;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666 To: sip:00302@10.0.242.61 Date: Thu, 30 Aug 2012 09:50:55 GMT Call-ID: 45469900-3f137a3-1c0-3af2000a@10.0.242.58 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-15.2.4.M1 Content-Length: 0 *Aug 30 09:50:56.495: ISDN Se0/0/0:15 Q931: RX - SETUP_ACK pd = 8 callref = 0x8087 Channel ID i = 0xA9839F Exclusive, Channel 31 *Aug 30 09:50:57.135: ISDN Se0/0/0:15 Q931: RX - DISCONNECT pd = 8 callref = 0x8087 Cause i = 0x8283 - No route to destination Facility i = 0x91A10802011F0201228100 Protocol Profile = Remote Operations Protocol 0xA10802011F0201228100 Component = Invoke component Invoke Id = 31 Operation = AOCDChargingUnit Progress Ind i = 0x8288 - In-band info or appropriate now available *Aug 30 09:50:57.135: ISDN Se0/0/0:15 Q931: call_disc: PI received in disconnect; Postpone sending RELEASE for callid 0x8008 *Aug 30 09:50:57.139: //84/45469900/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac From: sip:8000294@10.0.242.58 ;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666 To: sip:00302@10.0.242.61;tag=CF900-858 Date: Thu, 30 Aug 2012 09:50:55 GMT Call-ID: 45469900-3f137a3-1c0-3af2000a@10.0.242.58 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: sip:0302@10.0.242.61 ;party=called;screen=no;privacy=off Contact: sip:00302@10.0.242.61:5060;transport=tcp Supported: sdp-anat Reason: Q.850;cause=3 Server: Cisco-SIPGateway/IOS-15.2.4.M1 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 256 v=0 o=CiscoSystemsSIP-GW-UserAgent 7858 2021 IN IP4 10.0.242.61 s=SIP Call c=IN IP4 10.0.242.61 t=0 0 m=audio 16398 RTP/AVP 0 8 101 c=IN IP4 10.0.242.61 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 *Aug 30 09:51:37.203: ISDN Se0/0/0:15 Q931: RX - RELEASE pd = 8 callref = 0x8087 Cause i = 0x8283 - No route to destination Facility i = 0x91A10802011F0201228100 Protocol Profile = Remote Operations Protocol 0xA10802011F0201228100 Component = Invoke component Invoke Id = 31 Operation = AOCDChargingUnit *Aug 30 09:51:37.203: ISDN Se0/0/0:15 Q931: TX - RELEASE_COMP pd = 8 callref = 0x0087 *Aug 30 09:51:37.227: //84/45469900/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 404 Not Found Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac From: sip:8000294@10.0.242.58 ;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666 To: sip:00302@10.0.242.61;tag=CF900-858
Re: [OSL | CCIE_Voice] five lab workbook CUCM-DHCP issue
Hi, I don't know whether you are using your own equipment or remote proctorlabs. You should check your show ip route output. Also you can try to restart the dhcp monitor service or to disable the cisco security agent with utils csa disable (server reboot required). 2012/8/29 Krishna vinayak_...@yahoo.com hi folks... So far i have done 6 labs practicing ipexpert five lab workbook, and everytime i encountered the CUCM dhcp issue where site A and Site B phone are unable to register due to dhcp issue... on site A router A, and Site B router interface's had ip helper-address 10.10.210.10.. when i debup ip packet for acl that includes udp ports 67 68.. here is the message i found and tried google it but no luck in finding the solution... here is the debug that looks like.. FIBipv4-packet-proc: packet routing failed Note: the dhcp configured local to router's work fine with no issues... thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] call routing - There isn't just one way of doing it
Hi all, when ever I go through the call routing section I find several ways to answer the question, some straight, some inconvient. But I don't know for what I can get the points. Does the way matter to reach the goal/answer the question (as long as it not vialote the question itself)? Or is irrelevant as long as the phones ringing as they should and the isdn debug shows correct plan and type, channel and calling name? Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] iDivert/DND in SRST
Hi Ramy, does the question say that you should preserve this? There is no iDivert softey in CME/SRST 7.X. iDivert is supported for SIP Phones in CME with 8.5 or so but not for phones falling back to SRST. You can only preserve the feature with workarounds like this: Transfer a call directly into cue mailbox http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_tech_note09186a00802ab979.shtml Regards Steffen 2012/8/7 Ramy Abdelrahim ramyoth...@hotmail.com Dear All, When the phone is registered to UCM it has iDivert softkey button to transfer a call to VM while ringing. When this site goes into SRST, iDivert is not there. Do I have to preserve this feature in SRST? And if it's the case then how? Can anyone help on this? Regards, Ramy ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] new lab 3, 5.1 media resources
Hi all, in section 5.1 is written Configure a transcoder on Site A gateway...Site A devices *must have the privileges* to use the transcoder when necessary. Maybe somebody can help me to get a better understanding on this. Does this mean the Site A is the only site which should allow to use it, or could also other site use it? How would you interpret this request? Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Where all the phones gone?
Hi James, the ports where the phones should be connected are up and I assume the won't be up when no phone/device is attached to them. CDP is enabled but it doesn't show cdp information about the devices which are connected to these ports. Power Inline Auto is the default power mode and have never seen any explicit power command on the 3750 or switch-modules in the routers which could affect the power status of these ports. Regards Steffen 2012/7/13 James Dull jd...@ipexpert.com Steffen, Specifically for BR2-RTR and HQ-3750 the ports may need to have power inline auto applied, shut for about 30 seconds to 1 minute and no shut. BR1-RTR should never need this because it is using a power daughter card which always has power applied to the ports. For the 7962's they are located remotely and I do not directly control which pods they are assigned to but assume that there are not enough to have on every pod at the same time so they can be moved by the voice instructor according to where he needs them. There should always be phones on the devices though as the 7960s are in house and we do not remove those. I ask that you try what I recommended to get the phones to show up, they should immediately show up in show cdp neigh once power inline auto has been applied. James Dull - CCNA, Comp TIA A+, Network+, Security+ Technical Support/Technical Editor - IPexpert, Inc., Masonic (MIT), Inc. URL:http://www.IPexpert.com/Tell-Me-Morehttp://www.ipexpert.com/Tell-Me-More Phone: +1.810.326.1444 ext 206 Email: jd...@ipexpert.com Twitter: www.Twitter.com/IPexpert Check out OUR CATALOG: http://www.ipexpert.com/catalog On Jul 13, 2012, at 1:47 AM, Steffen Bruening stbruen...@gmail.com wrote: Hi all, during all my last 4 proctorlab sessions I missed phones connected to the switch/switch-modules. Often only one phone per site is available, sometimes a complete site has none. It is really really annoying to find a solution about how to bring enough phones (own physical devices or soft clients) to the pod. How are your experiences with that? I am using always the sessions at 12:00 AM PST. Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com image003.jpg___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Where all the phones gone?
Yes, the first three commands I put in on every device are, cdp run, cdp ad and cdp t 5. 2012/7/16 Ronmac ron...@solcon.nl Hi, Maybe stupid remark but, Is cdp enabled. AdverV2! V1 os not ok Regards ron Send from my mobile Op 16 jul. 2012 om 12:41 heeft Steffen Bruening stbruen...@gmail.com het volgende geschreven: Hi James, the ports where the phones should be connected are up and I assume the won't be up when no phone/device is attached to them. CDP is enabled but it doesn't show cdp information about the devices which are connected to these ports. Power Inline Auto is the default power mode and have never seen any explicit power command on the 3750 or switch-modules in the routers which could affect the power status of these ports. Regards Steffen 2012/7/13 James Dull jd...@ipexpert.com Steffen, Specifically for BR2-RTR and HQ-3750 the ports may need to have power inline auto applied, shut for about 30 seconds to 1 minute and no shut. BR1-RTR should never need this because it is using a power daughter card which always has power applied to the ports. For the 7962's they are located remotely and I do not directly control which pods they are assigned to but assume that there are not enough to have on every pod at the same time so they can be moved by the voice instructor according to where he needs them. There should always be phones on the devices though as the 7960s are in house and we do not remove those. I ask that you try what I recommended to get the phones to show up, they should immediately show up in show cdp neigh once power inline auto has been applied. James Dull - CCNA, Comp TIA A+, Network+, Security+ Technical Support/Technical Editor - IPexpert, Inc., Masonic (MIT), Inc. URL:http://www.IPexpert.com/Tell-Me-Morehttp://www.ipexpert.com/Tell-Me-More Phone: +1.810.326.1444 ext 206 Email: jd...@ipexpert.com Twitter: www.Twitter.com/IPexpert Check out OUR CATALOG: http://www.ipexpert.com/catalog image003.jpg On Jul 13, 2012, at 1:47 AM, Steffen Bruening stbruen...@gmail.com wrote: Hi all, during all my last 4 proctorlab sessions I missed phones connected to the switch/switch-modules. Often only one phone per site is available, sometimes a complete site has none. It is really really annoying to find a solution about how to bring enough phones (own physical devices or soft clients) to the pod. How are your experiences with that? I am using always the sessions at 12:00 AM PST. Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] New Lab 2, 8.2 tracing UCMCUPS signaling, missing picture in DSG
Hi all, there is a picture missing on page 197 in the solution guide, which should show how to setup the trace settings. Can somebody explain me how I should setup the trace? Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Where all the phones gone?
Hi all, during all my last 4 proctorlab sessions I missed phones connected to the switch/switch-modules. Often only one phone per site is available, sometimes a complete site has none. It is really really annoying to find a solution about how to bring enough phones (own physical devices or soft clients) to the pod. How are your experiences with that? I am using always the sessions at 12:00 AM PST. Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Ipexpert CCIE Voice Five Lab Handbook
Hi all, I have the BLS from IPexpert and I feel very comfortable with the 10 labs of it. Now I got a marketing email for 5 extra labs made by Ipexpert to learn for the real lab. www.ipexpert.com/cisco/ccie/voice/handbook Does somebody of you buyed this product? Is it it worth? What are the differents compared to the 10 old labs? Are they harder/more difficult? Regards Steffen ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Calling Number during Unity Connection Call Transfer to external number
I found it, CUCM Service Parameter: Display Original Calling Number on Transfer from Cisco Unity – Value needs to be set to true. Von: Cristobal Priego [mailto:cristobalpri...@gmail.com] Gesendet: Donnerstag, 15. September 2011 03:55 An: Steffen Bruening Cc: ccie_voice@onlinestudylist.com Betreff: Re: [OSL | CCIE_Voice] Calling Number during Unity Connection Call Transfer to external number I there is a service parameter that you have to chante to preserve the caller id. I don't remember the name of the option though Hth Enviado desde mi iPhone El Sep 14, 2011, a las 14:21, Steffen Bruening steffen.bruen...@intact-is.com escribió: Hi all, this question is not related to the CCIE labs, but hopefully you can help me to find an answer. When a caller reach a user voicemail and choose the caller input option to be forwarded to an alternate number (in this case the mobile number) I can see in the isdn debug that extension of the used voicemail-port for this call transfer is used as calling number instead of the originating number. Any ideas how I could achieve that the originating phone number will be presented? The german telco do not check or restrict the calling number presentation (CLIP no screening). Regards Steffen __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Calling Number during Unity Connection Call Transfer to external number
Hi all, this question is not related to the CCIE labs, but hopefully you can help me to find an answer. When a caller reach a user voicemail and choose the caller input option to be forwarded to an alternate number (in this case the mobile number) I can see in the isdn debug that extension of the used voicemail-port for this call transfer is used as calling number instead of the originating number. Any ideas how I could achieve that the originating phone number will be presented? The german telco do not check or restrict the calling number presentation (CLIP no screening). Regards Steffen __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email _ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] mixture of own and remote hardware for vRack session
Hi all, I have made two vRack sessions so far based on Remote Phones and Softphones. For my next session I want to use the hardware vpn session. I have a 2811 with 4-Port Hwic and 3 7962 phones, but as written on proctorlabs it seems that this is not enough (requirements are 5 phones and a 8-port switch) . Isn`t it possible/allowed to use a mixture of my own phones and some of the remote phones? Regards Steffen __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email _ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
