Re: [OSL | CCIE_Voice] Finally i got it! CCIE # 22885
Great to see another person pass. All the best. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Arevalo Sent: 05 December 2008 12:49 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Finally i got it! CCIE # 22885 After almost one year of hard study and more than one attempt, i finally got it!!! My brand new IE # is 22885 ! First of all many many thanks to my wife for her amazing support, and to my little two-months daugther who inspired me, she made me the things a little bit more difficult, because you know how the newborns are i'm happy that she is here. I really want to thank you all for this amazing group study, for all the posts and interesting discussions and opinions, thanks to Vik for the excellent bootcamp (twice) and thanks to Mark and Vik for clarifying our doubts when they got the answers or at least tips. Now it's time to take a nap then go for some specializations or maybe another IE, who knows... Brgds//r.a.
Re: [OSL | CCIE_Voice] Badly got stuck in one way ccm-gk-cme scenario , please help
Cause code looks like unallocated number. Can't be 100% with referring to a
spec. But the Q931CauseIe IEData= 08 02 80 81 -- 81 is same as 01 =
unallocated.
Do you have an xlate on ccm to strip the 5# and resubmit for lookup on
internal extensions?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jeremy co
Sent: 04 December 2008 16:15
To: CCIE Voice Maillist
Subject: [OSL | CCIE_Voice] Badly got stuck in one way ccm-gk-cme scenario
,please help
Hi,
consider this scenario:
(7001,7002 phones )ccm --trunk---GK-cme (1001,1002 phones)
I can call from ccm to cme but not reverse.
anybody have any idea why I cannot call form cme to ccm? from trace I can
see call reaches ccm, but I hear busy signal.
I waste 2 days to solve this but no progress
trace on ccm when calling from cme to ccm :
## CCM # H.225 Trunk (Gatekeeper Controlled) ,zone=voice prefix
=5#
Cisco CallManagervalue H323-UserInformation ::=
Cisco CallManager{* h323-uu-pdu * {* h323-message-body releaseComplete
: * {* protocolIdentifier { 0 0 8 2250 0 2 },* callIdentifier * {* guid
'1252F44D2D6C11D680B5B1728157865B'H* }* },* h245Tunneling FALSE* }
Cisco CallManager}
TraceCisco CallManager
TraceCisco CallManagerOut Message -- H225ReleaseCompleteMsg --
Protocol= H225Protocol
TraceCisco CallManagerIe - Q931CauseIe IEData= 08 02 80 81
TraceCisco CallManagerIe - H225UserUserIe IEData= 7E 00 21 05 25 80 06
00 08 91 4A 00 02 01 11 00 12 52 F4 4D 2D 6C 11 D6 80 B5 B1 72 81 57 86 5B
0A 80 01 00
TraceCisco CallManagerMMan_Id= 0. (iep= 0 dsl= 0 sapi= 0 ces= 0
IpAddr=fe42688e IpPort=36064)
TraceCisco CallManagerIsdnMsgData2= 08 02 80 43 5A 08 02 80 81 7E 00 21
05 25 80 06 00 08 91 4A 00 02 01 11 00 12 52 F4 4D 2D 6C 11 D6 80 B5 B1 72
81 57 86 5B 0A 80 01 00
TraceCisco CallManager
TraceCisco CallManagerIn Message -- H225ReleaseCompleteMsg -- Protocol=
H225Protocol
TraceCisco CallManagerIe - Q931CauseIe -- IEData= 08 02 80 81
TraceCisco CallManagerIe - H225UserUserIe -- IEData= 7E 00 22 05 25 80
06 00 08 91 4A 00 04 11 00 11 00 12 52 F4 4D 2D 6C 11 D6 80 B5 B1 72 81 57
86 5B 10 80 01 80
TraceCisco CallManagerMMan_Id= 0. (iep= 0 dsl= 0 sapi= 0 ces= 0
IpAddr=fe42688e IpPort=0)
TraceCisco CallManagerIsdnMsgData1= 08 02 00 43 5A 08 02 80 81 7E 00 22
05 25 80 06 00 08 91 4A 00 04 11 00 11 00 12 52 F4 4D 2D 6C 11 D6 80 B5 B1
72 81 57 86 5B 10 80 01 80
Cisco CallManagervalue H323-UserInformation ::=
Cisco CallManager{* h323-uu-pdu * {* h323-message-body releaseComplete : *
{* protocolIdentifier { 0 0 8 2250 0 4 },* callIdentifier * {* guid
'1252F44D2D6C11D680B5B1728157865B'H* }* },* h245Tunneling TRUE* }
###CME#
translation-rule 1
Rule 1 77 7
Rule 2 75 5
!
dial-peer voice 100 voip
destination-pattern 7
translate-outgoing called 1
voice-class codec 1
session target ras
tech-prefix 5#
dtmf-relay h245-alphanumeric
## GK #
gatekeeper
zone local ZONE-RS2 cisco.com 114.0.0.254
zone local voice cisco.com
zone prefix voice 5...
zone prefix voice 7...
no shutdown
!
debug gatekeeper main 10
Mar 2 23:43:15.237: gk_process: QUEUE_EVENT (minor 0) wakeup
Mar 2 23:43:15.241: gk_rassrv_arq: arqp=0x844FAA80, crv=0x44, answerCall=0
Mar 2 23:43:15.241: gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC
Mar 2 23:43:15.241: gk_dns_query: No Name servers
Mar 2 23:43:15.241: rassrv_get_addrinfo: (5#7003) Matched tech-prefix 5#
Mar 2 23:43:15.241: rassrv_get_addrinfo: (5#7003) Matched zone prefix 7 and
remainder 003
Mar 2 23:43:15.245: gk_rassrv_get_ingress_network: ARQ non-std ingress
network = 1
Mar 2 23:43:15.245: rassrv_arq_select_viazone: about to check the source
side, src_zonep=0x85EEA3C4
Mar 2 23:43:15.245: rassrv_arq_select_viazone: matched zone is ZONE-RS2,
and z_invianamelen=0
Mar 2 23:43:15.245: rassrv_arq_select_viazone: about to check the
destination side, dst_zonep=0x85EEA670
Mar 2 23:43:15.245: rassrv_arq_select_viazone: matched zone is voice, and
z_outvianamelen=0
Mar 2 23:43:15.245: gk_rassrv_get_ingress_network: ARQ non-std ingress
network = 1
M
C2600#ar 2 23:43:15.245: gk_zone_get_proxy_usage: local zone= voice, remote
zone= ZONE-RS2, call direction= 0, eptype= 67586 be_entry= 0
Mar 2 23:43:15.245: gk_zone_get_proxy_usage: returns proxied = 0
Mar 2 23:43:15.245: gk_gw_select_px: Source and destination endpoints in
different local zones
Mar 2 23:43:15.245: gk_zone_get_proxy_usage: local zone= ZONE-RS2, remote
zone= voice, call direction= 1, eptype= 67586 be_entry= 0
Mar 2 23:43:15.249: gk_zone_get_proxy_usage: returns proxied = 0
Mar 2 23:43:15.513: gk_process: QUEUE_EVENT (minor 0) wakeup
C2600#un al
Mar 2 23:43:17.865: gk_process: QUEUE_EVENT (minor 0) wakeup
Jeremy
Re: [OSL | CCIE_Voice] Badly got stuck in one way ccm-gk-cme scenario , please help
Jeremy,
It looks as though you set a service param to make the trunk use port 1719?
Do you remember doing this?
I would perhaps check the css of the trunk and check the partition used for
each of the lines on the phones. I am sure you have done this but it never
hurts to check once more.
4 digits is certainly a fair way of doing this but if you were ever to need
TEHO it will not work.
This scenario normally works no problems.
_
From: jeremy co [mailto:[EMAIL PROTECTED]
Sent: 04 December 2008 17:05
To: Stephen Collinson
Cc: CCIE Voice Maillist; rob bourne; [EMAIL PROTECTED]
Subject: Re: [OSL | CCIE_Voice] Badly got stuck in one way ccm-gk-cme
scenario ,please help
Hi,
I set significant digits to 4 on trunk to GK. but same result fast busy
signal.
trace out put:
Cisco CallManagerSPROCRas - value V2Message ::= registrationRequest : * {*
requestSeqNum 931,* protocolIdentifier { 0 0 8 2250 0 2 },*
discoveryComplete FALSE,* callSignalAddress * {* ipAddress : * {* ip
'8E04400B'H,* port 4889* }* },* rasAddress * {* ipAddress : * {* ip
'8E04400B'H,* port 1719* }* },* terminalType * {* gateway * {* protocol * {*
h323 : * {* },* voice : * {* supportedPrefixes * {* {* prefix e164 : 5#*
}* }* }* }* },* mc FALSE,* undefinedNode FALSE* },* gatekeeperIdentifier
voice,* endpointVendor * {* vendor * {* t35CountryCode 181,* t35Extension
0,* manufacturerCode 18* }* },* timeToLive 60,* keepAlive TRUE,*
endpointIdentifier 85EEB933,*
Cisco CallManagerwillSupplyUUIEs FALSE* }
Cisco CallManagerEnvProcessUdpPort - EnvProcessUdpHandler::fireSignal()
varId = 0
Cisco CallManagerEnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 99,
114.0.0.254:1719)
Cisco CallManagerEnvProcessUdpHandler::handle_input Status: 0, Id: 0
Cisco CallManagervalue V2Message ::= registrationConfirm : * {*
requestSeqNum 931,* protocolIdentifier { 0 0 8 2250 0 4 },*
callSignalAddress * {* },* gatekeeperIdentifier voice,* endpointIdentifier
85EEB933,* timeToLive 60,* willRespondToIRR FALSE* }
Cisco CallManagerGKIFHandler: 114.0.0.254 processRCFInd seqNum=931
Cisco CallManagerCMProcMon - --Entered Router Verification
Cisco CallManagerCMProcMon - Exited Router Verification
Cisco CallManagerCMProcMon - --Entered Router Verification
Cisco CallManagerCMProcMon - Exited Router Verification
Cisco CallManagerCMProcMon - --Entered Router Verification
Cisco CallManagerCMProcMon - Exited Router Verification
C2600#sh gatekee gw-type-prefix
GATEWAY TYPE PREFIX TABLE
=
Prefix: 5#*
Zone voice master gateway list:
100.0.0.254:4889 GK-2600-trunk_1
##
C2600#sh gatekeeper end
GATEKEEPER ENDPOINT REGISTRATION
CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags
--- - --- - - -
100.0.0.254 4889 100.0.0.254 1719 voice VOIP-GW
H323-ID: GK-2600-trunk_1
Voice Capacity Max.= Avail.= Current.= 0
140.0.0.254 1720 140.0.0.254 52545 ZONE-RS2 VOIP-GW
E164-ID: 1001
E164-ID: 442076301001
E164-ID: 1002
E164-ID: 442076301002
E164-ID: 1003
E164-ID: 442076301003
E164-ID: 1004
E164-ID: 442076301004
E164-ID: 1005
E164-ID: 442076301005
H323-ID: RS2
Voice Capacity Max.= Avail.= Current.= 0
Total number of active registrations = 2
#
seems call reaches ccm, but for unknown reason ,it would not processed by
ccm.
any idea?
Jeremy
On Fri, Dec 5, 2008 at 3:29 AM, Stephen Collinson
[EMAIL PROTECTED] wrote:
Cause code looks like unallocated number. Can't be 100% with referring to a
spec. But the Q931CauseIe IEData= 08 02 80 81 -- 81 is same as 01 =
unallocated.
Do you have an xlate on ccm to strip the 5# and resubmit for lookup on
internal extensions?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jeremy co
Sent: 04 December 2008 16:15
To: CCIE Voice Maillist
Subject: [OSL | CCIE_Voice] Badly got stuck in one way ccm-gk-cme scenario
,please help
Hi,
consider this scenario:
(7001,7002 phones )ccm --trunk---GK-cme (1001,1002 phones)
I can call from ccm to cme but not reverse.
anybody have any idea why I cannot call form cme to ccm? from trace I can
see call reaches ccm, but I hear busy signal.
I waste 2 days to solve this but no progress
trace on ccm when calling from cme to ccm :
## CCM # H.225 Trunk (Gatekeeper Controlled) ,zone=voice prefix
=5#
Cisco CallManagervalue H323-UserInformation ::=
Cisco CallManager{* h323-uu-pdu * {* h323
Re: [OSL | CCIE_Voice] UNITY - CCM DSCP markings - NO ACL
Would be good to see those registry keys. I could not find anything on Cisco which indicates there are registry keys which can be amended. I also went through the registry searching on a few things and did not get a result. I believe if you take the registry keys which TSP 8.2 adds and stick them in earlier versions they are ignored. But Vik is often correct so I guess they may be around somewhere. Cheers Steve _ From: Sergio Polizer [mailto:[EMAIL PROTECTED] Sent: 01 December 2008 21:44 To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] UNITY - CCM DSCP markings - NO ACL Steve, You are right. Vik wrote this some time ago: The QOS markings on Unity are really dependent on the version of the TSP. With Unity 4.0(5) the TSP (8.0x) hardcodes the DSCP markings to AF31 for SCCP and EF for Media. With Unity 5.x and TSP 8.2x you can change the markings in UTIM (which changes a registry setting). If you wanted to change the markings of Unity 4.0(5) traffic then the easiest way would be to remark on the switch (otherwise you have to add some registry keys) Cheers, Sergio. From: [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Date: Thu, 27 Nov 2008 18:50:48 + Subject: [OSL | CCIE_Voice] UNITY - CCM DSCP markings - NO ACL Anyone got any ideas on how to change unity DSCP markings either on the UNITY box or on the switch without using an ACL. I was wondering if setting the unity port to trut-ipprec and then use the ipprec-to-dscp map. Do not know of anything for making the change on unity. TSP version is too low for the registry fix, I believe. Cheers Steve _ Conheça já o Windows Live Spaces, o site de relacionamentos do Messenger! Crie já o seu! http://www.amigosdomessenger.com.br
[OSL | CCIE_Voice] UNITY - CCM DSCP markings - NO ACL
Anyone got any ideas on how to change unity DSCP markings either on the UNITY box or on the switch without using an ACL. I was wondering if setting the unity port to trut-ipprec and then use the ipprec-to-dscp map. Do not know of anything for making the change on unity. TSP version is too low for the registry fix, I believe. Cheers Steve
Re: [OSL | CCIE_Voice] VMware images
Phil, Are you just starting out on this mission for voice IE? Unless you are really close to the exam level I would wait about 8 months and then start. The exam is going to change for sure mid next year, announcement due Dec / Jan / Feb time. On average it takes 3-5 times to pass this track. With the current booking window you will at most get in 2 attempts between now and mid next year. If you happen to pick Brussels, Australia or Tokyo that will be one. Since the 90 day payment came into force it has really caused issues for those of us whom are able to take slots at short notice, there are very few coming through. Cisco have updated all the material over the past couple of months and the workbooks, DVD, etc are out of date. There are way too many tricks in the exam now. This may be to try and stop a run on it in the lead up to new exam next year. The pass rate also always drops massively when new material is introduced. When the new exam is introduced it is going to take some time for new training material to catch up. I am not quite sure how the guys manage it but give them a couple of months and they will have the skinny on what's going on and the material will be good. This time will be a true test of the guarantee provided by IPExpert and others and I am sure they will come good. I had to laugh today when someone put up something about NDA. This whole bootcamp CCIE training industry lark revolves around running close to the NDA. Watch out for the likes for Internetwork Expert with their CCIE 2.0. On the face of it their Webinar looks like they will seriously challenge the NDA and will no doubt attract some attention from Cisco. 2c from a man in the joyful pursuit of further certification Steve CCIE _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phillip Day Sent: 20 November 2008 19:49 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] VMware images Hi, I have recently built a home lab and I'm trying to get hold of some VMware call manager and IPCC images for my lab, does anyone know where I might acquire them on a budget? Thanks in advance Phill __ This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This footnote also confirms that this email message has been swept by a content checking tool for the presence of computer viruses. Nettitude Limited is a Company registered in England Registered Address Nettitude Limited, Fosse House, Fosse Way, Leamington Spa, Warwickshire, CV31 1XG Company Registration Number: 4705154 VAT Number: 812 4539 44 www.nettitude.com __
Re: [OSL | CCIE_Voice] VMware images
I find a good measure of readiness is to pick the hardest questions off of this list and Voiceie.com and make sure you can talk yourself through the config without missing a beat. As for Internetwork Expert it is just an observation on their proposed revolutionary approach and I will be watching with interest how it progresses. If they can get away with (and deliver) dynamic content to continually reflect the latest exam I (and probably many others) will be considering signing up next time. Why make the journey harder than it needs to be? I hope yours is a short and enjoyable pursuit of CCIE Voice, good luck. Steve _ From: Phillip Day [mailto:[EMAIL PROTECTED] Sent: 20 November 2008 20:30 To: Stephen Collinson; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] VMware images Steve, Thanks for your insight - interesting view of Internetwork Expert. I have just recently passed the written, which I know is no real measure of readiness for the exam, but I feel that I'd like to get some lab experience out of the kit I have just spend a fortune on. I think that even if the exam does change next year it would benefit my experience and understanding if I could get hold of some call manage images now and run through some labs. I already work with the appliance version fairly frequently through work, so I don't think it will be that big a shock to me when the syllabus changes (famous last words I'm sure) Phill __ This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This footnote also confirms that this email message has been swept by a content checking tool for the presence of computer viruses. Nettitude Limited is a Company registered in England Registered Address Nettitude Limited, Fosse House, Fosse Way, Leamington Spa, Warwickshire, CV31 1XG Company Registration Number: 4705154 VAT Number: 812 4539 44 www.nettitude.com __
[OSL | CCIE_Voice] Off topic - Lab swap got Feb 09 looking for Dec 08
Hello all, Firstly apologies for off topic, to those whom are not interested. I have a date end Feb 09 and am looking for something after 10 December 08. I will take a slot anywhere, my slot is Brussels. As an added incentive if I pass this time and we swap I will throw in a good amount of uninterrupted rack time on my lab, say 2-4 weeks continuous. It has everything one needs and is better than any on line lab (my biased opinion) due to it supporting VMWare snapshots, own dial plan and ip address scheme as well as FXS at HQ, old and new style PVDMs and perhaps most importantly not being limited to 8 hours per go. This amount of time should allow you to get very ready for your next attempt. Thanks Steve
Re: [OSL | CCIE_Voice] Dollar Sign or No Dollar Sign ?
Both dial peers are POTS. The prefix is required in both cases. The '$' is a more specific match than the 'no $' Also There is no interdigit timeout with an exact match on a DP. A $ based DP can be useful if you need to have a more specific match for some slightly odd application. For example of you show the DPs for the ephones they have a $. There for if you need to do a preference which is going to beat an ephone it needs a $, from memory. I wait to get flamed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Sent: 17 November 2008 13:01 To: Mike Brooks; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Dollar Sign or No Dollar Sign ? In this particular case you posted, there is no difference except interdigit timeout is not invoked for DP with $ sign. In other situations, the difference matters: ! dial-peer voice 911 pots destination-pattern 911 port 0/0:23 forward-digts 3 # You need forward-digits/prefix/translation-profile on this DP because POTS DP strips explicitly matched digits ! dial-peer voice 912 pots destination-pattern 911$ port 0/0:23 # There is no digit-strip when this DP is matched ! Rgds Alex - Original Message - From: Mike Brooks [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Sent: Monday, November 17, 2008 11:23 AM Subject: [OSL | CCIE_Voice] Dollar Sign or No Dollar Sign ? I believe Vik has answered this before but I need to be reminded. Should we use the $ in our destination-patterns or not ? I have seen it configured both ways. What is the difference ? ! dial-peer voice 11 pots destination-pattern 91[2-9]..[2-9]..$ port 0/0:23 forward-digits 11 ! or dial-peer voice 11 pots destination-pattern 91[2-9]..[2-9].. port 0/0:23 forward-digits 11 Thanks, Mike Brooks CCIE#16027 (RS)
Re: [OSL | CCIE_Voice] Dollar Sign or No Dollar Sign ?
I did it before I posted. We must have something different going on. Would you like an output of my debug isdn q931 and the accompanying config? Steve -Original Message- From: Alex [mailto:[EMAIL PROTECTED] Sent: 17 November 2008 17:07 To: Stephen Collinson; 'Mike Brooks'; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Dollar Sign or No Dollar Sign ? - Original Message - From: Stephen Collinson [EMAIL PROTECTED] Both dial peers are POTS. The prefix is required in both cases. Stephen, Have you tested this? I suggest you try to configure two POTS DP with destination-pattern 911 and 911$ and without any forward-digits/prefix or translation-profile additives. The DP_911$ will complete the call wherease DP_911 won't. I personally had it configured many times on PL racks, it always works this way for me. FYI, DP with destination-pattern ^911 also does not require a prefix/forwardi-digits/translation-profile. I guess this happens because IOS cannot strip ^ and/or $ from the digit string so it leaves the matched string intact. Rgds Alex
Re: [OSL | CCIE_Voice] Dollar Sign or No Dollar Sign ?
Alex, I should add, I have been impressed with the quality of your posts and am sure this is what is happening for you. I particularly like the one with different locations for the MOH servers, could be very useful, not something I had though of as yet. Steve -Original Message- From: Alex [mailto:[EMAIL PROTECTED] Sent: 17 November 2008 17:07 To: Stephen Collinson; 'Mike Brooks'; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Dollar Sign or No Dollar Sign ? - Original Message - From: Stephen Collinson [EMAIL PROTECTED] Both dial peers are POTS. The prefix is required in both cases. Stephen, Have you tested this? I suggest you try to configure two POTS DP with destination-pattern 911 and 911$ and without any forward-digits/prefix or translation-profile additives. The DP_911$ will complete the call wherease DP_911 won't. I personally had it configured many times on PL racks, it always works this way for me. FYI, DP with destination-pattern ^911 also does not require a prefix/forwardi-digits/translation-profile. I guess this happens because IOS cannot strip ^ and/or $ from the digit string so it leaves the matched string intact. Rgds Alex
Re: [OSL | CCIE_Voice] max session 0 - IOS Conf Bridge (failed)
Guys, Not to labor a point but I have set this up this morning and do not get the positive results you get. I have 2821 with pvdm2-32 + nm-hdv with 2 * pvdm-12. Upgraded IOS to something which support CME conference (12.4.22T) Run a few tests and as I say don't get quite the same results as you guys. Let me know if you have some different config ideas to get this working. My idea of lowering the number of channels allocated to the pri-group does not work either, due to it only being possible to have one type of DSP in the DSPFARM. Regards Steve PVDM Slot 0: 32-channel (G.711) Voice/Fax PVDMII DSP SIMM PVDM daughter card Slot 1: High Density Voice Port adapter HDV SIMMs: Product (FRU) Number: PVDM-12= SIMM slot 0: PVDM-12 SIMM present. SIMM slot 1: PVDM-12 SIMM present. SIMM slot 2: Empty. SIMM slot 3: Empty. Voice-card config voice-card 0 - PVDM2 dsp services dspfarm ! voice-card 1 - NM-HDV ! Need to enable dspfarm for voice-card 1 R3(config-voicecard)#dspfarm R3(config-voicecard)#dsp ser R3(config-voicecard)#dsp services dsp R3(config-voicecard)#dsp services dspfarm dspfarm is configured for NM-HDV2 card. Only one dspfarm type is allowed. This seems to indicate the DSPFARM can only be made up of either old or new DSPs. Test set 1 background With the testing below NO pri-group has been set up. So all PVDM12 DSPs are available for xcoder ! Config now looks like this voice-card 0 dsp services dspfarm ! voice-card 1 dspfarm ! Rest of config as follows sccp local GigabitEthernet0/0.102 sccp ccm xx.xx.xx.xx identifier 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register mtp00190665bf38 associate profile 2 register cfb00190665bf38 ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 12 associate application SCCP shutdown ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 4 simulate scenario using a PVDM2-16 - use all DSP associate application SCCP ! Test 1 Try to enable dspfarm profile 1 when both 5510 DSP allocated to CFB R3(config-voicecard)#dspfarm profile 1 transcode R3(config-dspfarm-profile)#no shut Enabling profile failed due to insufficient TRANSCODING resources, resources available to support 0 sessions; please modify the configuration and retry R3(config)#dspfarm profile 1 transcode R3(config-dspfarm-profile)#maximum sessions ? 0-0 Number of sessions assigned to this profile Test 2 Try to enable dspfarm profile 1 when 1.5 * 5510 DSP allocated to CFB dspfarm profile 2 conference maximum sessions 3 R3(config-dspfarm-profile)#dspfarm profile 1 transcode R3(config-dspfarm-profile)#max R3(config-dspfarm-profile)#maximum s R3(config-dspfarm-profile)#maximum sessions ? 0-0 Number of sessions assigned to this profile Test 2 Try to enable dspfarm profile 1 when 1 * 5510 DSP allocated to CFB dspfarm profile 2 conference maximum sessions 2-- R3(config)#dspfarm profile 1 R3(config-dspfarm-profile)#max R3(config-dspfarm-profile)#maximum s R3(config-dspfarm-profile)#maximum sessions ? 1-6 Number of sessions assigned to this profile R3(config-dspfarm-profile)#maximum sessions 6 R3(config-dspfarm-profile)#no shut R3(config-dspfarm-profile)# *Nov 16 08:54:56.819: %SDSPFARM-6-REGISTER_NEW: mtp-4:mtp00190665bf38 Works as expected. There is now a complete free DSP which can be assigned to xcoding. Test set 2 background Another set of tests carried out after changing the voice-card config so the DSPFARM uses the PVDM12 DSPs A similar set of results whereby once all the resources had been allocated to conferencing they were not available for xcoding. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erwan Erwan Sent: 16 November 2008 04:49 To: Kevin Porter; [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] max session 0 - IOS Conf Bridge (failed) Thks Kevin and Jacobs, I have not tried it yet, but I believe must work w/ your workaround,cause I remember it worked before w/ the same amount of DSP, and Kevin's explanation really make sense why it failed --- On Sun, 11/16/08, Kevin Porter [EMAIL PROTECTED] wrote: From: Kevin Porter [EMAIL PROTECTED] Subject: RE: [OSL | CCIE_Voice] max session 0 - IOS Conf Bridge (failed) To: Jacob Owen [EMAIL PROTECTED], [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Date: Sunday, November 16, 2008, 3:17 AM You must have a free DSP to configure any sessions for conferencing. With a PVDM2-16 you will be able to configure 2 conferencing
Re: [OSL | CCIE_Voice] max session 0 - IOS Conf Bridge (failed)
PVDM 16 - Single DSP supporting 6 high complexity xcoder or 2 * 8 party conference. It is my understanding a single 5510 DSP can not support both conference and trancoder. Check the following reference http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/interop/intcnf2.html#wp100328 7 I would hazard a guess you have a T1 which is using your 2 * PVDM12, is that correct? You could reduce the number of B chans in use on the PRI, if this is the case. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erwan Erwan Sent: 15 November 2008 15:43 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] max session 0 - IOS Conf Bridge (failed) hi, I try to config IOS Conf Bridge and Transcode in 2811 , DSP type C5110 - Config for Transcode succeed - But config for Conference failed, it said max session 0, i think I still hv enough DSP - voice-card 0 dspfarm dsp services dspfarm sccp ccm group 1 associate ccm 2 priority 2 associate ccm 1 priority 1 associate profile 1 register cfb001e7a5f9530 associate profile 2 register mtp001e7a5f9530 ! dspfarm profile 2 transcode codec g711ulaw codec g729r8 maximum sessions 2 associate application SCCP ! dspfarm profile 1 conference codec g711ulaw codec g729r8 associate application SCCP shutdown --- NAME: PVDMII DSP SIMM with one DSP, DESCR: PVDMII DSP SIMM with one DSP PID: PVDM2-16 , VID: V01 , SN: FOC0D7C NAME: High Density Voice, DESCR: High Density Voice PID: NM-HDV= , VID: 1.1, SN: JAB05400BRG NAME: PVDM 3-C549 Simm, DESCR: PVDM 3-C549 Simm PID: PVDM-12= , VID: 1.1, SN: NAME: PVDM 3-C549 Simm, DESCR: PVDM 3-C549 Simm PID: PVDM-12= , VID: 1.1, SN: Thks
Re: [OSL | CCIE_Voice] max session 0 - IOS Conf Bridge (failed)
Kevin, Are you sure about the xcoder with this type of secondary DSP? It is the old style of DSP on the NM-HDV, I believe. The 2 conferences will take the 5510. If it is a T1 and all Bchans are allocated both the additional PVDM 12 will be used in the NM-HDV. I find with the older DSP it does not share them. Once they are allocated for the controller they do not seems to be available in the dsp prof max sessions. Perhaps you know a way to configure them I have not come across yet? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Porter Sent: 15 November 2008 19:18 To: Jacob Owen; [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] max session 0 - IOS Conf Bridge (failed) You must have a free DSP to configure any sessions for conferencing. With a PVDM2-16 you will be able to configure 2 conferencing sessions as long as nothing else has grabbed any of the channels. Transcoding will share DSP's with Voice Port terminations, so I agree that it is best to configure the conferencing first, which will reserve the DSP channels first, then go to transcoding. Just for info, conferencing profiles grab 8 channels of a PVDM2 per session, so a PVDM2-8 will give you a 1 for Max sessions, a PVDM2-16 will give you a 2, PVDM2-32 gives you 4, so on and so forth, but the DSP must be free of any other channels being allocated. Kevin Kevin Porter Systems Engineer L4 Netelligent Corporation 400 South Woods Mill Drive, Suite 105 St. Louis, MO 63017 Office: (314) 392-6921 Cell: (314) 852-1252 Fax: (314) 392-9760 [EMAIL PROTECTED] www.netelligent.com Bridging The Gap Between Good and GREAT IP Communications! _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Owen Sent: Saturday, November 15, 2008 11:25 AM To: [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] max session 0 - IOS Conf Bridge (failed) You can also try to turn up your conference dspfarm profile first, and then the transcoder. I have run into the same issue when trying to do the transcoder first, so just shut on dspfarm profile 2, then no shut on dspfarm profile 1, and finally no shut on dspfarm profile 2. Let me know if that helps. On Sat, Nov 15, 2008 at 10:42 AM, Erwan Erwan [EMAIL PROTECTED] wrote: hi, I try to config IOS Conf Bridge and Transcode in 2811 , DSP type C5110 - Config for Transcode succeed - But config for Conference failed, it said max session 0, i think I still hv enough DSP - voice-card 0 dspfarm dsp services dspfarm sccp ccm group 1 associate ccm 2 priority 2 associate ccm 1 priority 1 associate profile 1 register cfb001e7a5f9530 associate profile 2 register mtp001e7a5f9530 ! dspfarm profile 2 transcode codec g711ulaw codec g729r8 maximum sessions 2 associate application SCCP ! dspfarm profile 1 conference codec g711ulaw codec g729r8 associate application SCCP shutdown --- NAME: PVDMII DSP SIMM with one DSP, DESCR: PVDMII DSP SIMM with one DSP PID: PVDM2-16 , VID: V01 , SN: FOC0D7C NAME: High Density Voice, DESCR: High Density Voice PID: NM-HDV= , VID: 1.1, SN: JAB05400BRG NAME: PVDM 3-C549 Simm, DESCR: PVDM 3-C549 Simm PID: PVDM-12= , VID: 1.1, SN: NAME: PVDM 3-C549 Simm, DESCR: PVDM 3-C549 Simm PID: PVDM-12= , VID: 1.1, SN: Thks -- Jacob Owen CCIE #14063 (RS, Voice, Service Provider), CCDP
Re: [OSL | CCIE_Voice] Stop routing on unallocated number serviceparam
Thanks Ricardo, Have you tested this scenario? I did this type of test get different results from you. This scenario does not seem to be controlled by this particular parameter. Changing value form TRUE to FALSE and back did not impact the CCM trying GW after GK fails. I shut the serial and I also tested shutting the lo 0, to which GK signals. Also restarted service on separate tests just to make sure. Thanks again. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Arevalo Sent: 25 October 2008 18:49 To: jonny vegas Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Stop routing on unallocated number serviceparam Jonny, Lets say that you hace an extension 3001 in BR2, BR2 and CCM are registered to GK, the normal path from 1001 in CCM to 3001 in BR2 y through GK, that means first RG in the RL. Now... suddenly the BR2-WAN is down, or BR2 get unregistered from GK for any reason (no gateway command in br2 for example) The CCM sends an arq to GK, since GK now does not know about BR2 nor its extensions, it sends back to CCM an arj, meaning unknown destination. If this option (Stop routing on unallocated number ) is set to false in CCM service parameters, the CCM does not stop there, instead, it will look for the next RG in the RL, which could be local gateway, let say in HQ the 6608 T1/Card of T1 Port in BR1... Hope this help let us know if doesn't //r.a. On Sat, Oct 25, 2008 at 12:55 PM, jonny vegas [EMAIL PROTECTED] wrote: Michael, Thanks for the response, much appreciated. You say it covers ANY scenario where the GW or GK can not complete the call. Is this correct? My understanding was it comes into play when the cause code returned for a call failure is 'unallocated number'. Something like ISDN cause code 0x81. Apologies for not being clear in my original mail. What I was asking was specific scenarios where it may actually be of use, in our sort of environment. For example. RL with GK primary and GW secondary. I dial 3009 - an unassigned number at BR2. The call is attempted via the GK first. The far end responds unallocated. If this param is set to FALSE then it will try the GW. But since the number is unallocated there is no point in setting this param and trying to reach the unallocated number a second time, is there? What I am looking for is a scenario where this param is actual of use, in our environment. Many thanks. On Sat, Oct 25, 2008 at 4:58 PM, Michael Shavrov [EMAIL PROTECTED] wrote: Everyone uses it. Basically what it does - when call goes to a gateway or gatekeeper, and the gateway cannot complete the call (for whatever reason - remote side is down, no bandwidth, etc.), CallManager continues searching for other Route Grous in the Route List. Without this parameter (when it set to True), if CallManager receives Unallocated Number signal, it will stop searching for other paths and give you fast busy (basically gateway redundancy will not work). - Original Message - From: jonny vegas mailto:[EMAIL PROTECTED] To: [EMAIL PROTECTED] mailto:ccie_voice@onlinestudylist.com Sent: Saturday, October 25, 2008 10:51 AM Subject: [OSL | CCIE_Voice] Stop routing on unallocated number service param Anyone got scenarios where we would specifically use this param. Thanks
Re: [OSL | CCIE_Voice] OSL | CCIE_Voice] WB Vol1 Sec 8: 4-Digit dialingfrom HQ to BR1 during SRST
You could try using the 'forward no answer internal' on the line of each phone. Just tested the following and it seems to do what you want: - add _pt-SRST - add css-srst - put above pt in it. - add route pattern () for external number to site b to _pt-SRST - on each phone of the SRST site got to the line and fillin the 'forward no answer internal' destination with and css-SRST. - Leave the external one blank or check VM, unless you really need it. Because 3001 may not answer and send it back to voice mail. Potential loop. This will of course break things during normal operation. However if you leave the external one as directed to VM it may end up there eventually. Perhaps a neater option would be to send it off to unity and have unity try the external number before going to voice mail. Something for you to test perhaps, and let us know if you can get it working? Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kumar, Narinder Sent: 30 August 2008 18:20 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] OSL | CCIE_Voice] WB Vol1 Sec 8: 4-Digit dialingfrom HQ to BR1 during SRST Robert, I don't think this scenario will work, When you place the 2003 in its own partition it starts working because it was more specific, in general it won't work not in atleat CCM 4.1(3), 4.2 and above have a option of forwarding during unregistered Cheers NK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 30 August 2008 12:02 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 30, Issue 97 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. WB Vol1 Sec 8: 4-Digit dialing from HQ to BR1 during SRST (Robert Schuknecht) -- Message: 1 Date: Fri, 29 Aug 2008 16:01:43 +0200 From: Robert Schuknecht [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] WB Vol1 Sec 8: 4-Digit dialing from HQ to BR1 during SRST To: ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=utf-8 Hi everybody, in addition to the tasks in WB Vol 1 Section 8, i want to establish 4-digit dialing from HQ to BR1, during SRST in BR1, but i cant get it to work. Here are some Infos: - In Non-SRST Mode all DNs are in the None Partition - During SRST the BR1 Phones are registered to BR1-RTR and reachable via PSTN - created a new Route-Pattern 2XXX which points to a new Route-List and lastly to HQ-GW - Route-Pattern is also in the None Partition When HQ-Ph (1012) dials 2003 i get Fast Busy and there is no ISDN Setup in HQ-GW (see attached CCM Trace Trace1.txt) Then i placed the Route-Pattern 2XXX in its own Partition (PT-HQ-SRST-BR1), same Result Fast Busy. (see attached CCM Trace Trace2.txt) After that, i changed the Route-Pattern to 2003 in ist own Partition (PT-HQ-SRST-BR1) and it works. (see attached CCM Trace Trace3.txt) But that is not the way i like it to work, because then i had to create a Route-Pattern for each DN in BR1. Is there a way to get 4-Digit Dialing, from HQ to BR1, to work? Any hints and tips are very welcome! /Robert -- next part -- 08/29/2008 15:40:31.371 CCM|StationD:(001) StartTone tone=33(InsideDialTone), direction=0.|CLID::StandAloneClusterNID::192.168.120.132CT::2,100,1 19,1.1864IP::192.168.120.143DEV::SEP0019E8F28295 08/29/2008 15:40:31.371 CCM|StationD:(001) DEBUG- star_DSetCallState(4) State of cdpc(11) is 3.|CLID::StandAloneClusterNID::192.168.120.132CT::2,100,119,1.1864 IP::192.168.120.143DEV::SEP0019E8F28295 08/29/2008 15:40:31.872 CCM|StationInit: (001) KeypadButton kpButton=2.|CLID::StandAloneClusterNID::192.168.120.132CT::2,100,11 9,1.1865IP::192.168.120.143DEV::SEP0019E8F28295 08/29/2008 15:40:31.872 CCM|StationD:(001) StopTone.|CLID::StandAloneClusterNID::192.168.120.132CT::2,100,119, 1.1865IP::192.168.120.143DEV::SEP0019E8F28295 08/29/2008 15:40:31.872 CCM|StationD:(001) SelectSoftKeys instance=1 reference=33554455 softKeySetIndex=6 validKeyMask=.|CLID::StandAloneClusterNID::192.168.120.132C T::2,100,119,1.1865IP::192.168.120.143DEV::SEP0019E8F28295 08/29/2008 15:40:31.872 CCM|Digit analysis: wait_DaReq - cepn=[] BlockFlag=[1]|CLID::StandAloneClusterNID::192.168.120.132CT::2,100, 119,1.1865IP::192.168.120.143DEV::SEP0019E8F28295 08/29/2008 15:40:31.872 CCM|Digit Analysis: getDaRes -
Re: [OSL | CCIE_Voice] call forward to CUE from a UCM phone givesdead-air...
This feature should work in that version of code. (should!!) It may be worth checking the output from debug ccsip message and media. You can also use the trace command on CUE. Clear trace then Trace ccn stacksip DBUG or Trace ccn vbrowsercore bdug then Show trace buff tail An interesting one. Steve _ From: Juan [mailto:[EMAIL PROTECTED] Sent: 28 August 2008 07:03 To: Stephen Collinson; OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] call forward to CUE from a UCM phone givesdead-air... Hi Stephen, just checked it: code 124-3j indeed it is... I guess from your reply I need to run another one ;) big thanks! ps: how do you go to find certain behaviour is a bug ... ? cheers, Juan On Wed, Aug 27, 2008 at 6:38 PM, Juan [EMAIL PROTECTED] wrote: I think it's running 12.4.3 jmz or something - need to verify as I don't have access to the lab at this moment. Why? Is this something known, aka. a cisco hidden feature? It's the first time I see it (but I changed CME/CUE platform, so the code is not 100% the same as I had before) cheers, Juan On Wed, Aug 27, 2008 at 1:39 PM, Stephen Collinson [EMAIL PROTECTED] wrote: Nice testing, great to see. Are you running 12.4.3j code? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Sent: 26 August 2008 21:25 To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] call forward to CUE from a UCM phone givesdead-air... hi all, I have yet another issue I did not see before: a call from UCM GK CME (cfwd noan) CUE All I get is dead-air. On the gatekeeper I see an active call (g729 / 16K) between HQ trunk and CME H323 interface (lo0) Also, on the UCM phone the connected party display gets updated to reflect the CUE VM number 3600. 'debug voice call' shows that an outbound dialpeer towards CUE is matched (after the redirect, the first outbound DP match is on the ephone-dn 20001, which is the called CME phone) But when checking the DSPs (sh dspfarm sessions), I only see a session with 1 connection (g711u), instead of the usual 2 connections per session (g711u and g729) Also, sh voice call status does NOT show any active calls. As told, in the meantime, the UCM phone shows connected (with connected party number = CUE VM number), and sh gatekeeper call shows also a G729 call between the HQ trunk and the CME's H323 interface (lo0). PS: To verify that the Xcoder DSPs are working, I can put a UCM - CME call onhold (holder=CME phone and this call is also using the GK) and see the Xcoder doing it's job: 2 connections/per session: 1x g711 (CME MOH stream to xcoder) and 1x g729 (xcoded MOH to UCM phone). So DSP xcoding is working fine on the CME. Also 'allow-connections h323 to sip' and all other possibilities are configured, together with the 'call-forward pattern .T' Any help is greatly appreciated! cheers, Juan
Re: [OSL | CCIE_Voice] GK --- IPIPGW issue
Try changing your source interface for SCCP to the loopback or another fe. I came across a bug of this nature a few weeks ago, it was related to the source interface. Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Key Sent: 25 August 2008 20:06 To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] GK --- IPIPGW issue Configured wb vol1 task 5.6 using GK to route calls through IPIPGW on HQ rtr to phone connected to an ATA. ATA is registered to GK. From an IP phone at HQ, I can ring the phone off of the ATA. After approx 4 seconds of being connected, I get fast busy. I believe issue is with transcoding. I configured local transcoding on the HQ rtr and tried registering the xcoder to both call-manager-fallback and using telephony-service. I get the same cause code error Cause Code: TCP_CONN_ERROR on both when doing sh sccp. Calls from ata phone to ip phone work fine. What am I doing wrong? interface Loopback0 ip address 10.200.200.1 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id VGK ipaddr 10.200.200.1 1719 h323-gateway voip h323-id IPIPGW h323-gateway voip bind srcaddr 10.200.200.1 gatekeeper zone local hqrtr41 matrixlab.local 10.200.200.1 zone local VGK matrixlab.local zone local ATA-GK matrixlab.local invia VGK outvia VGK enable-intrazone zone local UCM-GK matrixlab.local zone remote pstn-wan matrixlab.local 10.200.200.200 1719 no zone subnet hqrtr41 default enable zone subnet hqrtr41 10.168.10.71/32 enable zone subnet hqrtr41 10.168.10.70/32 enable zone prefix pstn-wan 011* zone prefix ATA-GK 208. gw-type-prefix 1#* default-technology no use-proxy ATA-GK default inbound-to terminal no use-proxy ATA-GK default outbound-from terminal bandwidth remote 144 no shutdown sccp local FastEthernet0/0.10 sccp ccm 10.200.200.1 identifier 1 sccp ip precedence 3 sccp sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register xcoder ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec gsmfr codec g729r8 maximum sessions 4 associate application SCCP telephony-service max-ephones 10 max-dn 10 ip source-address 10.200.200.1 port 2000 sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 xcoder max-conferences 8 gain -6 ! ! ephone-dn 1 number 0 ! ! ephone 1 mac-address .. button 1:1 main#sh sccp SCCP Admin State: UP Gateway IP Address: 10.168.10.250, Port Number: 2000 IP Precedence: 3 User Masked Codec list: None Call Manager: 10.200.200.1, Port Number: 2000 Priority: N/A, Version: 3.1, Identifier: 1 Transcoding Oper State: ACTIVE_IN_PROGRESS - Cause Code: TCP_CONN_ERROR Active Call Manager: NONE TCP Link Status: CONNECT_PENDING, Profile Identifier: 1 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: gsmfr, Maximum Packetization Period: 20 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 James Key Network Engineer Gladiator | Matrix Network Services Jack Henry Associates Inc. 2131 E. Primrose Suite H Springfield, MO 65804 Phone: +1.417.890.4294 Fax: +1.417.890.4259 http://www.jackhenry.com/ www.jackhenry.com NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies.
Re: [OSL | CCIE_Voice] B-ACD just dead air...
How are you calling it? PSTN or VOIP g729 or g711u? Going out on a limb here, to perhaps save a few emails. If you are calling in remotely via the GK the incoming call is perhaps g729, depending on what you set on your trunk. This voip call needs an inbound g729 voip dp to match on. When you get the dead air. Do show call active voice comp to see what the call legs are doing. Also do debug voip appl script and see what you get. HTH S _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: 23 August 2008 19:11 To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] B-ACD just dead air... So, I configured B-ACD (from the config on Cisco's site...) and when I call it I get dead air... ! ! interface FastEthernet0/0 ip address 10.0.0.131 255.255.255.0 speed auto no cdp log mismatch duplex h323-gateway voip interface h323-gateway voip id home ipaddr 10.0.0.63 1719 h323-gateway voip h323-id CCME h323-gateway voip tech-prefix 2# h323-gateway voip bind srcaddr 10.0.0.131 ! ! application service queue flash:app-b-acd-2.1.2.2.tcl param queue-len 15 param aa-hunt3 2001 param queue-manager-debugs 1 param aa-hunt2 2000 param number-of-hunt-grps 2 ! service aa flash:app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 param number-of-hunt-grps 2 param handoff-string aa param dial-by-extension-option 1 paramspace english language en param max-time-vm-retry 2 param aa-pilot 5000 paramspace english location flash: param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param voice-mail 4500 param max-time-call-retry 700 param service-name queue ! global service alternate Default ! dial-peer voice 3983 voip service aa destination-pattern 5000 session target ipv4:10.0.0.131 incoming called-number 5000 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! ephone-hunt 1 sequential pilot 2000 list 3003, 3002 statistics collect ! ! ! ! ephone-hunt 2 sequential pilot 2001 list 3002, 3003 ! ! CCME#dir Directory of flash:/ 1 -rw-22201360no date c1700-spservicesk9-mz.124-15.T3.bin 2 -rw- 11650 Aug 4 2008 12:04:24 +00:00 app-cme-did-2.0.0.0.ReadMe 3 -rw- 15020 Aug 4 2008 12:04:25 +00:00 app-cme-did-2.0.0.0.tcl 5 -rw- 18836 Aug 4 2008 12:04:49 +00:00 app-b-acd-2.1.2.2-ReadMe.txt 6 -rw- 24985 Aug 4 2008 12:04:49 +00:00 app-b-acd-2.1.2.2.tcl 7 -rw- 35485 Aug 4 2008 12:04:50 +00:00 app-b-acd-aa-2.1.2.2.tcl 8 -rw- 75650 Aug 4 2008 12:04:51 +00:00 en_bacd_allagentsbusy.au 9 -rw- 83291 Aug 4 2008 12:04:52 +00:00 en_bacd_disconnect.au 10 -rw- 63055 Aug 4 2008 12:04:52 +00:00 en_bacd_enter_dest.au 11 -rw- 37952 Aug 4 2008 12:04:52 +00:00 en_bacd_invalidoption.au 12 -rw- 496521 Aug 4 2008 12:04:58 +00:00 en_bacd_music_on_hold.au 13 -rw- 123446 Aug 4 2008 12:05:00 +00:00 en_bacd_options_menu.au 14 -rw- 42978 Aug 4 2008 12:05:00 +00:00 en_bacd_welcome.au 15 -rw- 34794 Aug 4 2008 12:05:00 +00:00 en_bacd_xferto_operator.au 72 -rw- 42484no date en_dest_busy.au 73 -rw- 26376no date en_dest_unreachable.au 74 -rw- 14352no date en_disconnect.au 75 -rw- 19512no date en_enter_dest.au 76 -rw- 17167no date en_reenter_dest.au 77 -rw- 17486no date en_welcome.au 78 -rw-6627no date its-CISCO.2.0.1.0.tcl 79 -rw-3106no date its_Cisco.2.0.1.0.ReadMe Any ideas? Jonathan
[OSL | CCIE_Voice] Fixing IOS / CME Gatekeeper / RAS source port
I am looking to use alias or hopoff from Gatekeeper towards CME box. Does anyone know how to fix the source port CME/IOS uses to register to gatekeeper? The one in bold underline below (if you accept that sort of email) Thanks Steve CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 1.1.1.1 1720 1.1.1.1 53719 CCM H323-GW H323-ID: Site3-Trunk Voice Capacity Max.= Avail.= Current.= 0
Re: [OSL | CCIE_Voice] Fixing IOS / CME Gatekeeper / RAS source port
Cheers Vik, Much appreciated. Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi Sent: 22 August 2008 18:35 To: Stephen Collinson Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Fixing IOS / CME Gatekeeper / RAS source port For hopoff you don't need to fix the ras port, only the h225 port. It doesn't make sense to use the alias static since the cme can register it's ephone-dn numbers...in any case I have never tried using a static ras port. May not be supported since the cme router could potentially be a gatekeeper. Vik Malhi - CCIE#13890 Senior Technical Instructor - IPexpert Inc Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Join IPexpert's Free CCIE Peer Groups Study Communities at http://www.IPexpert.com/communities www.IPexpert.com/communities On Aug 22, 2008, at 11:13 AM, Stephen Collinson [EMAIL PROTECTED] wrote: I am looking to use alias or hopoff from Gatekeeper towards CME box. Does anyone know how to fix the source port CME/IOS uses to register to gatekeeper? The one in bold underline below (if you accept that sort of email) Thanks Steve CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 1.1.1.1 1720 1.1.1.1 53719 CCM H323-GW H323-ID: Site3-Trunk Voice Capacity Max.= Avail.= Current.= 0
[OSL | CCIE_Voice] You can get a calling line # outbound from SIP trunk
Following adds a calling line number to a call coming in over a SIP trunk going to somewhere else. Add pt-sip-phone and css-sip-phone Set SIP trunk inbound css-sip-trunk Add a translation rule to pt-sip-phone - set calling line number to 12345 - set translation pattern to ! - set css to where you want to go, int + local, long, etc. Works with AAR too. Let me know if you find any problems with it. Steve.
[OSL | CCIE_Voice] AAR to H323 destination GW
Normal path Phone 1 - ccm - wan - phone 2 AAR Phone 1 - ccm -gw1 - PSTN - gw2 - phone 2 Scenario is AAR works fine with MGCP destination GW2. However change GW2 to H323 and AAR does not function. Get the rerouting etc, but no Q931 Setup at GW1. CCM seems to require the dst GW to be in communication. Is this expected behavior? Cheers Steve
Re: [OSL | CCIE_Voice] Auto registration to subscriber - strange problem
Anyone got any thoughts on the following results of testing? Phones have to be registered to the Publisher initially, after which they can be registered to either Pub or Sub as primary. The problem exists when a phone tries to autoreg to the sub. There seems to be a lack of some communication between the pub and sub. The SEP cfg file is never created on the sub and the phone can not register. The phone is sent the 8 block XMLDefault profile. The phone does NOT get registration rejected. Packet traces show all of this. Phone gives up every 20 minutes and start again. Once the auto reg has taken place to the pub the order can be changed to put the sub first in the CCM GROUP list, reload phones and then have Sub as the primary call processing device. The Annunciator / MTP / CFB etc also register to the Sub after a kick. The platform is 2 VMs now running with a single NIC. SR 7 applied to both. DBLHelper shows subscriptions in order. Both machines allowed to read/write to DB. Something I do find which is suspicious is if you do any form of operation requiring a write to the DB, initiated from the subscriber HTTP interface, the CPU goes to 100% and browser hangs. If anyone is really bored and into this sort of thing I can send you packet traces. Sorry to bother you all with this very tedious stuff. Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Collinson Sent: 11 August 2008 23:04 To: 'Vik Malhi'; 'OSL CCIE Voice Lab Exam' Subject: Re: [OSL | CCIE_Voice] Auto registration to subscriber - strangeproblem Thanks Vik, I wish it was something nice like that. Would be nice and quick to fix. Phone/s shows TFTP - sub Call Manager 1 - sub Call Manager 2 - Pub TFTP running on sub as is CCM process. Packet capture on sub shows some TFTP activity for Default phone load and SCCP register, along with on going keepalives. SDL trace shows activity on SUB. Building a new subscriber. Thanks Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi Sent: 11 August 2008 22:46 To: 'Stephen Collinson'; 'OSL CCIE Voice Lab Exam' Subject: Re: [OSL | CCIE_Voice] Auto registration to subscriber - strangeproblem looks like the sub is missing from the ccm group which has auto-reg enabled. Or maybe the sub is beneath the pub in the list of ccm's within the group. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Collinson Sent: Monday, August 11, 2008 12:49 PM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Auto registration to subscriber - strange problem Hi I would appreciate any experience people have with solving the following problem. Auto registration in a clean environment to the subscriber. DHCP address allocation works fine TFTP request for SEPX fails, as expected Request for xml default results in a short tftp transfer with sub Phone state shows registering. Trace shows ongoing SCCP keepalives Phone/s never register. Background info Pub and sub have ip addresses as server identifiers. DHCP running on pub .1 Option 150 set to sub .2 Pub has ccm service running Sub has ccm and tftp running Subscriber set to auto register phones - leave all possible params as default. - Assign extension range 9980 - Same process works fine to publisher. Checked the SDL and CCM trace files for errors indicating subscriber is not part of the cluster, but it seems to be. Is there anything specifically I can look for to prove this? In the Application event log I get some Alarms from the phone, indicating no SEPmacxxx file found. Call manager is at latest SR-7. Thanks Steve
Re: [OSL | CCIE_Voice] Auto registration to subscriber - strange problem
Thanks for the input, always appreciated. Changing option 150 to the publisher may mask/hide the problem, it does not fix what is underlying it. Indeed one does not in this case need to change option 150, since tftp to the Sub is working OK. It is the registration process which fails, it is the ordering of the CCM Group List which determines this behavior. I am trying to understand what causes this behavior. Regards Steve _ From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: 12 August 2008 12:32 To: Stephen Collinson Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Auto registration to subscriber - strange problem Change your option 150 to the Publisher. Jonathan On Tue, Aug 12, 2008 at 6:07 AM, Stephen Collinson [EMAIL PROTECTED] wrote: Anyone got any thoughts on the following results of testing? Phones have to be registered to the Publisher initially, after which they can be registered to either Pub or Sub as primary. The problem exists when a phone tries to autoreg to the sub. There seems to be a lack of some communication between the pub and sub. The SEP cfg file is never created on the sub and the phone can not register. The phone is sent the 8 block XMLDefault profile. The phone does NOT get registration rejected. Packet traces show all of this. Phone gives up every 20 minutes and start again. Once the auto reg has taken place to the pub the order can be changed to put the sub first in the CCM GROUP list, reload phones and then have Sub as the primary call processing device. The Annunciator / MTP / CFB etc also register to the Sub after a kick. The platform is 2 VMs now running with a single NIC. SR 7 applied to both. DBLHelper shows subscriptions in order. Both machines allowed to read/write to DB. Something I do find which is suspicious is if you do any form of operation requiring a write to the DB, initiated from the subscriber HTTP interface, the CPU goes to 100% and browser hangs. If anyone is really bored and into this sort of thing I can send you packet traces. Sorry to bother you all with this very tedious stuff. Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Collinson Sent: 11 August 2008 23:04 To: 'Vik Malhi'; 'OSL CCIE Voice Lab Exam' Subject: Re: [OSL | CCIE_Voice] Auto registration to subscriber - strangeproblem Thanks Vik, I wish it was something nice like that. Would be nice and quick to fix. Phone/s shows TFTP - sub Call Manager 1 - sub Call Manager 2 - Pub TFTP running on sub as is CCM process. Packet capture on sub shows some TFTP activity for Default phone load and SCCP register, along with on going keepalives. SDL trace shows activity on SUB. Building a new subscriber. Thanks Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi Sent: 11 August 2008 22:46 To: 'Stephen Collinson'; 'OSL CCIE Voice Lab Exam' Subject: Re: [OSL | CCIE_Voice] Auto registration to subscriber - strangeproblem looks like the sub is missing from the ccm group which has auto-reg enabled. Or maybe the sub is beneath the pub in the list of ccm's within the group. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Collinson Sent: Monday, August 11, 2008 12:49 PM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Auto registration to subscriber - strange problem Hi I would appreciate any experience people have with solving the following problem. Auto registration in a clean environment to the subscriber. DHCP address allocation works fine TFTP request for SEPX fails, as expected Request for xml default results in a short tftp transfer with sub Phone state shows registering. Trace shows ongoing SCCP keepalives Phone/s never register. Background info Pub and sub have ip addresses as server identifiers. DHCP running on pub .1 Option 150 set to sub .2 Pub has ccm service running Sub has ccm and tftp running Subscriber set to auto register phones - leave all possible params as default. - Assign extension range 9980 - Same process works fine to publisher. Checked the SDL and CCM trace files for errors indicating subscriber is not part of the cluster, but it seems to be. Is there anything specifically I can look for to prove this? In the Application event
Re: [OSL | CCIE_Voice] Auto registration to subscriber - strange problem
Thanks Jonathan, Are you saying the only server to which phones can auto reg is the Publisher? Are you sure option 150 can only point to the Publisher? For some strange reason I thought it was possible to do the following: - Move the ordering of the nodes in the Call Manage Group list, putting the Sub at the top - Enables Auto Registration on the Sub by adding a DN range to the Call Manager entry for the subscriber. This unchecks the disable autoreg on this server check box automatically. - Set option 150 to the subscriber - Make sure TFTP and CCM services are enabled (and running) on the Subscriber Steve _ From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: 12 August 2008 13:26 To: Stephen Collinson Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Auto registration to subscriber - strange problem OK, the cause of the behavior is simple. Whatever is the default CCM group (even if you renamed it), must have auto-registration enabled on the first node listed. Option 150 must point to the CCM publisher, however, the publisher can then redirect the server to download files elsewhere (alternate TFTP locations), as I understand it... I have not implemented off-box TFTP. Jonathan On Tue, Aug 12, 2008 at 7:10 AM, Stephen Collinson [EMAIL PROTECTED] wrote: Thanks for the input, always appreciated. Changing option 150 to the publisher may mask/hide the problem, it does not fix what is underlying it. Indeed one does not in this case need to change option 150, since tftp to the Sub is working OK. It is the registration process which fails, it is the ordering of the CCM Group List which determines this behavior. I am trying to understand what causes this behavior. Regards Steve _ From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: 12 August 2008 12:32 To: Stephen Collinson Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Auto registration to subscriber - strange problem Change your option 150 to the Publisher. Jonathan On Tue, Aug 12, 2008 at 6:07 AM, Stephen Collinson [EMAIL PROTECTED] wrote: Anyone got any thoughts on the following results of testing? Phones have to be registered to the Publisher initially, after which they can be registered to either Pub or Sub as primary. The problem exists when a phone tries to autoreg to the sub. There seems to be a lack of some communication between the pub and sub. The SEP cfg file is never created on the sub and the phone can not register. The phone is sent the 8 block XMLDefault profile. The phone does NOT get registration rejected. Packet traces show all of this. Phone gives up every 20 minutes and start again. Once the auto reg has taken place to the pub the order can be changed to put the sub first in the CCM GROUP list, reload phones and then have Sub as the primary call processing device. The Annunciator / MTP / CFB etc also register to the Sub after a kick. The platform is 2 VMs now running with a single NIC. SR 7 applied to both. DBLHelper shows subscriptions in order. Both machines allowed to read/write to DB. Something I do find which is suspicious is if you do any form of operation requiring a write to the DB, initiated from the subscriber HTTP interface, the CPU goes to 100% and browser hangs. If anyone is really bored and into this sort of thing I can send you packet traces. Sorry to bother you all with this very tedious stuff. Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Collinson Sent: 11 August 2008 23:04 To: 'Vik Malhi'; 'OSL CCIE Voice Lab Exam' Subject: Re: [OSL | CCIE_Voice] Auto registration to subscriber - strangeproblem Thanks Vik, I wish it was something nice like that. Would be nice and quick to fix. Phone/s shows TFTP - sub Call Manager 1 - sub Call Manager 2 - Pub TFTP running on sub as is CCM process. Packet capture on sub shows some TFTP activity for Default phone load and SCCP register, along with on going keepalives. SDL trace shows activity on SUB. Building a new subscriber. Thanks Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi Sent: 11 August 2008 22:46 To: 'Stephen Collinson'; 'OSL CCIE Voice Lab Exam' Subject: Re: [OSL | CCIE_Voice] Auto registration to subscriber - strangeproblem looks like the sub is missing from the ccm group which has auto-reg enabled. Or maybe the sub is beneath the pub in the list of ccm's within the group. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification
[OSL | CCIE_Voice] Auto registration to subscriber - strange problem
Hi I would appreciate any experience people have with solving the following problem. Auto registration in a clean environment to the subscriber. DHCP address allocation works fine TFTP request for SEPX fails, as expected Request for xml default results in a short tftp transfer with sub Phone state shows registering. Trace shows ongoing SCCP keepalives Phone/s never register. Background info Pub and sub have ip addresses as server identifiers. DHCP running on pub .1 Option 150 set to sub .2 Pub has ccm service running Sub has ccm and tftp running Subscriber set to auto register phones - leave all possible params as default. - Assign extension range 9980 - Same process works fine to publisher. Checked the SDL and CCM trace files for errors indicating subscriber is not part of the cluster, but it seems to be. Is there anything specifically I can look for to prove this? In the Application event log I get some Alarms from the phone, indicating no SEPmacxxx file found. Call manager is at latest SR-7. Thanks Steve
Re: [OSL | CCIE_Voice] Auto registration to subscriber - strange problem
Thanks Vik, I wish it was something nice like that. Would be nice and quick to fix. Phone/s shows TFTP - sub Call Manager 1 - sub Call Manager 2 - Pub TFTP running on sub as is CCM process. Packet capture on sub shows some TFTP activity for Default phone load and SCCP register, along with on going keepalives. SDL trace shows activity on SUB. Building a new subscriber. Thanks Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi Sent: 11 August 2008 22:46 To: 'Stephen Collinson'; 'OSL CCIE Voice Lab Exam' Subject: Re: [OSL | CCIE_Voice] Auto registration to subscriber - strangeproblem looks like the sub is missing from the ccm group which has auto-reg enabled. Or maybe the sub is beneath the pub in the list of ccm's within the group. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Collinson Sent: Monday, August 11, 2008 12:49 PM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Auto registration to subscriber - strange problem Hi I would appreciate any experience people have with solving the following problem. Auto registration in a clean environment to the subscriber. DHCP address allocation works fine TFTP request for SEPX fails, as expected Request for xml default results in a short tftp transfer with sub Phone state shows registering. Trace shows ongoing SCCP keepalives Phone/s never register. Background info Pub and sub have ip addresses as server identifiers. DHCP running on pub .1 Option 150 set to sub .2 Pub has ccm service running Sub has ccm and tftp running Subscriber set to auto register phones - leave all possible params as default. - Assign extension range 9980 - Same process works fine to publisher. Checked the SDL and CCM trace files for errors indicating subscriber is not part of the cluster, but it seems to be. Is there anything specifically I can look for to prove this? In the Application event log I get some Alarms from the phone, indicating no SEPmacxxx file found. Call manager is at latest SR-7. Thanks Steve
[OSL | CCIE_Voice] NM-16ESW Architecture
Hi, It's been a while since I look at these NMs in detail. Would appreciate if someone could remind me which switch this card is based on. Thanks Steve
Re: [OSL | CCIE_Voice] NM-16ESW Architecture
Jonathan, If you think about it, it wouldn't make too much sense for cisco to develop a completely new switch just to shove it in a router. They have so many architectures around it has to be based on something. After a little more searching I have an answer. The EtherSwitch network modules are based on the Cisco Catalyst 2950 chipset and have their configuration integrated directly into the hosting router. This provides a single image and configuration file while forcing a feature lag from the compatible desktop switches. http://www.cisco.com/en/US/prod/collateral/routers/ps5854/prod_qas0900aecd80 28d16a.html Steve _ From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: 10 August 2008 17:08 To: Stephen Collinson Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] NM-16ESW Architecture It isn't. the NM-16ESW is basically just 16 PoE ethernet interfaces on a network module... unlike the new NM that replaced it, which is a 3750 which you session into... Jonathan On Sun, Aug 10, 2008 at 11:04 AM, Stephen Collinson [EMAIL PROTECTED] wrote: Hi, It's been a while since I look at these NMs in detail. Would appreciate if someone could remind me which switch this card is based on. Thanks Steve
Re: [OSL | CCIE_Voice] Voice Control Ports
Attached is a list which I copied from some book. Added a couple. I am sure there are a few more, feel free to update and repost. Steve. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Turner Sent: 10 August 2008 22:46 To: Ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Voice Control Ports Does any one have a list of ports used for Voice Signaling? I would like an accurate port list for SIP, SCCP, MGCP and H323. There is some discrepancies with the materials that I am studying from. From my understanding SIP is port 5060 tcp or udp, H323 is tcp 1720, SCCP is tcp 2000, and MGCP is udp 2427 and tcp 2428. Please correct me if I am wrong or if there are other ports that I am missing. Thanks David Voice Ports-1b.doc Description: MS-Word document
[OSL | CCIE_Voice] OSL Password Phishing????
Anyone getting mails from some account in the Netherlands prompting you to go to a web page and enter account details for this list?
Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip
Have you checked debug CCSIP. This gives you info about the codecs. Also some good stuff in the workbook 3 solutions vids. Goes through the complete solution for this sort of thing. That's me done for today. Later - Supper time over here. _ From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: 06 August 2008 20:33 To: Stephen Collinson Cc: cisco voip; OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip Right, the question is, how do you configure it correctly? What would cuz the audio to not cut thru and the call to drop... I was suspecting codec, but it is G711 all the way thru (hard coded on each dial peer) Jonathan On Wed, Aug 6, 2008 at 2:21 PM, Stephen Collinson [EMAIL PROTECTED] wrote: SIP to SIP should work fine, when configured correctly. I was just trying to give you a scenario where we may need to use it. Apologies if this was not helpful _ From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: 06 August 2008 19:55 To: Stephen Collinson Cc: cisco voip; OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip Perhaps I wasn't clear... There is no CUE. This is a SCCP phone on a CCME, and a SCCP phone on CCM with a SIP trunk to an IPIPGW, and a SIP dial-peer to CCME... Jonathan On Wed, Aug 6, 2008 at 1:52 PM, Stephen Collinson [EMAIL PROTECTED] wrote: Perhaps worth looking at your config. You will need sip to sip, say to access CUE VM from a CCM SIP trunk. Check all G711 etc. Debug CCSIP _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: 06 August 2008 18:41 To: OSL CCIE Voice Lab Exam; [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] IPIPGW Sip to Sip So, I was playing with an IPIPGW CCM on one side (SIP trunk) and CCME on the other (SIP dial-peer)... call worked, but as soon as you answered it dropped. I changed the SIP dial-peer from the IPIPGW to H.323 (no session protocol) and RTP cuts thru fine... Am I misreading something, is SIP to SIP not supported, or is my config retarded? Jonathan
Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip
You say from 1008 to 3003. - inbound dial-peer on IPIPGW 501 - outbound dial-peer on IPIPGW 501 - inbound on CME 2001 - desired? What is your outbound protocol for dial-peer 501? Is it perhaps default H323? When you change CME to H323 you then get a match, as opposed to DP 0. Perhaps debug dial-peer on IPIP and CME. Apologies if I am missing something, its getting late over here. Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: 06 August 2008 18:56 To: Nguyen Le Cc: OSL CCIE Voice Lab Exam; [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip Call is from x1008 (on CCM) to x3003 (on CCME) On the IPIPGW: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 modem passthrough nse codec g711ulaw sip dial-peer voice 500 voip destination-pattern 1008 session protocol sipv2 session target ipv4:10.0.0.124 dtmf-relay sip-notify rtp-nte codec g711ulaw ! dial-peer voice 501 voip destination-pattern 3... session target ipv4:10.0.0.131 incoming called-number 3003 codec g711ulaw ! On CCME: dial-peer voice 201 voip destination-pattern 1008 session protocol sipv2 session target ipv4:10.0.0.63 incoming called-number 3003 dtmf-relay rtp-nte codec g711ulaw Jonathan On Wed, Aug 6, 2008 at 12:54 PM, Nguyen Le [EMAIL PROTECTED] wrote: You also have under Voice service voip Allow connections sip to sip ? Also, just double check and make sure your SIP Trunk is in a region that is set to G711 to all other sites Nguyen From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 06, 2008 12:52 PM To: Nguyen Le Cc: OSL CCIE Voice Lab Exam; [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip Yeah, all dial-peers have the codec hard set to 711ulaw Jonathan On Wed, Aug 6, 2008 at 12:43 PM, Nguyen Le [EMAIL PROTECTED] wrote: Jonathan - Make sure your call codec is g711 Nguyen From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Charles Sent: Wednesday, August 06, 2008 12:41 PM To: OSL CCIE Voice Lab Exam; [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] IPIPGW Sip to Sip So, I was playing with an IPIPGW CCM on one side (SIP trunk) and CCME on the other (SIP dial-peer)... call worked, but as soon as you answered it dropped. I changed the SIP dial-peer from the IPIPGW to H.323 (no session protocol) and RTP cuts thru fine... Am I misreading something, is SIP to SIP not supported, or is my config retarded? Jonathan
[OSL | CCIE_Voice] Volume 3 Lab 2 Q 43 Ring no answer to ready state
Chaps, To refresh your memories without having to go and read question. Agent meant to go into a ready state after ring no answer. Configured, tested and watched vid. How did you get it to happen on the phone agent. I can not see it in the vid. Thanks Steve
Re: [OSL | CCIE_Voice] Volume 3 Lab 2 Q 43 Ring no answer to ready state
Nice one, cheers. -Original Message- From: Jonathan Charles [mailto:[EMAIL PROTECTED] Sent: 02 August 2008 14:58 To: Stephen Collinson Cc: Mark Snow; [EMAIL PROTECTED]; OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Volume 3 Lab 2 Q 43 Ring no answer to ready state IPCC System parameters, Agent State after Ring No Answer, set it to Ready... you may need to restart the engine for it to take effect. Jonathan On Sat, Aug 2, 2008 at 5:55 AM, Stephen Collinson [EMAIL PROTECTED] wrote: Chaps, To refresh your memories without having to go and read question. Agent meant to go into a ready state after ring no answer. Configured, tested and watched vid. How did you get it to happen on the phone agent. I can not see it in the vid. Thanks Steve
Re: [OSL | CCIE_Voice] IPMA config bug / problem
No Joy, I had that config but tried it again after restarting the lot. Cheers S _ From: Ricardo Arevalo [mailto:[EMAIL PROTECTED] Sent: 31 July 2008 15:03 To: Stephen Collinson Cc: OSL CCIE Voice Lab Exam; Mark Snow; [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] IPMA config bug / problem Is the Uses Shared Lines option unchecked? Automatic Configuration, Mobile Manager and Uses Shared Lines options must be unchecked in the manager user configuration page, before going to ipma assistant page. rgds//r.a. On Thu, Jul 31, 2008 at 9:52 AM, Stephen Collinson [EMAIL PROTECTED] wrote: When configuring the assistant in IPMA the third row of drop down boxes should get populated with the number of the manager's phone for which we are going to proxy. I do not get this row. Normally it is dynamically created once you select the manager in the middle column. Anyone got any ideas how to sort it? Thanks Steve
Re: [OSL | CCIE_Voice] IPMA config bug / problem
Indeed it is, thanks. Solved the problem by using the browser on the Pub / sub as opposed to my local one. One for me to remember. Thanks for your assistance Ricardo and Vik. Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Arevalo Sent: 31 July 2008 15:03 To: Stephen Collinson Cc: OSL CCIE Voice Lab Exam; Mark Snow; [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] IPMA config bug / problem Is the Uses Shared Lines option unchecked? Automatic Configuration, Mobile Manager and Uses Shared Lines options must be unchecked in the manager user configuration page, before going to ipma assistant page. rgds//r.a. On Thu, Jul 31, 2008 at 9:52 AM, Stephen Collinson [EMAIL PROTECTED] wrote: When configuring the assistant in IPMA the third row of drop down boxes should get populated with the number of the manager's phone for which we are going to proxy. I do not get this row. Normally it is dynamically created once you select the manager in the middle column. Anyone got any ideas how to sort it? Thanks Steve
Re: [OSL | CCIE_Voice] issues with 6608 port going faulty
That gets my vote for best piece of advice this century. Finally we no longer need to be scared of a poxy 6608 blade. That is some great troubleshooting, you should be in TAC or playing chess. Cheers Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tyson Scott Sent: 31 July 2008 20:36 To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] issues with 6608 port going faulty As a note I had several people look at this inside Cisco as well and could never find a solution. We had several ports going out on the 6500's for a while and took me a while to figure out as well. We even replaced the 6608 module at times and the port would give the same error with a new module as the old port. What finally cleared it for us is removing the modules that report the error, clear the configuration for the slot. Putting a module in the slot that is not a 6608 like a 6348 let it finish loading the configuration for that type of module, remove it and clear the configuration for the slot. And put back in the 6608. This is the only thing I have found to clear the 6608 failed to enable port, in the event a reboot or a reset of the module doesn't fix the problem. I would hope they wouldn't make you do this in the lab but I have given away my secret formula to you. This worked in 4 different scenarios so I think it is a shoe win for a work around. Regards, Tyson Scott - CCIE #13513 RS and Security Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Cell: +1.248.504.7309 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video On Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jane Ryer Sent: Thursday, July 31, 2008 3:24 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] issues with 6608 port going faulty On my first attempt at the lab exam in RTP in March of 2008, I was having problems getting the 6608 ports to get an IP address from the DHCP server, and I disabled the ports I was working with and then tried to re-enable them. When I tried to re-enable them, the switch returned an error message Failed to enable port x/1. Port may be faulty., and subsequent sho module commands showed that port as disabled. I tried resetting the module, it did not recover. I went to the proctor, who asked what I had done already and what I wanted him to do, and I asked him to power cycle the switch. He did so, and the port still came up with an orange light and showed as disabled on the sho module command. He then told me to just move over one port (i.e. use port 2,3, and 4 instead of 1,2, and 3) and continue on. What I thought was significant was that even before he power cycled the switch, he made the comment I don't think it's going to help, which indicated to me that he had seen this scenario before. When I returned from that attempt, I did some research on Cisco's internal web site. There are a _lot_ of TAC cases where customers reported that after they had disabled a 6608 port, they got the Port may be faulty error when they tried to re-enable it. In some of these cases, after a few tries the port did come back, but in the vast majority of them, the board was RMA'ed. My second attempt was in SJ in May. There was a sticky note at my pod saying to use ports 2, 3, and 4 of the 6608 blade instead of what was printed in the test. I thought aha, I bet a previous candidate had the same issue I did in RTP. My third attempt was also in SJ in June, and there were no issues with the 6608 blade on that attempt. There may be other hw problems also going on in RTP and/or SJ, but my advice is to never, ever do a set port disable x/x for your 6608 ports. In my first lab attempt, once I moved on (using ports 2, 3, and 4), I found my configuration mistake that was preventing the ports from getting their IP addresses from the DHCP server. If you read through Vik's list of troubleshooting steps, he does not list set port disable as one of the things to try. I do think that the problems re-enabling a port on a 6608 card (after it has been disabled) are due to either a CatOS software bug or (more likely) a firmware bug. But I made the decision not to spend my time and energy tracking it down in detail, because I doubt that the CCIE lab is going to upgrade software or firmware even if we were able to point out the bug to them. Obviously, if the entire 6608 blade is not responding and will not come back after a reload of the switch, then you're in big trouble for that attempt. But if it's a single port that is showing up as disabled, my advice
Re: [OSL | CCIE_Voice] CCM SIP trunk to CCME FXS
I have just tested this setup with very basic config and it works just fine. CCM MTP in NONE, SIP Trunk, RG/RL/RP on CCM 1 SIP dial-peer on IOS GW for in and out voip, no vad, g711, rtp-nte 1 POTS dial-peer for in and out Check out debug voip dial-peer and debug ccsip HTH Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Preethi Thamina Sent: 28 July 2008 19:50 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CCM SIP trunk to CCME FXS Hi, I'm trying the following call scenario: I configured a SIP trunk on CallManager. I checked the MTP Required, assigned it to G711 device pool and MRGL with appropriate G711 media resources. On CCME I have and FXS port with an analog phone and the following config: Dial-peer voice pots Destination pattern Port 1/1/1 Dial-peer voice 3334 voip Incoming called-number Session protocol sipv2 Codec g711ulaw No vad Dtmf-relay rtp-nte Voice service voip allow-connections sip to h323 allow-connections h323 to h323 allow-connections sip to sip allow-connections h323 to h323 voice-port 1/1/1 station-id number Station-id name MyFXS I also have telephony-service configured as well as sccp dspfarm for transcoding. When I make a call from CCM phone to the CCM FXS analog phone, the analog phone rings for 1 second and then I get a busy signal. Same thing happens when the analog phone dials the CCM phone. If I use a H323 ICT trunk as opposed to a Sip trunk on CallManger, the call goes through just fine - so it is not the FXS port or the analog phone. What am I missing? Thanks, Preethi Thamina _ With Windows Live for mobile, your contacts travel with you. Connect on the go. http://www.windowslive.com/mobile/overview.html?ocid=TXT_TAGLM_WL_mobile_07 2008
Re: [OSL | CCIE_Voice] UCCX play ringback
Cheers Michael, Nice one. Was just about to test this, saved me a couple hours work. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Burkett, Michael Sent: 24 June 2008 14:02 To: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] UCCX play ringback Do a search on the Network Hold MOH for the CTI ports, you will find your solution in that direction. Look in the NetPro forums http://forum.cisco.com/eforum/servlet/NetProf?page=netprofforum=Unified %20Communications%20and%20Videotopic=Contact%20CentertopicID=.ee6fe12 CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40%40.1ddec7 09/1#selected_message It will point you here http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item 09186a0080094766.shtml#foura you will find your solution there. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Wow Sent: Tuesday, June 24, 2008 2:16 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX play ringback Does anyone know how to play ringback during the Connected state rather than playing MOH while searching for an available agent? thanks, Chuck ~ This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information of Cameron and its Operating Divisions. Any unauthorized use or disclosure is prohibited. If you are not the intended recipient, please contact the sender by reply email and delete and destroy all copies of the original message inclusive of any attachments. ~
Re: [OSL | CCIE_Voice] Voice lab changes
Cheers Mark, great bit of info. S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Snow Sent: 23 June 2008 15:05 To: OSL CCIE Voice Lab Exam Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Voice lab changes Thanks Jacob - yes announce somewhere between 6-12 mos then 4-6 mos to implement from announce date ... Mark Snow Sr Technical Instructor IPexpert, Inc. Sent from my iPhone On Jun 23, 2008, at 9:47 AM, Jacob Owen [EMAIL PROTECTED] wrote: Chris, What I think they are implying is the announcement of changes won't come for 6 months, and they would take effect 6 months from the announcement date. So for example, a January 1, 2009 announcement = July 1, 2009 take effect and so on. --- Ellington, Chris [EMAIL PROTECTED] wrote: Does that mean that the announcement won't come for 6 months or so and thus the lab will remain the same until next year or possibly 18 months out? chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Snow Sent: Monday, June 23, 2008 9:26 AM To: CCIE Voice Maillist Subject: [OSL | CCIE_Voice] Voice lab changes After talking with some of the Cisco folks in the know here at Networkers, it seems that the Voice lab will remain the same for at minimum 6 but probably up to as much ad 12 months. Look for a announcement to come sometime around Jan to June '09 for a June to Dec '09 rollout. Most likely will be to go to 6.1 since it is in fact the lowest bug count ok any UCM release to date - but of course all specific details such as version number are all being worked out. So good news for those of you who are in the middle of your studies - keep plugging away and watch for future updates! Mark Snow Sr Technical Instructor IPexpert, Inc. Sent from my iPhone Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP
Re: [OSL | CCIE_Voice] IMAP starts and stops on Unity Connection Server
Try following, may be of some help. from cmd Netstat -abno There are a few possible IMAP ports depending on version, etc. Check for TCP 143 or 993 (IMAP / SSL) If you find an entry in the output of the above command you can go into services and turn off the associated process, if its not important. It is not always as simple as the above makes out, process may not relate directly to a service or may be a system process, etc. If no entry in the above list then there may be another problem causing the initialization to fail, check the event log for any further info. HTH S _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bekithemba Nkala (ZA) Sent: 20 June 2008 11:42 To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] IMAP starts and stops on Unity Connection Server My IMAP service keeps starting and stopping on the Unity Connection(version 2.1) server with CallManager 6.1. When I run the RTMT I get this trace . How can I tell which program is taking the IMAP ports ??? 6/11/2008 12:42:13.868 |23831,,,-1,-1,The Connection IMAP Server could not initialize one of the listening sockets. Another application may be using one of ports reserved for IMAP. Stop that application and restart the Connection IMAP service.| 06/11/2008 12:43:18.065 |23941,,,-1,-1,The Connection IMAP Server could not retrieve its configuration from the configuration database. Error code [0x80046600; E_NODE_NOT_FOUND; The Node does not exist.]. The IMAP Server will use default values instead.| 12:41:27.213 |23673,,,-1,-1,The Connection IMAP Server could not retrieve its configuration from the configuration database. Error code [0x80046600; E_NODE_NOT_FOUND; The Node does not exist.]. The IMAP Server will use default values instead.| 06/11/2008 12:42:13.868 |23831,,,-1,-1,The Connection IMAP Server could not initialize one of the listening sockets. Another application may be using one of ports reserved for IMAP. Stop that application and restart the Connection IMAP service.| 06/11/2008 12:43:18.065 |23941,,,-1,-1,The Connection IMAP Server could not retrieve its configuration from the configuration database. Error code [0x80046600; E_NODE_NOT_FOUND; The Node does not exist.]. The IMAP Server will use default values instead.| 06/11/2008 12:44:25.456 |24052,,,-1,-1,The Connection IMAP Server could not initialize one of the listening sockets. Another application may be using one of ports reserved for IMAP. Stop that application and restart the Connection IMAP service.| 06/11/2008 12:45:30.187 |24162,,,-1,-1,The Connection IMAP Server could not retrieve its configuration from the configuration database. Error code [0x80046600; E_NODE_NOT_FOUND; The Node does not exist.]. The IMAP Server will use default values instead.| 06/11/2008 12:46:16.933 |24251,,,-1,-1,The Connection IMAP Server could not initialize one of the listening sockets. Another application may be using one of ports reserved for IMAP. Stop that application and restart the Connection IMAP service.| 06/11/2008 12:47:21.195 |24355,,,-1,-1,The Connection IMAP Server could not retrieve its configuration from the configuration database. Error code [0x80046600; E_NODE_NOT_FOUND; The Node does not exist.]. The IMAP Server will use default values instead.| 06/11/2008 12:48:30.202 |24544,,,-1,-1,The Connection IMAP Server could not initialize one of the listening sockets. Another application may be using one of ports reserved for IMAP. Stop that application and restart the Connection IMAP service.| 06/11/2008 12:49:31.925 |24646,,,-1,-1,The Connection IMAP Server could not retrieve its configuration from the configuration database. Error code [0x80046600; E_NODE_NOT_FOUND; The Node does not exist.]. The IMAP Server will use default values instead.| 06/11/2008 12:50:39.184 |24827,,,-1,-1,The Connection IMAP Server could not initialize one of the listening sockets. Another application may be using one of ports reserved for IMAP. Stop that application and restart the Connection IMAP service.| Thanks BD This email and all contents are subject to the following disclaimer: http://www.dimensiondata.com/emaildisclaimer.htm;
[OSL | CCIE_Voice] CME - Unity MWI problem
Hi, I don't seem to be able to spot the problem here, any help appreciated. - Unity integration tested and working from UTIM. - 4th port set only for out MWI - CME can dial MWI on/off locally and it works - Dial into VM works fine from CME phones - Call-forward works into unity and messages are taken from all locations - Check unity AvCsMgr logs and can see continual MWI dial attempts. CME config is - 3 * ephone + dn number 3600 to access VM - 1 * ephone + dn number A01 - 1 * ephone-dn 3999/3998 MWI on/off On the face of it, looks like and SCCP outbound issue from Unity to CME. Don't seem to be able to find a decent unity log which gives any level of detail as to what is happening. Ephone mwi / vm-int debugs on RTR shows nothing coming in from unity. Cheers S
[OSL | CCIE_Voice] Quick B-ACD query
Would appreciate any insight into why I am getting the following warning when doing BACD config on ISR. The warning comes when one tries to enter the param expressions under the service aa or service queue. Warning: Parameter has not been registered under yy namespace. Xxxx is pretty much any parameter from the scripts YY is the application name - aa or queue Have done some reading on CCO but don't seem to be able to get to the bottom of it. Thanks Steve
Re: [OSL | CCIE_Voice] Quick B-ACD query
Thanks, much appreciated. Steve _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Narvaez Sent: 17 June 2008 15:58 To: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Quick B-ACD query This message is innocuous, disregard it. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Stephen Collinson Enviado el: 17 June 2008 10:43 Para: OSL CCIE Voice Lab Exam Asunto: [OSL | CCIE_Voice] Quick B-ACD query Would appreciate any insight into why I am getting the following warning when doing BACD config on ISR. The warning comes when one tries to enter the param expressions under the service aa or service queue. Warning: Parameter has not been registered under yy namespace. Xxxx is pretty much any parameter from the scripts YY is the application name - aa or queue Have done some reading on CCO but don't seem to be able to get to the bottom of it. Thanks Steve
Re: [OSL | CCIE_Voice] CME - Unity MWI problem
Event log shows a collision. Checked UTIM and MWI is only box with check in it for this port. I am just in the process of flattening everything to start some new labs. I suspect it is possibly something I have been fiddling with in Unity. Will find out in a day or two with the clean setup. Thanks Steve Event Type: Information Event Source:CiscoUnity_TSP Event Category: None Event ID: 127 Date:6/17/2008 Time:03:50:16 User:N/A Computer: EZ Description: Cisco Unity-CM TSP device 12 (Cisco Unity port 8): An attempt to turn ON the message waiting indicator (MWI) for extension 3004 failed because a collision occurred with an incoming call on the same port. The MWI request will be retried. But to prevent collisions, we recommend that ports setting MWIs be isolated from ports handling incoming calls. If the MWI status remains unchanged for an extended period of time or if there are many of these warnings from Cisco Unity in a short period of time, there may be an MWI misconfiguration or another problem. For more information, refer to the Message Waiting Indicators chapter in the Cisco Unity Troubleshooting Guide. For more information, click: http://www.CiscoUnitySupport.com/find.php Repeated again and again and some 1000s more times
Re: [OSL | CCIE_Voice] Unity Call Queueing
Hi Bartosz, Try the following as a starter, it can be enhanced for a more friendly interaction, by adding a call handler. It will queue the call and continually try the extension. Create subscriber - SC x2003 Subscriber-greetings - set 'after greeting' to 'Attempt transfer for SC' Subscriber-call transfer - 'transfer incoming calls to subscriber phone' 'yes, ring subscribers extension 2003' Subscriber-call transfer - 'type transfer' - supervised. Subscriber-call transfer - 'If this call is busy' - always hold or 'ask user' This is quite a nice one to test what happens to the transfer, while the call is in progress, by messing with the call transfer options and if this call is busy options. Don't forget to get your CSS / partitions correct and make sure no restrictions blocking call back in. HTH S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Sokolowski Sent: 13 June 2008 17:20 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unity Call Queueing Hello, I'd like to configure Unity in such way that if I call busy subscriber directly I'd like the call to be queued waiting for subscriber to become available. So for example if I call John at 101 I hear something like John is busy, if you want to wait press 1. For now this functionality works only if I call Unity at pilot number eg. 199 and then within Unity call subscriber extension 101. If I call directly 101 I only hear something like Sorry John is not available, leave a message. Anyone has an idea how to queue calls without calling Unity pilot number? -- Best regards, Bartosz Sokolowski -- Tania telefonia internetowa! Sprawdz http://link.interia.pl/f1e2e
Re: [OSL | CCIE_Voice] QoS Question
For a decent paper on the subject check out. http://www.cisco.com/en/US/tech/tk543/tk757/technologies_tech_note09186a0080 103eae.shtml Worth a quick read. S From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ccievoice1 Sent: 12 June 2008 16:14 To: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] QoS Question Hi Mark, Thanks for the reply :-) Well, the reason for my uncertainty was because I remember some time ago, I was being told that I can't use both percent and bandwidth within a policy-map. And, now I got it clear. Thanks Mark and Devildoc for the explanation. On Thu, Jun 12, 2008 at 11:09 PM, Mark Snow [EMAIL PROTECTED] wrote: It looks like you already typed this in the router (unless you just did it in notepad) - but if you indeed did do it in the router - then you know the answer -right? :) To be more precise - yes - you can have percent on one and bandwidth on another but ONLY if they differ (percent and bandwidth) between the Strict Priority Queue and the normal Congestion Management Queues. You CANNOT have percent and bandwidth both in any of your multiple Congestion Management Queues - you must stick with whatever you choose for them all. -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Jun 12, 2008, at 10:09 AM, ccievoice1 wrote: Hi all, I have some questions on QoS 1.) Can I define LLQ with qos percent in one class and qos bandwifth in another class? ! class-map Media match ip dscp ef ! class-map Control match ip dscp cs3 ! policy-map LLQ class media priority percent 33 class control bandwidth 18 class class-default fair-queue 2.) I have 1M frame-relay link between 2 sites. Based on QoS SRND, frame-relay cir = mincir = 95% of Link Speed. My question is, should I use 1M = 1000 or 1024? IF 1M = 1000, then frame-relay cir = 95 IF 2M = 1024, then frame-relay cir = 972800 Please kindly advice. Thanks
[OSL | CCIE_Voice] Lab exam CCM Features and services PDF
Hi, Would appreciate if anyone could let me know if we get the following guide in the exam. Cisco CallManager Features and Services Guide Cheers S
[OSL | CCIE_Voice] Fax on/off ramp with FXS into Unity
G'Day, Scenario - Microsoft Fax server, into and FXS port, via various different modems. Goal faxing into unity. Not the easiest set of items to troubleshoot. Going through tcl on-ramp script it looks like it should function, but should and working are oceans apart in this case. Out of interest has anyone ever got the above scenario working, with analog modem into FXS? Given it the one day limit, now moving on. Cheers S
Re: [OSL | CCIE_Voice] lab equipments
Hi, I am a big fan of the online labs however I prefer to run my own lab for most things. If you want details mail me off line and Ill be happy to help. HTH S From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marwa Ahmed Sent: 28 May 2008 23:03 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] lab equipments hello, I have just passed my voice written exam, and i am preparing for the lab, i need the list of all equipments that i will use in lab
Re: [OSL | CCIE_Voice] Configuring DSPs Issue on BR2
Hi, Below is an example of the DSP config I employ, with a few annotations. HTH S voice-card 0 dspfarm- Make this cards DSP resources available to FARM pool. dsp services dspfarm - Enable the DSP resources as a DSPFarm. ! sccp local GigabitEthernet0/1.923 - Local SCCP bind sccp ccm 10.0.203.3 identifier 1 - Identify Call mgrs sccp- Start SCCP client side ! sccp ccm group 1 associate ccm 1 priority 1 - 'Which CCM to register to' associate profile 1 register mtp00190665bf39 'What to register' ! dspfarm profile 1 transcode - 'Type of resource' codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 4 associate application SCCP - Bind this resource to SCCP Client side ! telephony-service - Local CCM - SCCP server side for this example sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 mtp00190665bf39 - Resource to be registered. NOTE the name 'mtp00190665bf39' is a MAC format. This format is not necessary on ISR routers, from tests I have run. People seem to spend ages finding MACs off interfaces, etc. I just give it a unique meaningful name, such as XCODER-BR2. If dspfarm profile is shut I find it does not work and nothing is registered. Default when creating seems to be shut. I always forget to no shut it! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Abdalla Abdalla Sent: 04 June 2008 13:39 To: CCIE Voice StudyList Subject: [OSL | CCIE_Voice] Configuring DSPs Issue on BR2 Hi, I was trying to setup a transcoder on BR2 yesterday and could not configure the dspfarm profile. I was getting errors as if the router did not recognise the commands. This is the partial config that I had on BR2. Any explanation to help me understand what is going on will be appreciated. I checked the router had a PVDM2. voice-card 0 dspfarm ! ! voice translation-rule 1 rule 1 /^331323\(3...\)/ /\1/ ! voice translation-rule 2 rule 1 /\(3...\)/ /331323\1/ ! ! voice translation-profile ANI translate calling 2 ! voice translation-profile DNIS translate called 1 ! ! controller E1 0/0/0 pri-group timeslots 1-3,16 ! ! interface Loopback0 ip address 172.3.102.1 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id CME ipaddr 172.3.100.1 1719 h323-gateway voip h323-id BR2-CME h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.3.102.1 ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/0.130 encapsulation dot1Q 130 native no snmp trap link-status ! interface FastEthernet0/0.230 encapsulation dot1Q 230 ip address 10.3.202.1 255.255.255.0 no snmp trap link-status ! interface Service-Engine0/0 no ip address shutdown ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:15 no ip address isdn switch-type primary-net5 isdn incoming-voice voice no cdp enable ! interface Serial0/1/0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/1/0.1 point-to-point ip address 162.3.102.2 255.255.255.0 frame-relay interface-dlci 102 ! router ospf 1 log-adjacency-changes network 10.3.102.0 0.0.0.255 area 0 network 10.3.202.0 0.0.0.255 area 0 network 162.3.102.0 0.0.0.255 area 0 network 172.3.102.0 0.0.0.255 area 0 network 192.3.102.0 0.0.0.255 area 0 ! ip classless ! ! ip http server ip http authentication local no ip http secure-server ! ! ! ! tftp-server flash:P00305000400.bin tftp-server flash:P00303020214.bin tftp-server flash:P00305000301.sbn tftp-server flash:P00403020214.bin tftp-server flash:P00307020200.bin tftp-server flash:P00307020200.loads tftp-server flash:P00307020200.sb2 tftp-server flash:P00307020200.sbn tftp-server flash:P00307020400.bin tftp-server flash:P00307020400.sbn tftp-server flash:P00307020400.loads tftp-server flash:P00307020400.sb2 ! control-plane ! ! ! voice-port 0/0/0:15 translation-profile incoming DNIS translation-profile outgoing ANI ! ! ! sccp local FastEthernet0/0.230 sccp ccm 172.3.102.1 identifier 1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 ! dspfarm profile 1 transcode shutdown ! Regards Abdalla
[OSL | CCIE_Voice] off topic - All email's twice
Does anyone else get every email on this list twice? I have contacted support and asked them to look at it. Have also unsubscribed multiple times and resubscribed with different email address. Simple things. Cheers S
