Re: [OSL | CCIE_Voice] dtmf through gk
Hi Angel, Debug voip ccapi inout and debug h245 events should show the digits received by the cme2 h.323 incoming dial peer. CUE dtmf relay method are sip-notify (unsolicited) and sub-notify by default (in that order), however this can be changed to accept rtp-nte as the main or secondary dmtf relay method, Please take a look the cue dtmf-relay command considerations and allowed combinations, For the MTP I believe this is not necessary as the h245 alpha/signal <-> sip-notify/rtp-nte interworking is supported natively for the cube h.323 <-> sip connections, I would start double checking the dmtf digits are received at cme2, then that are forwarded to cue (debug ccsip mess), Please let us know your findings, Thanks, On Wed, Apr 21, 2010 at 4:40 AM, Angel Perez wrote: > Hi: > > I was thinking that a MTP resource maybe required at CME2 to translate from > h245-alpha to RFC2833 (sip) dtmf > > Could this be a possible failure reason? > > Thanks again > > -- > From: gorr...@hotmail.com > To: ar...@ipexpert.com; ccie_voice@onlinestudylist.com > Date: Wed, 21 Apr 2010 07:57:47 + > > Subject: Re: [OSL | CCIE_Voice] dtmf through gk > > Hi Amy and Otto: > > Both phones are sccp, I've try with h245-alpha at outgoing from cme1 and > incoming from cme2, but no sure if cue was sip-not or rtp... the problem is > that I've reset all the lab configs so I can't do further testing :( > > Just a side note, with the command dtmf-relay it's possible to add more > than one option for example: dtmf-relay h245-alpha rtp-nte, in this case > the second option will be use in case h245 can't be used, is this correct? > > Wich deb or sh commads can I use in this case to check dtmf, I've tried sh > call active voice brief, but I think that dtmf are not shown, maybe deb h245 > asn1? > > I'll try the next time and update > > Best regards > > -- > Date: Tue, 20 Apr 2010 12:22:53 -0400 > Subject: Re: [OSL | CCIE_Voice] dtmf through gk > From: ar...@ipexpert.com > To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com > > Angel, > > Are you using SIP phones? Ensure you are using the below recommendations: > > 1. SIP Phones should have *dtmf-relay rtp-nte > *2. Inbound and Outbound voip dial-peers should have *dtmf-relay > h245-alphanumeric > *3. Dialpeer to CUE should have *dtmf-relay sip-notify > * > Let us know your results. > Amy > > > --- > Amy Ryan – CCIE #24677 (Voice) > Technical Instructor - IPexpert, Inc. > Mailto: *ar...@ipexpert.com <http://ipexpert.com/> > *Telephone: +1.810.326.1444 > Live Assistance, Please visit: www.ipexpert.com/chat <* > http://www.ipexpert.com/chat*> > eFax: +1.810.454.0130 > > IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, > Audio Tools, Online Hardware Rental and Classroom Training for the Cisco > CCIE (R&S, Voice, Security & Service Provider) certification(s) with > training locations throughout the United States, Europe, South Asia and > Australia. Be sure to visit our online communities at > www.ipexpert.com/communities <*http://www.ipexpert.com/communities*> and > our public website at www.ipexpert.com <*http:// > www.ipexpert.com/*> > > > > -- > *From: *Angel Perez http://hotmail.com/>> > *Date: *Tue, 20 Apr 2010 11:56:15 + > *To: *osl osl http://onlinestudylist.com/> > > > *Subject: *[OSL | CCIE_Voice] dtmf through gk > > Hi all; > > I've the following scenario: > > 1001ph--CME1 --- GK --- CME2--2001ph CFALL-> CUE > > > I make a call from 1001 phone at CME1 to 2001 phone at CME2, both gw are > registered to GK, the call arrives at 2002 (that is CFALL to CUE),then the > call is forwarded to CUE (xcd at CME2) and hits 2001 VM, also I can leave a > message, the problem I had is that CUE doesn't read the DTMF I play at > 1001ph > > I've try with some combinations of DTM relay at outbound dial-peer from > CME1, inbound dial-peer at CME2 and outbound dial-peer to CUE but with no > luck > > Any help would be apreciated > > Best regards > > -- > Hotmail: Powerful Free email with security by Microsoft. Get it now. < > https://signup.live.com/signup.aspx?id=60969> > -- > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- > Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up > now. <https://signu
Re: [OSL | CCIE_Voice] Unity Connection Timezones
Hello Ken, 1) This is the time zone used for OS, traces, etc 2) This will be used for user and features withing UC application itself. This timezone is the default timezone schedules use for call handlers, users, etc, hth, On Fri, Apr 9, 2010 at 10:40 AM, Ken Kov wrote: > Can someone tell me how the timezones relate for Unity Connection > > 1) Timezone configured in CLI (change requires reboot) > > versus > > 2) UC Admin > System Settings > General Config > Time Zone (change requires > no reboot) > > Thanks, > Ken > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lets put together Calling party transformation pattern on UCM 7.0.1 behaviour
Hello Jeremy, I think that an important fact to know is that cg/cd xform patterns get matched at the time the rp is being hit, so cg/cd xforms will override the transformations you perform at the rp level, If you configure the EPNM in a TP, the cg number getting to the rp will be the globalized number, right?, in this case, a cg xform pattern will be matched only if it corresponds to that EPNM, Please let me know if this clears up things a little bit for you, On Sun, Apr 11, 2010 at 10:13 PM, jeremy co wrote: > Hi guys , > > I sent an email to this group about the wired problem I had with CngPTP and > EPNM. > > > How does EPNM and CngPTP configuration in multiple place relate to each > other? > > > EPNM could be configured in three places (considering SLRG): TP,RP,GW > > which config override the other one? > > > > How CngPTP kick in? as I can see if I have TP---> RP> SLRG ---> RG and > check TP and RP EPNM then CngPTP will kick in. But checking RP EPNM will > NOT cause CNGPTP to kick in at all. > > another wired issue: > > normal numbers say 911 hit TP and then RP (/+!) > globalized numbers in missed calls say +911 RP(/+!) > > > funny thing is for globalized calls that hit RP directly, checking EPNM on > TP make CngPTP to kick in or not!! > > > > My understanding is CngPTP override should work like this. RP---> RL---> > RG--->CngPTP , > > so CngPTP should override all manipulation in RP,RL,RG, and if EPNM checked > on CngPTP with predot stripping ,it should strip + from EPNM of phones just > before leaving gateway. But CngPTP in this scenario will not kick in at all. > > > > > Anybody can shed some light on how this mess works ? > > > Cheers, > > > Jeremy > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Help with + Dialing Question
Ken, That is the expected behavior when transferring incoming pstn calls to other internal (configured with localization/cg xform patterns css) phones, Please take a look at the following: http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html#wp1266646 hth, On Thu, Apr 8, 2010 at 6:35 PM, Beck, Ken wrote: > > > Direct inbound dialing from the PSTN displays correctly on the phone and in > missed/received calls, however if I dial the AA built in CUC and dial an > extension; the phone displays +1 and the number instead of just the ten > digits. Can someone help me out where to look? Essentially the call is > coming from the VM-ports and everything seems to be configured correctly. > > > > Thanks, > > Ken > > > > Information contained in this e-mail message is intended only for the > individual to whom it is addressed and is private and confidential. If you > are not the intended recipient, or the employee or agent responsible for > delivering this message to the intended recipient, any dissemination, > distribution or copying of this communication is strictly prohibited. If > you are not the intended recipient of this e-mail, please kindly destroy it > and notify the sender immediately by reply e-mail. Thank you for your > cooperation. > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 13A : CUPC Deskphone mode not working and CUPC shows Connection(Limited)
Hello, Please refer to the following blog entry, http://blog.ipexpert.com/cti-phone-control-from-cupc/#more-2272 hth, On Thu, Apr 8, 2010 at 11:16 PM, vccie2010 wrote: > I am doing Lab 13A : CUPC Deskphone mode not working and CUPC shows > Connection(Limited). I have followed the steps in verbatim. I am labbing it > on IPX remote laba nd have CUPC on my laptop. I use SW VPN. Am I missing > something here or is it becoz of VPN. > > thanks for your help... > > -M > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Local Route Group Related question
Hello, You can use 1 route group, in which you will have your two carriers, (the distribution algorithm for them will be the same as the old school model), this route group will be the LRG for your local phones. Then, create a RL that contains the Standard Local Route Group, the RPs should point to this RL, Then, create two sets of css, pt and cd xform patterns (patterns will reflect the different dnis requirements/transformations for each of the carriers), finally, for each of the gw/trunk set the Called Party Transformation CSS field to the already created css, reset the gws/trunks and it should work, hth, On Sun, Apr 4, 2010 at 4:59 AM, Mann Chaddha wrote: > Hi Everyone > > I was wondering how will we manage dual carrier (with diff DNIS > requirements) with the LRG Approach. This is very much a standard at our > firm wherein we always have dual carriers for resiliency sake. With the old > school approach its pretty straightforward with 2 RG for each carrier & > putting them in the Site Specific RL. > > Thnx > Mann > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] BLF speed dial CME
Ash, The blf-speed-dial as well as the speed-dial buttons are assigned to available buttons on the phone, so if you want to enable the blf-speed-dial on the button 3 on the phone, the blf tag has to be 1 as it is the next available button after the line buttons are assigned, hth, On Mon, Apr 5, 2010 at 6:07 PM, Ashar Siddiqui wrote: > Hi all, > > I was doing this blf-speed-dial thing for CME phones and I am not able to get > blf-speed-dial at button 3 of my phone. I don't know how can I set it at > button 3 of the phone. It's coming on button 5 at the moment and I have no > idea why its selecting button 5. > > Here is my config for blf. > > > presence > > presence call-list > > ! > > sip-ua > > presence enable > > ! > > > > ephone-dn 1 > > number 3001 > > allow watch > > > > ephone 2 > > blf-speed-dial *3* 3001 label “BLF 3001” > > I thought the "3" above is used for button but this is not happening! > > ! > ephone 1 > mac-address 0017.9497.1F89 > ephone-template 1 > max-calls-per-button 5 > busy-trigger-per-button 4 > username "scph1" > type 7961 > button 1:1 2:3 > > ephone 2 > mac-address 0017.E089.7382 > ephone-template 1 > max-calls-per-button 3 > busy-trigger-per-button 2 > username "scph2" > type 7961 > button 1:2 2:3 > > > -- > Thanks, > Ashar Siddiqui > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] codecs, BR2/CME/CUE and GK Trunk via HQ
Stephen, A codec mismatch is occurring in br2 when you set codec the sip codec phone as g.711, the outgoing dial peer codec is g.729, therefore you will need a transcoder in br2, To answer to you latter question, yes, you will need the enable inbound faststart, On Mon, Apr 5, 2010 at 3:09 PM, Stephen Greszczyszyn wrote: > I've been working on some standard configurations: > > BR2, one SIP, on SCCP phone, CUE, registered to GK on HQ, uses G729 > over the WAN to HQ phones via GK > > HQ, MGCP local GW, GK trunk registered for calls to BR2 > > The configuration on BR2 for CUE needs to have a G711 voip dial-peer: > > dial-peer voice 3600 voip > destination-pattern 3[126]00 > session protocol sipv2 > session target ipv4:10.10.202.2 > dtmf-relay sip-notify sip-kpml sip-notify > codec g711ulaw > no vad > > The SIP phone on BR2 also needs to have a codec selected. So if I > select G711 for the BR2 SIP phone 3005 all is fine and I can connect > to CUE for vmail. > > The problem comes when calling HQ phones over this GK dial-peer from > the SIP phone to HQ phones: > > dial-peer voice 3510 voip > destination-pattern [15]... > session target ras > dtmf-relay h245-alphanumeric > no vad > > Now when I try to call from the CME/SIP phone to CUCM/HQ phones over > the GK controlled trunk, the call fails. I believe that this has > something to do with not having incoming "fast start" enabled on the > trunk as when I set the SIP phone to codec G729r8 everything works > fine. However, setting the codec of the SIP phone on BR2 to G729r8 > breaks the calls to CUE which needs G711, unless I set up a local > transcoder. > > So how do I get this to setup to work with the CME SIP phone codec set > to G711, and calls to HQ going G729 over the GK trunks? I still > believe that I need incoming "fast start" on the trunk? > > Thanks for any help, > > Stephen > _______ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Directed Call Park - Follow-up from a post back in January
Matthew, I also wish the functionality would work in a simpler way, without that prefix code, however it's not, the blf call park button should be used to park and monitor directed call park slots. I think that if you to try to configure the blf to retrieve the call, then you cannot monitor it, which eliminates the function for the lamp button, Regards, On Fri, Apr 2, 2010 at 8:44 PM, Matthew Berry wrote: > Otto, > > One thing more to add to this discussion. I tried to configure a > translation pattern to take 8555 and prefix it with an 80. My hope was that > pressing the blf directed call park button on the phone would send 8555, > translate to 80-8555, and pickup the parked call. Unfortunately, I could > not do this. Have you noticed if the blf directed call park button sends a > special function to CUCM instead of the configured digits? > > Perhaps my test was wrong to begin with? > > > Matthew Berry > > *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written* > > > > *Gmail:* ciscovoiceguru > > *Skype:* ciscovoiceguru > > *Twitter:* ciscovoiceguru > > *1st Lab Attempt: *Aug 16, 2010 > > On 4/2/2010 7:14 PM, Otto Sanchez wrote: > > Hi Scott, > > For the first question, the directed call park blf button can also be used > to transfer calls to the directed call park slot, i.e. when on an active > call press the transfer softkey then the blf directed call park button and > finally the transfer button again or go on hook (if on hook transfer was > enabled) > > For the second question, correct, the blf is useless to retrieve the parked > call, you must use the call park retrieval prefix, > > For the last statement, the blf directed call park was designed to monitor > and park calls to the directed call park slot to perform the function > already mentioned in the first paragraph, also, in the case of not having > the blf call park button, the systems needs the prefix to differentiate > between an attempt to park a call to the directed call park number or > retrieve it, > > hope this makes sense, > > > On Fri, Apr 2, 2010 at 7:42 AM, scott carruthers > wrote: > >> A follow up on the directed call park scenario with the BLF monitor of the >> slot. Been awhile since I played with this so I want to ensure my >> understanding is correct. Essentially the BLF monitor of directed call park >> really allows nothing beyond having an appearance on the phone that will >> show if a call is currently parked in that slot - correct? The BLF is >> completely useless for actually being able to pickup a parked call - >> correct? Seemed odd to me when I was playing around with it - why program a >> feature like BLF Directed Call Park but not allow the feature to be used to >> actually capture the call. Just want to ensure my memory of the feature is >> correct. >> >> Thanks >> Scott >> >> -- >> Date: Thu, 1 Apr 2010 20:22:11 -0430 >> From: o...@ipexpert.com >> To: ciscovoiceg...@gmail.com >> CC: ccie_voice@onlinestudylist.com >> Subject: Re: [OSL | CCIE_Voice] Directed Call Park - Follow-up from a post >> back in January >> >> >> Matthew, >> >> Right, the statement has to be corrected in the PG, >> >> About the sip dial rule, make sure that for each pattern description, >> there's only one dial parameter pattern and timeout pair, >> >> hth, >> >> >> On Thu, Apr 1, 2010 at 1:13 PM, Matthew Berry >> wrote: >> >> Otto - >> >> I am working through Vol 1 Lab 8 Question 8.2. In the verifications >> section of the PG (p. 478) I am told:* >> * >> >> *"Retrieve the call by pressing the BLF Speed Dial from one of the >> phones..."* >> >> However, whenever I try to retrieve the call this way I get a reorder tone >> and "Park Slot Unavailable." >> >> I was reading one of your responses on the OSL archive to this issue and >> you said: >> >> *"When you hit the directed call park BLF SD, the ucm thinks that you >> want to park a call in that slot, not that are going to retrieve it, and >> since there can be only one call in the park slot, you see the unavailable >> message when a call is already there."* >> >> If that is true, it seems that there is an error in the PG. Does that >> sound right? I am able to retrieve the call by dialing 80-8555, but not by >> going off-hook and pressing the Call Park BLF Speed Dial. >> >> Also, my 8... SIP dial rule with Timeout = 0 has not taken effect after >> several restarts. Any ideas? >> >>
Re: [OSL | CCIE_Voice] Registration Issue - BR2 Phone 4 SIP Phone
Why not get proficient in doing that?, although it will save time doing that with the UCM, perhaps you will have to do it at your everyday job (you won't have a UCM close to you everytime, right?). I have seen so many problems in this list related to not having the proper sip phone firmware when registering them with UCME, that the first thing I would double check before converting from one protocol to another using the UCM, is the phone load version in UCM and compare it with the supported version of UCME, should the version is higher in UCM, I would choose the UCME as my tftp server for upgrading, How would I do it with UCME? The key here is to rely on the universal application loader of the phone and the phone configuration/firmware files in the tftp, then configure your phone as usual on the UCME, so, lets say you want to convert a sccp phone to sip with UCME, follow these guidelines, 1.- Shut down the phone port 2.- Remove the sccp (telephony-service) configurations for that phone and disable auto-register 3.- Make sure the phone tftp configuration is no in the tftp, sh telephony-service tftp will help, if still there recreate the cnf files in telephony service 4.- Create the tftp bindings for all the firmware files, the compatibility matrix will help here to see what are the proper files and the one you should use with the load command in the voice register global configuration 5.- Configure sip as you always do (registrar server, bind commands, allow sip to sip communications, voice register global including the load command and create profile, voice register pool and voice register dn) 6.- Verify the xml config file for the sip phone is in the flash or the default location, sh voice register tftp will help 7.- Turn on the phone 8.- See the debug tftp events output This procedure applies with newer software supporting universal application loader, Try that at your home lab and let us know the results, Helpful reading at: http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cuipph/7960g_7940g/mgcp/firmware/matrix/frmwrup.html HTH, On Fri, Apr 2, 2010 at 8:05 PM, Matthew Berry wrote: > Ugh... So that basically means that I need to get proficient at upgrading > firmware in CUCME instead of using CUCM to move firmware from SCCP to SIP? > If that's the case, then I need to find a good document on that procedure. > I've walked through Cisco's documents on this upgrade procedure and found > them to have deprecated commands and errors. > > Any suggestions, Otto? > > > > Matthew Berry > > *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written* > > > > *Gmail:* ciscovoiceguru > > *Skype:* ciscovoiceguru > > *Twitter:* ciscovoiceguru > > *1st Lab Attempt: *Aug 16, 2010 > > On 4/2/2010 6:30 PM, Otto Sanchez wrote: > > Matthew, > > I experienced a similar behavior before, matching the sip phone firmware to > the ucme version as per the compatibility matrix, solved my problem, > > hth, > > On Thu, Apr 1, 2010 at 10:30 AM, Matthew Berry > wrote: > > For the past few weeks, I've noticed a problem with my BR2 Phone 4 SIP > phone sitting at my desk. It will register just fine when I start my life. > I have to change the "id mac" and enter "authenticate register" into the > "voice register global" section. > > However, I noticed that it will common unregister and reregister. My gut > reaction is that it is a timeout issue that could easily be remedied with a > modification to a SIP timer. However, I'm not sure if this is an issue > that others have seen or if it's just me. > > Below is the latest status messages from the phone. One additional note is > that I see it throwing local errors, CTL errors, and DNS errrors. If not > asked to address those issues in the real lab, would I still lose points for > such errors? > > 10:52:07 DNS Unknown Host > 10:52:08 Error Updating Locale > 10:52:08 Error Updating Locale > 10:52:44 File Not Found : CTLFile.tlv > 10:52:44 No CTL installed > 10:52:44 SEP0021D1F6.cnf.xml > 10:52:53 DNS Unknown Host > 10:52:53 DNS Unknown Host > 10:52:54 Error Updating Locale > 10:52:54 Error Updating Locale > -- > > Matthew Berry > > *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written* > > > > *Gmail:* ciscovoiceguru > > *Skype:* ciscovoiceguru > > *Twitter:* ciscovoiceguru > > *1st Lab Attempt: *Aug 16, 2010 > > _______ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Transcoding at CME
Ash, You can actually configure any name containing max 15 characters, the same that you defined in the associate profile command and not need to be in the mtp and the mac address format. For the interface, you can use any other, however I would stay away from the same defined for the telephony-service as the transcoding/conf resources sccp registration is going to use the same address/port (2000) as the telephony-service, which may cause conflicts, hth, On Fri, Apr 2, 2010 at 10:30 AM, Ashar Siddiqui wrote: > Hi, > > I want to configure transcoding on a CME router but as you know its bit > different of how we configure transcoding for CUCM. > Is it necessary to have a physical interface for binding? Can I bind it on > loopback or Voice Vlan? > > Below I have used fastethernet 0/0 so that I can have a physical > mac-address but was wondering if there is any other way round (like using L0 > or Vlan) > 00164767cc20 - MAC address of fa0/0. > > voice-card 0 > > dspfarm > > dsp services dspfarm > > > > sccp local fa0/0 > > sccp ccm 10.10.110.3 identifier 1 > > sccp > > > > sccp ccm group 1 > > bind int fa0/0 > > associate ccm 1 priority 1 > > associate profile 1 regsiter mtp00164767cc20 > > keepalives retries 5 > > switchover medthod immediate > > switch-back interval 5 > > > > dspfarm profile 1 transcode > > codec g729r8 > > codec g711ulaw > > codec g711alaw > > max session 3 > > associate application sccp > > end > > > > > > telephony-service > > sdspfarm units 2 > > sdspfarm transcode sessions 16 > sdspfarm tag 1 mtp*00164767cc20 * > > -- > Thanks, > Ashar Siddiqui > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Directed Call Park - Follow-up from a post back in January
Hi Scott, For the first question, the directed call park blf button can also be used to transfer calls to the directed call park slot, i.e. when on an active call press the transfer softkey then the blf directed call park button and finally the transfer button again or go on hook (if on hook transfer was enabled) For the second question, correct, the blf is useless to retrieve the parked call, you must use the call park retrieval prefix, For the last statement, the blf directed call park was designed to monitor and park calls to the directed call park slot to perform the function already mentioned in the first paragraph, also, in the case of not having the blf call park button, the systems needs the prefix to differentiate between an attempt to park a call to the directed call park number or retrieve it, hope this makes sense, On Fri, Apr 2, 2010 at 7:42 AM, scott carruthers wrote: > A follow up on the directed call park scenario with the BLF monitor of the > slot. Been awhile since I played with this so I want to ensure my > understanding is correct. Essentially the BLF monitor of directed call park > really allows nothing beyond having an appearance on the phone that will > show if a call is currently parked in that slot - correct? The BLF is > completely useless for actually being able to pickup a parked call - > correct? Seemed odd to me when I was playing around with it - why program a > feature like BLF Directed Call Park but not allow the feature to be used to > actually capture the call. Just want to ensure my memory of the feature is > correct. > > Thanks > Scott > > -- > Date: Thu, 1 Apr 2010 20:22:11 -0430 > From: o...@ipexpert.com > To: ciscovoiceg...@gmail.com > CC: ccie_voice@onlinestudylist.com > Subject: Re: [OSL | CCIE_Voice] Directed Call Park - Follow-up from a post > back in January > > > Matthew, > > Right, the statement has to be corrected in the PG, > > About the sip dial rule, make sure that for each pattern description, > there's only one dial parameter pattern and timeout pair, > > hth, > > > On Thu, Apr 1, 2010 at 1:13 PM, Matthew Berry wrote: > > Otto - > > I am working through Vol 1 Lab 8 Question 8.2. In the verifications > section of the PG (p. 478) I am told:* > * > > *"Retrieve the call by pressing the BLF Speed Dial from one of the > phones..."* > > However, whenever I try to retrieve the call this way I get a reorder tone > and "Park Slot Unavailable." > > I was reading one of your responses on the OSL archive to this issue and > you said: > > *"When you hit the directed call park BLF SD, the ucm thinks that you want > to park a call in that slot, not that are going to retrieve it, and since > there can be only one call in the park slot, you see the unavailable message > when a call is already there."* > > If that is true, it seems that there is an error in the PG. Does that > sound right? I am able to retrieve the call by dialing 80-8555, but not by > going off-hook and pressing the Call Park BLF Speed Dial. > > Also, my 8... SIP dial rule with Timeout = 0 has not taken effect after > several restarts. Any ideas? > > Please let me know if my assumption is correct. > > -- > Matthew Berry > *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written* > > *Gmail:* ciscovoiceguru > *Skype:* ciscovoiceguru > *Twitter:* ciscovoiceguru > *1st Lab Attempt: *Aug 16, 2010 > > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com > > -- > The New Busy is not the old busy. Search, chat and e-mail from your inbox. Get > started.<http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3> > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Registration Issue - BR2 Phone 4 SIP Phone
Matthew, I experienced a similar behavior before, matching the sip phone firmware to the ucme version as per the compatibility matrix, solved my problem, hth, On Thu, Apr 1, 2010 at 10:30 AM, Matthew Berry wrote: > For the past few weeks, I've noticed a problem with my BR2 Phone 4 SIP > phone sitting at my desk. It will register just fine when I start my life. > I have to change the "id mac" and enter "authenticate register" into the > "voice register global" section. > > However, I noticed that it will common unregister and reregister. My gut > reaction is that it is a timeout issue that could easily be remedied with a > modification to a SIP timer. However, I'm not sure if this is an issue > that others have seen or if it's just me. > > Below is the latest status messages from the phone. One additional note is > that I see it throwing local errors, CTL errors, and DNS errrors. If not > asked to address those issues in the real lab, would I still lose points for > such errors? > > 10:52:07 DNS Unknown Host > 10:52:08 Error Updating Locale > 10:52:08 Error Updating Locale > 10:52:44 File Not Found : CTLFile.tlv > 10:52:44 No CTL installed > 10:52:44 SEP0021D1F6.cnf.xml > 10:52:53 DNS Unknown Host > 10:52:53 DNS Unknown Host > 10:52:54 Error Updating Locale > 10:52:54 Error Updating Locale > -- > > Matthew Berry > > *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written* > > > > *Gmail:* ciscovoiceguru > > *Skype:* ciscovoiceguru > > *Twitter:* ciscovoiceguru > > *1st Lab Attempt: *Aug 16, 2010 > > _______ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] trasnfer to distribution list
Make sure the distribution list partition is included in the user/subscriber (the one addressing the message) search space, If still doesn't work, send your detailed configuration for partitions, search spaces, user and distribution list. Make sure also you followed the directions already given and provide feedback on the exact sequence of the call, On Thu, Apr 1, 2010 at 9:57 AM, J Hogan wrote: > I am using 7.1.3.1-68 > And yes I am using partitions. > > Thanks > > > Sent from my iPhone > J. Hogan > VP of Engineering > 1-217-337-1005 > > > On Apr 1, 2010, at 7:15 AM, Otto Sanchez wrote: > > Hi, > > What unity connection version are you using? > Are you using any partitioning within your unity connection system? > > If using UC 7.x, please make sure of the following: > > 1.- You create the distribution list according to the following (assign a > extension number to it): > <http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag230.html#wp1049747> > http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag230.html#wp1049747 > > 2.- The COS the subscriber is assigned to allows to send messages to > distribution lists, i.e. check the "Allow Users to Send Messages to System > Distribution Lists" chechbox for that user > > 3.- You send the message to the DL using this: > > <http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/user/guide/phone/7xcucugphone030.html#wp1010945> > http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/user/guide/phone/7xcucugphone030.html#wp1010945 > Press ## when addressing the message to specify the DL extension number, > > 4.- Also, record a name for the list, > > > > > On Wed, Mar 31, 2010 at 11:07 PM, J Hogan < > j.jho...@gmail.com> wrote: > >> I have not recorded a name for it but I Do have members >> >> >> Sent from my iPhone >> J. Hogan >> VP of Engineering >> 1-217-337-1005 >> >> >> On Mar 31, 2010, at 9:47 PM, Otto Sanchez < >> o...@ipexpert.com> wrote: >> >> Have you recorded a name for it?, have you included members? >> >> On Wed, Mar 31, 2010 at 10:10 PM, J Hogan < >> >> j.jho...@gmail.com> wrote: >> >>> Busy tone >>> >>> Sent from my iPhone >>> J. Hogan >>> VP of Engineering >>> 1-217-337-1005 >>> >>> >>> On Mar 31, 2010, at 9:29 PM, Otto Sanchez < >>> >>> o...@ipexpert.com> wrote: >>> >>> What happens when a message to the distribution list is sent from the >>> subscriber conversation menu?, >>> >>> >>> On Wed, Mar 31, 2010 at 9:24 AM, J Hogan < >>> >>> j.jho...@gmail.com> wrote: >>> >>>> I have tried to create a call handler to do that But All i get is busy. >>>> I even created a interview handler that goes to a distribution list. >>>> >>>> >>>> On Wed, Mar 31, 2010 at 6:33 AM, Otto Sanchez < >>>> >>>> o...@ipexpert.com> wrote: >>>> >>>>> Hello, >>>>> >>>>> I guess you are trying to do this from a call handler right?, a >>>>> possibility for this is to use the take message option for the >>>>> corresponding >>>>> caller input key, and specify the distribution list as the message >>>>> recipient >>>>> for the call handler, >>>>> >>>>> Let me know if that works for you, >>>>> >>>>> On Tue, Mar 30, 2010 at 10:38 PM, J Hogan < >>>>> >>>>> j.jho...@gmail.com> wrote: >>>>> >>>>>> Has any one configured unity connection to transfer calls to a >>>>>> distribution list? >>>>>> I have been tyring this and have been extremely unsuccessful. >>>>>> >>>>>> ___ >>>>>> For more information regarding industry leading CCIE Lab training, >>>>>> please visit <http://www.ipexpert.com> >>>>>> <http://www.ipexpert.com><http://www.ipexpert.com> >>>>>> www.ipexpert.com >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> >>>>> Otto Sanchez >>>>> CCIE #25592 (Voice) >>>>> Support Engineer - IPexpert, Inc. >>>&g
Re: [OSL | CCIE_Voice] Directed Call Park - Follow-up from a post back in January
Matthew, Right, the statement has to be corrected in the PG, About the sip dial rule, make sure that for each pattern description, there's only one dial parameter pattern and timeout pair, hth, On Thu, Apr 1, 2010 at 1:13 PM, Matthew Berry wrote: > Otto - > > I am working through Vol 1 Lab 8 Question 8.2. In the verifications > section of the PG (p. 478) I am told:* > * > > *"Retrieve the call by pressing the BLF Speed Dial from one of the > phones..."* > > However, whenever I try to retrieve the call this way I get a reorder tone > and "Park Slot Unavailable." > > I was reading one of your responses on the OSL archive to this issue and > you said: > > *"When you hit the directed call park BLF SD, the ucm thinks that you want > to park a call in that slot, not that are going to retrieve it, and since > there can be only one call in the park slot, you see the unavailable message > when a call is already there."* > > If that is true, it seems that there is an error in the PG. Does that > sound right? I am able to retrieve the call by dialing 80-8555, but not by > going off-hook and pressing the Call Park BLF Speed Dial. > > Also, my 8... SIP dial rule with Timeout = 0 has not taken effect after > several restarts. Any ideas? > > Please let me know if my assumption is correct. > > -- > > Matthew Berry > > *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written* > > > > *Gmail:* ciscovoiceguru > > *Skype:* ciscovoiceguru > > *Twitter:* ciscovoiceguru > > *1st Lab Attempt: *Aug 16, 2010 > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] trasnfer to distribution list
Hi, What unity connection version are you using? Are you using any partitioning within your unity connection system? If using UC 7.x, please make sure of the following: 1.- You create the distribution list according to the following (assign a extension number to it): http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/administration/guide/7xcucsag230.html#wp1049747 2.- The COS the subscriber is assigned to allows to send messages to distribution lists, i.e. check the "Allow Users to Send Messages to System Distribution Lists" chechbox for that user 3.- You send the message to the DL using this: http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/user/guide/phone/7xcucugphone030.html#wp1010945 Press ## when addressing the message to specify the DL extension number, 4.- Also, record a name for the list, On Wed, Mar 31, 2010 at 11:07 PM, J Hogan wrote: > I have not recorded a name for it but I Do have members > > > Sent from my iPhone > J. Hogan > VP of Engineering > 1-217-337-1005 > > > On Mar 31, 2010, at 9:47 PM, Otto Sanchez wrote: > > Have you recorded a name for it?, have you included members? > > On Wed, Mar 31, 2010 at 10:10 PM, J Hogan < > j.jho...@gmail.com> wrote: > >> Busy tone >> >> Sent from my iPhone >> J. Hogan >> VP of Engineering >> 1-217-337-1005 >> >> >> On Mar 31, 2010, at 9:29 PM, Otto Sanchez < >> o...@ipexpert.com> wrote: >> >> What happens when a message to the distribution list is sent from the >> subscriber conversation menu?, >> >> >> On Wed, Mar 31, 2010 at 9:24 AM, J Hogan < >> >> j.jho...@gmail.com> wrote: >> >>> I have tried to create a call handler to do that But All i get is busy. I >>> even created a interview handler that goes to a distribution list. >>> >>> >>> On Wed, Mar 31, 2010 at 6:33 AM, Otto Sanchez < >>> >>> o...@ipexpert.com> wrote: >>> >>>> Hello, >>>> >>>> I guess you are trying to do this from a call handler right?, a >>>> possibility for this is to use the take message option for the >>>> corresponding >>>> caller input key, and specify the distribution list as the message >>>> recipient >>>> for the call handler, >>>> >>>> Let me know if that works for you, >>>> >>>> On Tue, Mar 30, 2010 at 10:38 PM, J Hogan < >>>> >>>> j.jho...@gmail.com> wrote: >>>> >>>>> Has any one configured unity connection to transfer calls to a >>>>> distribution list? >>>>> I have been tyring this and have been extremely unsuccessful. >>>>> >>>>> ___ >>>>> For more information regarding industry leading CCIE Lab training, >>>>> please visit <http://www.ipexpert.com> <http://www.ipexpert.com> >>>>> www.ipexpert.com >>>>> >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> Otto Sanchez >>>> CCIE #25592 (Voice) >>>> Support Engineer - IPexpert, Inc. >>>> URL: <http://www.IPexpert.com> <http://www.IPexpert.com> >>>> http://www.IPexpert.com >>>> >>> >>> >>> >>> -- >>> J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI >>> Yahoo ID: jhogan552000 >>> AIM ID: jhogan55 >>> MSN ID: jhogan55 >>> ICQ ID: 257599283 >>> >>> Work hard and get a check, >>> Work smart and earn a living >>> >> >> >> >> -- >> Regards, >> >> Otto Sanchez >> CCIE #25592 (Voice) >> Support Engineer - IPexpert, Inc. >> URL: <http://www.IPexpert.com> <http://www.IPexpert.com> >> http://www.IPexpert.com >> >> > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: <http://www.IPexpert.com>http://www.IPexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Priority Queue on UCCX 5.0.2
Hi Cris, Please take a look to the following docs: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_5_0/programming/script_devp/gs_with_scripts/crs501gs.pdf Cisco CRS Scripting and Development Series: Volume 1, Getting Started with Scripts 5.0(1) , Chapter 17, shows an example on how to set up the set priotity step http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_5_0/programming/script_devp/editor_step_ref/crs501sr.pdf Cisco CRS Scripting and Development Series: Volume 2, Editor Step Reference 5.0(1) , show the set priority step properties Also make sure you are not using standard licenses with your uccx deployment, On Wed, Mar 31, 2010 at 7:09 PM, Cristobal Priego wrote: > Hello all, > > I'd like to know if you have any documentation or if you could point me on > the proper track. I'd like to set up a priority queue on my script. I'm > using the priority step, but it doesn't seem to be working properly. how do > i do it > > thanks > > Cris > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] trasnfer to distribution list
Have you recorded a name for it?, have you included members? On Wed, Mar 31, 2010 at 10:10 PM, J Hogan wrote: > Busy tone > > Sent from my iPhone > J. Hogan > VP of Engineering > 1-217-337-1005 > > > On Mar 31, 2010, at 9:29 PM, Otto Sanchez wrote: > > What happens when a message to the distribution list is sent from the > subscriber conversation menu?, > > > On Wed, Mar 31, 2010 at 9:24 AM, J Hogan < > j.jho...@gmail.com> wrote: > >> I have tried to create a call handler to do that But All i get is busy. I >> even created a interview handler that goes to a distribution list. >> >> >> On Wed, Mar 31, 2010 at 6:33 AM, Otto Sanchez < >> o...@ipexpert.com> wrote: >> >>> Hello, >>> >>> I guess you are trying to do this from a call handler right?, a >>> possibility for this is to use the take message option for the corresponding >>> caller input key, and specify the distribution list as the message recipient >>> for the call handler, >>> >>> Let me know if that works for you, >>> >>> On Tue, Mar 30, 2010 at 10:38 PM, J Hogan < >>> j.jho...@gmail.com> wrote: >>> >>>> Has any one configured unity connection to transfer calls to a >>>> distribution list? >>>> I have been tyring this and have been extremely unsuccessful. >>>> >>>> ___ >>>> For more information regarding industry leading CCIE Lab training, >>>> please visit <http://www.ipexpert.com>www.ipexpert.com >>>> >>>> >>> >>> >>> -- >>> Regards, >>> >>> Otto Sanchez >>> CCIE #25592 (Voice) >>> Support Engineer - IPexpert, Inc. >>> URL: <http://www.IPexpert.com>http://www.IPexpert.com >>> >> >> >> >> -- >> J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI >> Yahoo ID: jhogan552000 >> AIM ID: jhogan55 >> MSN ID: jhogan55 >> ICQ ID: 257599283 >> >> Work hard and get a check, >> Work smart and earn a living >> > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: <http://www.IPexpert.com>http://www.IPexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] trasnfer to distribution list
What happens when a message to the distribution list is sent from the subscriber conversation menu?, On Wed, Mar 31, 2010 at 9:24 AM, J Hogan wrote: > I have tried to create a call handler to do that But All i get is busy. I > even created a interview handler that goes to a distribution list. > > > On Wed, Mar 31, 2010 at 6:33 AM, Otto Sanchez wrote: > >> Hello, >> >> I guess you are trying to do this from a call handler right?, a >> possibility for this is to use the take message option for the corresponding >> caller input key, and specify the distribution list as the message recipient >> for the call handler, >> >> Let me know if that works for you, >> >> On Tue, Mar 30, 2010 at 10:38 PM, J Hogan wrote: >> >>> Has any one configured unity connection to transfer calls to a >>> distribution list? >>> I have been tyring this and have been extremely unsuccessful. >>> >>> ___ >>> For more information regarding industry leading CCIE Lab training, please >>> visit www.ipexpert.com >>> >>> >> >> >> -- >> Regards, >> >> Otto Sanchez >> CCIE #25592 (Voice) >> Support Engineer - IPexpert, Inc. >> URL: http://www.IPexpert.com >> > > > > -- > J. Hogan MCP,CCDA,CCDP, CCNA, CCNP, CCSP, CCAI > Yahoo ID: jhogan552000 > AIM ID: jhogan55 > MSN ID: jhogan55 > ICQ ID: 257599283 > > Work hard and get a check, > Work smart and earn a living > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] trasnfer to distribution list
Hello, I guess you are trying to do this from a call handler right?, a possibility for this is to use the take message option for the corresponding caller input key, and specify the distribution list as the message recipient for the call handler, Let me know if that works for you, On Tue, Mar 30, 2010 at 10:38 PM, J Hogan wrote: > Has any one configured unity connection to transfer calls to a distribution > list? > I have been tyring this and have been extremely unsuccessful. > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PL - Inability To Change CUE License
Hi Scott, What I would do in this case is reinstall of the module application software, using the following procedure: http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel7_0/install/upgrade.html#wp1148189 If still with problems I would use the reimage procedure from the bootloader, located in the same document, If you want to move on, just ask to be moved to another pod and have support reinstall the module, Thanks, On Fri, Mar 26, 2010 at 8:42 PM, scott carruthers wrote: > > Anyone ever encounter the following error when attempting to change the CUE > license on proctorlab's modules? Attempting to change from the CME to CM > license. Would seem to be an obvious flash space issue but I cleared some > crash files, etc but no actions seem to allow the license install. > Additionally - below the file install attempt you can see the flash is > really not all that low on space. I've tried resetting the module several > times but nothing helps. > > se-10-10-202-250# $p://10.10.210.5/cue-vm-license_12mbx_ccm_7.0.1.pkg > Online install/download is not allowed due to insufficient FLASH capacity > > 256503808 bytes total (123088896 bytes free) > > Thanks > Scott > > -- > Hotmail: Trusted email with powerful SPAM protection. Sign up > now.<http://clk.atdmt.com/GBL/go/210850553/direct/01/> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 7 - Multicast MoH
Hi Matthew, I would put the moh server in the same location as all devices in hq, why?, well, I wouldn't care too much about multicast because no matter the location where you put it, bandwidth is not going to be tracked by ucm cac mechanisms, however the same moh server can be used for unicast moh in which case the ucm cac does track the bandwidth consumption for all the calls being put on hold. As SRND states, what you should take care about when configuring multicast moh is to over provision the priority queue in downstream router interfaces, hth, On Sun, Mar 28, 2010 at 9:38 AM, Matthew Berry wrote: > Question: According to what you understand of the Cisco SRND, would the > music on hold servers need to be in a location that was using RSVP-enabled > CAC? For this lab, I put them in a location called LOC-HQ-NoRSVP. When it > comes to multicast, I would assume that this would be appropriate. I could > see a reason for putting MoH under RSVP CAC when using unicast streams since > that could definitely tax the system. > > Screenshot attached/inline > > > Matthew Berry > *Gmail: ciscovoiceguru > Skype: ciscovoiceguru > Twitter: ciscovoiceguru > 1st Lab Attempt: Aug 16, 2010* > > > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com <>___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Warning for all those getting ready to go through Vol 1 Lab 7 - Multicast MOH, MMOH
Hi Matthew, You can also use the pstn phone to listen the multicast moh, i.e, if you want to hear multicast moh streamed to br1 devices, you may call 1002 at br1 from the pstn phone and press the hold softkey from 1002. What will happen is that ucm will stream multicast moh from hq to br1 rtr (all within PL, no multicast over the VPN tunnel) then rtr will send that music to the pstn phone over the ds0 channel used, hth, On Sun, Mar 28, 2010 at 2:24 PM, Matthew Berry wrote: > As clearly stated in the Proctor Labs login screen: > > > > Just be aware that multicast music on hold does not work on the Proctor > Labs setup. > > Your best bet will be to use *show perf query class "Cisco MOH Device" *to > make sure the multicast is flowing properly. > -- > > Matthew Berry > > *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written* > > > > *Gmail:* ciscovoiceguru > > *Skype:* ciscovoiceguru > > *Twitter:* ciscovoiceguru > > *1st Lab Attempt: *Aug 16, 2010 > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com <>___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Integration of UCM and UC
Hello, Did you also checked that: 1.- Sip trunk security profile has Accept Unsolicited Notification checked 2.- Some ports in UC are enabled to Send MWI Requests Thanks, On Mon, Mar 22, 2010 at 11:40 PM, Omotayo wrote: > Hello Otto, > > I checked the Redirecting Diversion Header Delivery - Inbound and Redirecting > Diversion Header Delivery - outbound > > > Voicemail works now but MWI is not working > > what do i need to do to fix it > > thanks > > > On Mon, Mar 22, 2010 at 10:56 AM, Omotayo wrote: > >> Hello, >> That should be on the sip trunk right? >> >> I am not sure i checked that. i will confirm today and give you update >> Regards >> >> On Mon, Mar 22, 2010 at 2:07 AM, Otto Sanchez wrote: >> >>> I meant for the *Out*bound direction, i.e., from ucm to uc, >>> >>> >>> >>> On Sun, Mar 21, 2010 at 5:51 PM, Otto Sanchez wrote: >>> >>>> Hi, >>>> >>>> Did you take a look at this document? >>>> >>>> >>>> http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.html >>>> >>>> <http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.html>Also, >>>> make sure the Redirecting Diversion Header Delivery - Inbound is >>>> checked, >>>> >>>> hth, >>>> >>>> On Fri, Mar 19, 2010 at 1:35 PM, Omotayo wrote: >>>> >>>>> i have been able to get this work. i have checked all doc but no >>>>> solution >>>>> I still need help on this >>>>> thanks >>>>> >>>>> On Wed, Mar 17, 2010 at 3:29 PM, Omotayo wrote: >>>>> >>>>>> Hello, >>>>>> Any ideas? >>>>>> >>>>>> On Wed, Mar 17, 2010 at 9:54 AM, Omotayo wrote: >>>>>> >>>>>>> Hello All, >>>>>>> >>>>>>> On Lab 7, after integrating the UCM with the UC using SIP. Pressing >>>>>>> the subscriber button, i get the personal greeting message >>>>>>> But, when pstn or a local call dials hq phone 2 or br1 phone 2, i >>>>>>> hear Hello Cisco unity connection messaging system from a text tone >>>>>>> phone. >>>>>>> Any one with an idea why this i s happening >>>>>>> >>>>>>> NB: I deleted all the preconfigured voicemail port, huntlist, hunt >>>>>>> group and hunt pilot on the UCM as the gude does not indicate that it is >>>>>>> needed for the integration to wor >>>>>>> >>>>>>> Thanks for the anticipated response >>>>>>> Regards >>>>>>> >>>>>> >>>>>> >>>>> >>>>> ___ >>>>> For more information regarding industry leading CCIE Lab training, >>>>> please visit www.ipexpert.com >>>>> >>>>> >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> Otto Sanchez >>>> CCIE #25592 (Voice) >>>> Support Engineer - IPexpert, Inc. >>>> URL: http://www.IPexpert.com <http://www.ipexpert.com/> >>>> >>> >>> >>> >>> -- >>> Regards, >>> >>> Otto Sanchez >>> CCIE #25592 (Voice) >>> Support Engineer - IPexpert, Inc. >>> URL: http://www.IPexpert.com <http://www.ipexpert.com/> >>> >> >> > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CB Traffic shapping compress rtp headers
0 bytes > 5 minute rate 0 bps > Match: protocol sip > 0 packets, 0 bytes > 5 minute rate 0 bps > Match: protocol skinny > 158 packets, 13676 bytes > 5 minute rate 0 bps > Queueing > queue limit 64 packets > (queue depth/total drops/no-buffer drops) 0/0/0 > (pkts output/bytes output) 158/13676 > bandwidth 16 kbps* > *Class-map: class-default (match-any) > 3972 packets, 238836 bytes > 5 minute offered rate 0 bps, drop rate 0 bps > Match: any > Queueing > queue limit 64 packets > (queue depth/total drops/no-buffer drops/flowdrops) 0/0/0/0 > (pkts output/bytes output) 3972/238836 > Fair-queue: per-flow queue limit 16* > ** > > > -- > Hotmail: Powerful Free email with security by Microsoft. Get it > now.<https://signup.live.com/signup.aspx?id=60969> > -- > Hotmail: Trusted email with powerful SPAM protection. Sign up > now.<https://signup.live.com/signup.aspx?id=60969> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Standard Local Route List/Group ?
Hi Mike, I would get used to the new features and the new way to tackle the dial plan, as they, in many cases will optimize the call routing configurations, lets think about your example, if you would have used cd xform patterns, only one route pattern and rl/ rg would have been used, also think in the case you had not two but twenty sites and 4 route patterns each..., in this case I would have used the new approach instead as my life would definitely be easier, Please don't get me wrong as I'm not telling to forget about using the old approach, just that you should take advantage of the new features and use them wisely when sitting in the lab as in your daily basis job, hth, On Sun, Mar 21, 2010 at 6:16 PM, Mike Brooks <2xcci...@gmail.com> wrote: > Thanks Otto. > > Yes I would not use standard local route groups if I had the option not to, > but am exploring methods if forced to use standard local route groups based > on requirements of the lab. > > For call routing what was your approach... old school ..or new school ? > and why ? > > Thanks, > > Mike Brooks > CCIE#16027 (R&S) > > On Sun, Mar 21, 2010 at 6:34 PM, Otto Sanchez wrote: > >> Hi Mike, >> >> If using old school method, I think you wouldn't be using local route >> group concept as well, right?. In that case you will need two separate RP, >> two different RL/RG, in which case you can perform manipulations in RG >> within RL and send the leading 9 only to the h.323 gw, >> >> If still want to use one RP and the standard local route group RL without >> using cd xform patterns, I think the best way to sort this out for the h.323 >> gw is to perform the manipulation in the incoming dial peer for the h.323 >> like you mentioned as the srst dialing won't be affectedand will be the same >> as when dialing from the ucm, also the dial plan will be simpler, >> >> If you don't add the leading 9 to the incoming dial peer, you will have to >> configure two dial peers set for the same dialing and they will overlap with >> each other, so this option is going to be much more complicated, >> >> hth, >> >> >> On Sun, Mar 21, 2010 at 5:19 PM, Mike Brooks <2xcci...@gmail.com> wrote: >> >>> ahh..yes very good point... I have been moving away from the "new school" >>> routing methods lately because of some of the flexibility issues I have ran >>> into while using it. >>> >>> For instance ANI manipulation based on the type of call. >>> >>> I do prefer sticking to the "old school" routing method if possible, but >>> I will play around with the called party transformations on the gateway. >>> >>> Thanks Randall !! >>> >>> Mike Brooks >>> CCIE#16027 (R&S) >>> >>> >>> >>> >>> >>> On Sun, Mar 21, 2010 at 5:32 PM, Randall Saborio wrote: >>> >>>> Mike (sorry, missed to copy the list on first try), >>>> >>>> >>>> You are missing one option, which is the one I like most. >>>> >>>> For almost all situations, I would go about using Calling Party >>>> Transformations and Called Party Transformations. >>>> >>>> Since these are device specific, you would make a Called Party >>>> Transformation that would be used only by the H323 gateway, and there >>>> prepend a 9. >>>> >>>> Then, on the gateway, you would stick with just one set of dial-peers >>>> which are the same used for SRST. >>>> >>>> On Sun, Mar 21, 2010 at 2:39 PM, Mike Brooks <2xcci...@gmail.com>wrote: >>>> >>>>> Just want to know how most would handle this type of scenario. It >>>>> appears there are multiple ways to configure this. >>>>> >>>>> >>>>> 9.[2-9]XX -> Standard Local RL (must strip predot because HQ is >>>>> mgcp gw) >>>>> >>>>> *HQ Device Pool:* >>>>> Standard Local Route Group: HQ-MGCP-RG >>>>> >>>>> *BR1 Device Pool:* >>>>> Standard Local Route Group: BR1-H323-RG >>>>> >>>>> >>>>> If either an HQ or BR1 phone makes a local call 9.[2-9]XX we must >>>>> strip the 9, regardless of which gateway it goes to because HQs standard >>>>> local route group is an MGCP gateway. >>>>> >>>>> So on the BR1-H323 gateway you have 2 options: >>>>> >>>>> 1. ma
Re: [OSL | CCIE_Voice] SIP Integration of UCM and UC
I meant for the *Out*bound direction, i.e., from ucm to uc, On Sun, Mar 21, 2010 at 5:51 PM, Otto Sanchez wrote: > Hi, > > Did you take a look at this document? > > > http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.html > > > <http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.html>Also, > make sure the Redirecting Diversion Header Delivery - Inbound is checked, > > hth, > > On Fri, Mar 19, 2010 at 1:35 PM, Omotayo wrote: > >> i have been able to get this work. i have checked all doc but no solution >> I still need help on this >> thanks >> >> On Wed, Mar 17, 2010 at 3:29 PM, Omotayo wrote: >> >>> Hello, >>> Any ideas? >>> >>> On Wed, Mar 17, 2010 at 9:54 AM, Omotayo wrote: >>> >>>> Hello All, >>>> >>>> On Lab 7, after integrating the UCM with the UC using SIP. Pressing the >>>> subscriber button, i get the personal greeting message >>>> But, when pstn or a local call dials hq phone 2 or br1 phone 2, i hear >>>> Hello Cisco unity connection messaging system from a text tone >>>> phone. >>>> Any one with an idea why this i s happening >>>> >>>> NB: I deleted all the preconfigured voicemail port, huntlist, hunt group >>>> and hunt pilot on the UCM as the gude does not indicate that it is needed >>>> for the integration to wor >>>> >>>> Thanks for the anticipated response >>>> Regards >>>> >>> >>> >> >> ___ >> For more information regarding industry leading CCIE Lab training, please >> visit www.ipexpert.com >> >> > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Standard Local Route List/Group ?
Hi Mike, If using old school method, I think you wouldn't be using local route group concept as well, right?. In that case you will need two separate RP, two different RL/RG, in which case you can perform manipulations in RG within RL and send the leading 9 only to the h.323 gw, If still want to use one RP and the standard local route group RL without using cd xform patterns, I think the best way to sort this out for the h.323 gw is to perform the manipulation in the incoming dial peer for the h.323 like you mentioned as the srst dialing won't be affectedand will be the same as when dialing from the ucm, also the dial plan will be simpler, If you don't add the leading 9 to the incoming dial peer, you will have to configure two dial peers set for the same dialing and they will overlap with each other, so this option is going to be much more complicated, hth, On Sun, Mar 21, 2010 at 5:19 PM, Mike Brooks <2xcci...@gmail.com> wrote: > ahh..yes very good point... I have been moving away from the "new school" > routing methods lately because of some of the flexibility issues I have ran > into while using it. > > For instance ANI manipulation based on the type of call. > > I do prefer sticking to the "old school" routing method if possible, but I > will play around with the called party transformations on the gateway. > > Thanks Randall !! > > Mike Brooks > CCIE#16027 (R&S) > > > > > > On Sun, Mar 21, 2010 at 5:32 PM, Randall Saborio wrote: > >> Mike (sorry, missed to copy the list on first try), >> >> >> You are missing one option, which is the one I like most. >> >> For almost all situations, I would go about using Calling Party >> Transformations and Called Party Transformations. >> >> Since these are device specific, you would make a Called Party >> Transformation that would be used only by the H323 gateway, and there >> prepend a 9. >> >> Then, on the gateway, you would stick with just one set of dial-peers >> which are the same used for SRST. >> >> On Sun, Mar 21, 2010 at 2:39 PM, Mike Brooks <2xcci...@gmail.com> wrote: >> >>> Just want to know how most would handle this type of scenario. It >>> appears there are multiple ways to configure this. >>> >>> >>> 9.[2-9]XX -> Standard Local RL (must strip predot because HQ is mgcp >>> gw) >>> >>> *HQ Device Pool:* >>> Standard Local Route Group: HQ-MGCP-RG >>> >>> *BR1 Device Pool:* >>> Standard Local Route Group: BR1-H323-RG >>> >>> >>> If either an HQ or BR1 phone makes a local call 9.[2-9]XX we must >>> strip the 9, regardless of which gateway it goes to because HQs standard >>> local route group is an MGCP gateway. >>> >>> So on the BR1-H323 gateway you have 2 options: >>> >>> 1. make 2 sets of dial-peers with a leading "9" (for SRST) and without a >>> leading "9". >>> >>> 2. on the inbound voip dial-peer preprend a "9" to the called number >>> using a translation pattern. >>> >>> >>> Which option seems to be the best choice and why ? Are there any other >>> options ? >>> >>> Thank you, >>> >>> Mike Brooks >>> CCIE#16027 (R&S) >>> >>> >>> >>> >>> >>> >>> ___ >>> For more information regarding industry leading CCIE Lab training, please >>> visit www.ipexpert.com >>> >>> >> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Integration of UCM and UC
Hi, Did you take a look at this document? http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.html <http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.html>Also, make sure the Redirecting Diversion Header Delivery - Inbound is checked, hth, On Fri, Mar 19, 2010 at 1:35 PM, Omotayo wrote: > i have been able to get this work. i have checked all doc but no solution > I still need help on this > thanks > > On Wed, Mar 17, 2010 at 3:29 PM, Omotayo wrote: > >> Hello, >> Any ideas? >> >> On Wed, Mar 17, 2010 at 9:54 AM, Omotayo wrote: >> >>> Hello All, >>> >>> On Lab 7, after integrating the UCM with the UC using SIP. Pressing the >>> subscriber button, i get the personal greeting message >>> But, when pstn or a local call dials hq phone 2 or br1 phone 2, i hear >>> Hello Cisco unity connection messaging system from a text tone >>> phone. >>> Any one with an idea why this i s happening >>> >>> NB: I deleted all the preconfigured voicemail port, huntlist, hunt group >>> and hunt pilot on the UCM as the gude does not indicate that it is needed >>> for the integration to wor >>> >>> Thanks for the anticipated response >>> Regards >>> >> >> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE reset
Hi Iwan, I experienced this before, and applied the same troubleshooting you mentioned, but it was to long to restore the cue functioning. A way I found to get the module back up quickly, was to going to srst mode by shutting down the serial interface between br2 and hq, so, restoring the connection between sites solved my cue issue. I think however, that other ways to get into srst mode would help like setting ucm servers route from br2 to a null interface. Before I encountered the problem, I used to get into srst mode by subscribing br2 devices to a single server ucm group/ device pool and shutting down the call manager service for the corresponding server, hth, On Sun, Mar 21, 2010 at 3:55 PM, Iwan Hoogendoorn wrote: > Hi, > > > > I am having this strange problem that my Cisco Unity Express … is just > suddenly not working anymore… > > Is this a common problem? > > > > My phones went into SRST more and from there CUE was just working fine … > but when coming out of SRST mode the phones registered to the CUCM… CUE > suddenly was not working. > > And this is not the first time that I had this… > > > > The only option here is to reload the CUE and as we all know reloading > takes a lot of time (which we don’t have) and it was ok again… > > > > I Reset the Device Pool, Reset the Voicemail Profile, Reset the CTI > ports, for the CUE in offline and then in online modus again … > > All of these things did not work but reloading works … > > But then again isn’t there another solution? > > > > > > Met vriendelijke groet, > > With kind regards, > > > > ing. Iwan Hoogendoorn, CCIE3 #13084 (R&S, Sec, SP) > > Homepage: http://www.i-1.nl > > Blog: http://blog.i-1.nl > > Twitter: http://www.twitter.com/iwan_ccie > > LinkedIn: http://www.linkedin.com/in/iwanhoogendoorn > > > > _______ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] srst integration with unity connection
My bad!, sorry, I didn't read it correctly, So in that case are you looking for a way to reach the centralized messaging server from the remote location running srst using the pstn?, if that's the case, the following doc might help, http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/admin/srst/configuration/guide/srs_mail.html Thanks, On Sat, Mar 20, 2010 at 10:46 AM, Otto Sanchez wrote: > Hi, > > When working with cue and ucm, you have the option to still use cue with > srst mode, the following document shows guidelines on how to perform this > integration: > > > http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml#srst > > > <http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml#srst>The > mwi functionality should still work when in srst mode, the next document > gives more detail on how to do that, use mwi unsolicited notification method > with srst > > > http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_2/administrator/voicemail/1syscmp_ps5520_TSD_Products_Administration_Guide_Chapter.html#wp1012124 > <http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_2/administrator/voicemail/1syscmp_ps5520_TSD_Products_Administration_Guide_Chapter.html#wp1012124> > Finally, when configuring the cue module using the gui, do not forget to > enable the voicemail application in stst mode and enable the mwi > notification as unsolicited, > > HTH, > > > On Sat, Mar 20, 2010 at 9:10 AM, anupam TYAGI wrote: > >> >> Hi >> >> Can someone guide , how to achieve the voice mail functionlity when in >> srst mode , i am using unity connection 7.x .MWI should also work >> >> Thanks >> >> >> _______ >> For more information regarding industry leading CCIE Lab training, please >> visit www.ipexpert.com >> >> > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phone registration problem - SIP/2.0 401 Unauthorized message
Hi Kalyan, "authenticate register" command is missing from you voice register global configuration, also, be sure that the registrar server and bind commands under sip configuration mode are properly implemented, If still with issues, please send full configuration and sip debugs, Thanks, On Sat, Mar 20, 2010 at 10:00 AM, Kalyan iyer wrote: > Hi guys, > > I am having a hard time registering my home SIP phone with CUCME on BR2 > RTR. > > Weird thing is the phone display has the DN number, I get a dial tone and I > can dial from the phone to the BR2 SCCP Phone. However, I can't receive any > phone calls and the output of the show voice register pool 2 > is > Output of deb ccsip message > > REGISTER sip:10.10.202.1 SIP/2.0 > Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 > From: > >;tag=003094c2f26b663e5919-699325d0 > To: > > Call-ID: 003094c2-f26b-0a633263-5f6ad...@192.168.12.13 > Max-Forwards: 70 > CSeq: 205 REGISTER > User-Agent: Cisco-CP7960G/8.0 > Contact: ;user=phone;transport=udp>;+sip.instance=" BR2-RTR#200>";+u.sip!model.ccm.cisco.com="7" > Supported: > replaces,join,norefersub,X-cisco-callinfo,X-cisco-service-control > Content-Length: 0 > Expires: 3600 > > > Mar 20 14:09:45.663: //-1//SIP/Msg/ccsipDisplayMsg: > Sent: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 > From: > >;tag=003094c2f26b663e5919-699325d0 > To: > > Date: Sat, 20 Mar 2010 14:09:45 GMT > Call-ID: 003094c2-f26b-0a633263-5f6ad...@192.168.12.13 > Server: Cisco-SIPGateway/IOS-12.x > CSeq: 205 REGISTER > Content-Length: 0 > > > Mar 20 14:09:46.163: //-1//SIP/Msg/ccsipDisplayMsg: > Sent: > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 > From: > >;tag=003094c2f26b663e5919-699325d0 > To: >;tag=7DED54-1B43 > Date: Sat, 20 Mar 2010 14:09:45 GMT > Call-ID: 003094c2-f26b-0a633263-5f6ad...@192.168.12.13 > Server: Cisco-SIPGateway/IOS-12.x > CSeq: 205 REGISTER > WWW-Authenticate: Digest > realm="",nonce="94FB834E000C97EE",algorithm=MD5,qop="auth" > Content-Length: 0 > > > Mar 20 14:09:46.259: //-1//SIP/Msg/ccsipDisplayMsg: > Received: > REGISTER sip:10.10.202.1 SIP/2.0 > Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 > From: > >;tag=003094c2f26b663e5919-699325d0 > To: > > Call-ID: 003094c2-f26 > br2-rtr#b-0a633263-5f6ad...@192.168.12.13 > Max-Forwards: 70 > CSeq: 205 REGISTER > User-Agent: Cisco-CP7960G/8.0 > Contact: ;user=phone;transport=udp>;+sip.instance="";+u.sip! > model.ccm.cisco.com="7" > Supported: > replaces,join,norefersub,X-cisco-callinfo,X-cisco-service-control > Content-Length: 0 > Expires: 3600 > > > > > BR2-RTR#show voice register pool 2 > Pool Tag 2 > Config: > Mac address is 0030.94C2.F200 > Type is 7960 > Number list 1 : DN 2 > Proxy Ip address is 0.0.0.0 > DTMF Relay is enabled, rtp-nte > Call Waiting is enabled > DnD is disabled > Description is 32143006 > keep-conference is enabled > username cisco password 123 > template is 1 > service-control mechanism is not supported > > Dialpeers created: > > Statistics: > Active registrations : 0 > > Total SIP phones registered: 0 > Total Registration Statistics > Registration requests : 0 > Registration success : 0 > Registration failed : 0 > unRegister requests : 0 > unRegister success : 0 > unRegister failed : 0 > > > SHOW RUN Config from the BR2 RTR is > voice register global > mode cme > source-address 10.10.202.1 port 5060 > max-dn 2 > max-pool 2 > load 7960-7940 P0S3-08-6-00 > timezone 13 > time-format 24 > date-format D/M/Y > voicemail 3600 > tftp-path flash: > create profile sync 0001011653248495 > ntp-server 10.10.100.2 mode unicast > ! > voice register dn 1 > number 3005 > name br2 phn 3 > ! > voice register dn 2 > number 3006 > name br2 phn 4 > ! > voice register template 1 > dialplan 1 > no conference enable > ! > voice register dialplan 1 > type 7940-7960-others > pattern 1 3... > pattern 2 999 > ! > voice register pool 1 > id mac 0011.BBEF.6FB9 > type 7960 > number 1 dn 1 > template 1 > dtmf-relay rtp-nte > username 3005 password cisco > description 32143005 > codec g711ulaw > ! > voice register pool 2 > id mac 0030.94C2.F200 > type 7960 > number 1 dn 2 > template 1 > username cisco password 123 > description 32143006 > codec g711ulaw > > > I also removed the username and password and tr
Re: [OSL | CCIE_Voice] Translation Pattern: Negate a range
Hi Angel, It should work. Just curious, have you tried to do it not using the caret character?, i.e. \+[3-9]!, Just to understand a bit what you are trying to achieve, and according to the pattern you sent, rephrasing your sentence, is your requirement?: "I want to match all pattern that begin with + and then any number except 1 *OR* 2" Thanks, On Fri, Mar 19, 2010 at 7:00 AM, Angel Perez wrote: > Hi: > > I wan't to negate a range of numbers in a tranlation pattern like in a > router pattern, but I think that this is not possible after doing some > tests. > > For example, I want to match all pattern that begin with + and then any > number except 1 and 2 > > \+[^1-2]! > > In a router pattern this would match, but in a translation pattern do > not... > > Any suggestions? Thanks > > -- > Hotmail: Powerful Free email with security by Microsoft. Get it > now.<https://signup.live.com/signup.aspx?id=60969> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] srst integration with unity connection
Hi, When working with cue and ucm, you have the option to still use cue with srst mode, the following document shows guidelines on how to perform this integration: http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml#srst <http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml#srst>The mwi functionality should still work when in srst mode, the next document gives more detail on how to do that, use mwi unsolicited notification method with srst http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_2/administrator/voicemail/1syscmp_ps5520_TSD_Products_Administration_Guide_Chapter.html#wp1012124 <http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_2/administrator/voicemail/1syscmp_ps5520_TSD_Products_Administration_Guide_Chapter.html#wp1012124> Finally, when configuring the cue module using the gui, do not forget to enable the voicemail application in stst mode and enable the mwi notification as unsolicited, HTH, On Sat, Mar 20, 2010 at 9:10 AM, anupam TYAGI wrote: > > Hi > > Can someone guide , how to achieve the voice mail functionlity when in srst > mode , i am using unity connection 7.x .MWI should also work > > Thanks > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE Voice Vol 2 Lab 5 Task 4.1
Hello, Have you tried calling the vm using only g.711?, in that way you can discard whether it is a problem with vm integration itself or a transcoding issue. Also, are your cti route points and ports registered?, does the issue only happens when calling from remotes sites?, is the transcoding being invoked?, are there any qos policies in place?, Thanks, On Fri, Mar 19, 2010 at 10:39 PM, CCIETalk.com wrote: > wow am I the only one having this issue? cue is registered, transcoder is > registered so I am not sure what I might be missing unless PG missed > something. > > > On Fri, Mar 19, 2010 at 10:12 PM, CCIETalk.com wrote: > >> I followed the PG on this one to the last dot but I cannot get the calls >> to ring the VM. After completing all the tasks, PG instructs to dial from HQ >> or BR1 and call fwd to BR2 but I get a busy signal. >> >> I can see that my CUE is registered to CUCM. >> >> Any clues? >> >> -- >> www.ccietalk.com >> > > > > -- > www.ccietalk.com > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RDNIS ISSUE WITH LAB1 HA (10.2)
Hi Sean, Did you make sure the isdn outgoing ie redirecting-number command is present in the voice serial interface?, how does the br1 debug isdn q931 look like?, what is getting to the vm?, did you take a look at the rtmt for uc? Thanks, On Wed, Mar 17, 2010 at 9:07 AM, sean hurricane wrote: > > > > >> Question: Ensure if nobody within huntgroup is available to answer call, >> the caller hears subscriber greeting for br1-phn2 >> >> Config below: >> >> telephony-service >> srst mode auto-provision all >> srst dn line-mode dual >> em logout 0:0 0:0 0:0 >> max-ephones 2 >> max-dn 10 preference 8 >> ip source-address 10.10.201.1 port 2000 >> time-zone 12 >> voicemail 5600 >> max-conferences 8 gain -6 >> moh music-on-hold.au >> multicast moh 239.1.1.1 port 16384 route 10.10.110.2 10.10.201.1 >> transfer-system full-consult >> create cnf-files version-stamp 7960 Mar 14 2010 13:16:05 >> >> voice hunt-group 1 peer >> final 5600 >> list 1001,1002 >> timeout 12 >> pilot 1000 >> >> dial-peer voice 30 pots >> destination-pattern 5600 >> port 0/2/0:23 >> forward-digits 11 >> prefix 1212394 >> >> RDNIS inbound is checked on HQ gw and RDNIS outbound checked on BR1 gw >> >> Alternate extension with dn 1000 is checked in UC for user mailbox 1002 >> and also 6178631000 just in case >> >> Caller hears UC opening greeting instead of mailbox 1000 greeting, it >> works if i use call-manager fallback, but the requirement for hunt-group >> with not be met if call-manager fallback is used. >> >> My question is does telephony-system send RDNIS when configured in srst >> mode? >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Sun, Mar 14, 2010 at 4:41 PM, >> wrote: >> >>> Send CCIE_Voice mailing list submissions to >>>ccie_voice@onlinestudylist.com >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>>http://onlinestudylist.com/mailman/listinfo/ccie_voice >>> or, via email, send a message with subject or body 'help' to >>>ccie_voice-requ...@onlinestudylist.com >>> >>> You can reach the person managing the list at >>>ccie_voice-ow...@onlinestudylist.com >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of CCIE_Voice digest..." >>> >>> >>> Today's Topics: >>> >>> 1. QoS Calculation Value for L2 MLPoFR (Berry, Matthew J.) >>> 2. Re: Proctorlabs PSTN Router As A GK (CCIETalk.com) >>> 3. Re: Proctorlabs PSTN Router As A GK (scott carruthers) >>> 4. Re: Proctorlabs PSTN Router As A GK (CCIETalk.com) >>> 5. UCM with SIP account Configuration Implementation (Saeed IDris) >>> 6. Re: SIP Hardware Transcoder (Otto Sanchez) >>> 7. Re: MVA (Otto Sanchez) >>> >>> >>> -- >>> >>> Message: 1 >>> Date: Sun, 14 Mar 2010 11:13:49 -0500 >>> From: "Berry, Matthew J." >>> Subject: [OSL | CCIE_Voice] QoS Calculation Value for L2 MLPoFR >>> To: "CCIE Voice OSL (ccie_voice@onlinestudylist.com)" >>>, Vik Malhi >>> Message-ID: >>>< >>> 0d69df41cdb9bf4bb30c481d414a8be70af63b0...@usepx2pmxmbx04.corp.kroll.com >>> > >>> >>> Content-Type: text/plain; charset="us-ascii" >>> >>> Working on Vol 1 Lab 10A, Question 10.4 >>> >>> The Proctor Guide calculates L2 MLPoFR as 9 bytes per packet. However, >>> the QoS SRND defines the following on page 1-15: >>> - PPP = 12 bytes >>> - MLP = 13 bytes >>> - FR = 4 bytes >>> - FR with FRF.12 = 8 bytes >>> >>> None of those match up. Why did IPexpert chose 9 bytes per packet? >>> >>> Matthew Berry >>> >>> Digital Footprint: >>> Twitter: ciscovoiceguru >>> Skype: ciscovoiceguru >>> 1st Lab Attempt: Aug 16th, 2010 >>> >>> -- >>> >>> Message: 2 >>> Date: Sun, 14 Mar 2010 11:17:56 -0500 >>> From: "CCIETalk.com" >>> Subject: Re: [OSL | CCIE_Voice] Proctorlabs PSTN Router As A GK >>> To: scott carruthers , >>>ccie_voice@onlinestudylist.com >>>
Re: [OSL | CCIE_Voice] GlobalizationAndLocalization Issue
Hi Naoufal, Please stay away from cupc when testing globalization and localization, this is not supported for this endpoint, Thanks, On Tue, Mar 16, 2010 at 9:37 PM, Kevin Damisch wrote: > What version of IP Blue are you using? A couple of weeks ago, I downloaded > the new version of IP Blue and was able to get + dialing to work just fine. > This was using GNS3 too. I set the incoming calling party settings to > prefix +1212 on subscriber calls (for example), and it showed +12123942123 > in the missed calls and was able to dial from there too. Plus (pun > intended), I could manually dial +12123942123 within IP Blue and it called > the PSTN phone. > > Kevin > > > -Original Message- > From: ccie_voice-boun...@onlinestudylist.com [mailto: > ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego > Sent: Tuesday, March 16, 2010 8:53 PM > To: kerboute kerboute > Cc: CCIE Voice Maillist > Subject: Re: [OSL | CCIE_Voice] GlobalizationAndLocalization Issue > > Correct me if im wrong but Those phones doesn't support + dialing. I > had the same problem that you had and i replaced the ipc with a 41 > and ot worked great. You need to try it on a 7941 and it will work. > Did you add the translation pattern to call back? > > Sent from my iPhone > > On Mar 16, 2010, at 5:47 PM, kerboute kerboute > wrote: > > > Hi guys, > > > > Actually I'm working on LAB5A Vol1, and I have some problems related > > to > > globalization and localization. > > Just for Information, my voice lab using GNS3 routers and vmware for > > CUCM ..., Phones (IP Blue, xlite and CIPC 7) > > My problem related to question 5.3, I setup everything for > > globalization > > and when i call from PSTN to HQ phone 2 i can see the calling number > > as > > +12123942123 type subscriber. After localization configuration i can > > see > > the calling number as a local call 3942123 but when i cheked the > > history > > (Missed calls) i cannot see the global number as +12123942123, what i > > see is the local number 3942123. > > > > Any Ideas?? > > > > I have another question; Is Cisco IP Communicator support + dialing? > > (i > > tested but not working I've got reorder) > > > > > > Regards > > Naoufal > > ___ > > For more information regarding industry leading CCIE Lab training, > > please visit www.ipexpert.com > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > This communication (including any attachments) is intended only for the use > of the individual or entity to which it is addressed, and may contain > information that is privileged, confidential and exempt from disclosure > under applicable law. If you are not the intended recipient, any > dissemination, distribution or copying of this communication is strictly > prohibited. If you have received this communication in error, please notify > Vital Support Systems at 515 334 5700 and delete or destroy all copies and > the original document. > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] QoS SRND - Page 105
Correct Mike, The audio stream between phones is not maintained when the holder phone press the hold softkey, in that moment an audio stream is sent from the holdee moh server to the holdee itself, when the resume sofkey is pressed, the audio stream is reestablished between the two original endpoints, On Thu, Mar 18, 2010 at 6:51 AM, Mike Thompson wrote: > Damn good question!! My thought is that the call on hold will be > 'termiated' at the call control and not to the phone. > > So while a call is on hold, the bandwidth will be near 0 to the phone. > > > > Sent from my phone, apologies for any typos. > > On Mar 18, 2010, at 6:17 AM, "Berry, Matthew J." > wrote: > > Pulling from the QoS SRND, the following configuration is only supposed to > allow the bandwidth for one voice call per switchport VLAN. Obviously, > based on the 128k, we're focused on G.711 calls (so my next question will > not apply to G.729). > > I want to know if the following command would disable the ability to have > multiple calls (different lines) on the same phone. For example: Phone A > (with the policing command below) calls Phone B. At this point, 128k of > G.711 bandwidth is consumed. If Phone A puts Phone B on hold and calls > Phone C, would the call no go through due to policing? > > CAT2970(config-cmap)#policy-map IPPHONE+PC-BASIC > CAT2970(config-pmap)#class VVLAN-VOICE > CAT2970(config-pmap-c)# set ip dscp 46 ! DSCP EF (Voice) > CAT2970(config-pmap-c)# police 128000 8000 exceed-action drop > > I guess what I am asking is what happens to the initial call when it is > placed on hold? Is the audio stream maintained between phones (128k), > thereby eliminating the ability to place another call? > > Matthew Berry > > Digital Footprint: > Twitter: ciscovoiceguru > Skype: ciscovoiceguru > 1st Lab Attempt: Aug 16th, 2010 > > ___ > For more information regarding industry leading CCIE Lab training, please > visit <http://www.ipexpert.com>www.ipexpert.com > > > _______ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Need Clarification: mls qos map cos-dscp
Hello Matthew, Mike, The Cisco recommendation for marking signaling traffic has changed from dscp 26 to dscp 24 for call signalling in a production network, so I would do that consistently over the network, however in the lab, you should do exactly what they are asking you to, so should they ask you to apply best marking recommendations from the access layer you know what to do, otherwise just ask the proctor. More information at: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/netstruc.html#wp1044051 Please let me know if that answer your question, Thanks, On Thu, Mar 18, 2010 at 6:57 AM, Mike Thompson wrote: > This has been a topic of conversation at work between a few of us, so I'm > interested in the feedback. > > > Sent from my phone, apologies for any typos. > > On Mar 18, 2010, at 5:55 AM, "Berry, Matthew J." > wrote: > > The QoS SRND states that the "auto qos voip" command adds the following > config to the router: > > C2970(config)# mls qos > C2970(config)# mls qos map cos-dscp 0 8 16 *26* 32 46 48 56 > > Earlier in the SRND, around page 40, it says that the old marking for audio > signaling was AF31 (26). That is the same DSCP marking listed above. > > As part of our "best-practice" scenario, should we be changing the command > to consider audio signaling as CS3 (24)? The command would need to be > modified: > > C2970(config)# mls qos map cos-dscp 0 8 16 *24 *32 46 48 56 > > Is this true? Otto, can you weigh in on this one? > > Thanks! > > Matthew Berry > > Digital Footprint: > Twitter: ciscovoiceguru > Skype: ciscovoiceguru > 1st Lab Attempt: Aug 16th, 2010 > > ___ > > For more information regarding industry leading CCIE Lab training, please > visit <http://www.ipexpert.com>www.ipexpert.com > > > _______ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] media resources query
Hi, MOH server should be placed in your case in site A, as this is where the MOH server is located. Phones from remote locations will still be using the site A moh service using the codec defined in the region relationship between the phone and moh server, which in most cases will be g.729 if you are using unicast or multicast moh over the wan, On Tue, Mar 16, 2010 at 11:20 AM, anupam TYAGI wrote: > > > guys , > > i am configuring my media resources like moh server , transcoder and > hardware conference bridges etc . > i need to mention the device pool for these resources . i have device pool > DP_A , DP_B,DP_C for three sites A, B & C . > Phones at different sites are assigned to their site device pool .MOH > server is at site A and transcoder and hardware conference > can be at any site according to the usage . I will be putting these media > resources in media resources group and these assigned to mrgl . > My MOH server will be used by all three sites IP phones .So in which > device pool should i put my MOH server , so that all the ip phone can use > the MOH server . > > > Can someone clarify on this > > > thanks > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME busy-trigger-per-button
no supplementary-service sip refer > > sip > > bind control source-interface Loopback0 > > bind media source-interface Loopback0 > > registrar server expires max 600 min 60 > > > > > > > > > > > > > > > > voice register global > > mode cme > > source-address 10.10.110.3 port 5060 > > max-dn 3 > > max-pool 6 > > authenticate register > > mwi reg-e164 > > voicemail 3600 > > tftp-path flash: > > create profile sync 0006855418337003 > > ! > > voice register dn 1 > > number 3002 > > call-forward b2bua busy 3600 > > call-forward b2bua mailbox 3002 > > call-forward b2bua noan 3600 timeout 12 > > name br2 phone 2 > > no-reg > > label br2 phone 2 > > mwi > > ! > > voice register dn 2 > > number 3003 > > call-forward b2bua busy 3600 > > call-forward b2bua mailbox 3003 > > call-forward b2bua noan 3600 timeout 12 > > name br2 phone 3 > > no-reg > > label br2 phone 3 > > mwi > > ! > > voice register pool 1 > > id mac .. > > type 7941 > > number 1 dn 1 > > dtmf-relay rtp-nte > > username 3002 password cisco > > ! > > voice register pool 2 > > id mac 001F.6C7E.D6FE > > type 7941 > > number 1 dn 2 > > dtmf-relay rtp-nte > > username 3003 password cisco > > > > > > > > > > > > dial-peer voice 200 voip > > max-conn 1 > > destination-pattern 3600 > > session protocol sipv2 > > session target ipv4:10.10.210.13 > > dtmf-relay rtp-nte > > codec g711ulaw > > ! > > ! > > > > > > > > telephony-service > > no auto-reg-ephone > > em logout 0:0 0:0 0:0 > > max-ephones 8 > > max-dn 8 > > ip source-address 10.10.202.1 port 2000 > > voicemail 3600 > > mwi relay > > max-conferences 8 gain -6 > > transfer-system full-consult > > transfer-pattern .T > > create cnf-files version-stamp 7960 Mar 10 2010 15:22:39 > > ! > > ! > > ephone-dn 1 dual-line > > number 3001 no-reg primary > > label Br2 pHone 1 > > name Br2 Phone 1 > > call-forward busy 3600 > > call-forward noan 3600 timeout 12 > > ! > > ! > > > > sip-ua > > mwi-server ipv4:10.10.210.13 expires 3600 port 5060 transport udp > > unsolicited > > > > ! > > ! > > ephone 1 > > device-security-mode none > > mac-address 001E.EC15.996D > > type CIPC > > button 1:1 > > ! > > > > > > > > Thanks for the anticipated support > > > > > > > -- next part -- > An HTML attachment was scrubbed... > URL: > http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100312/76526596/attachment-0001.htm > > -- > > Message: 2 > Date: Fri, 12 Mar 2010 14:33:41 -0800 > From: Jeff Cotter > Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder > To: Omotayo , Otto Sanchez > Cc: "ccie_voice@onlinestudylist.com" > Message-ID: <54cc1bd3093b6e41b86926c1657432f187a06...@ssfex1> > Content-Type: text/plain; charset="us-ascii" > > FYI, I was only able to get this to work using transcoder on CME. Had to > match the codec between UCM trunk and incoming dial-peer on CME...then > xcoder would engage on CME for the SIP phone. I have a hardware limitation > in my home lab so I am not able to configure a xcoder on both UCM and CME > simultaneously. > > > > > From: Omotayo [mailto:adefilabi...@gmail.com] > Sent: Friday, March 12, 2010 6:33 AM > To: Otto Sanchez > Cc: Jeff Cotter; ccie_voice@onlinestudylist.com > Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder > > Hello Otto, > > i had same issue > > The transcoder can be on the trunk? > > When i did the transcoder on the br2 router, i get a busy tone when the sip > phone is being called from the hq phone > > REgards > On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.com>> wrote: > Hi Jeff, > > Would you please tell us more about the call flow and the end to end codec > requirements for this call. If doing g.729 over the wan, and your sip phone > is using g.711 you should transcode at br2, > > Please let us know, > > Thanks, > On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com>> wrote: > Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on > UCM. Can't seem to get a call from Call Manager to CME sip phone wor
Re: [OSL | CCIE_Voice] Fw: Vol1 Lab 5c - 5.12
Hi Kalyan, Did you make sure the Enable Mobile Voice Access service parameter is set to enable and that the end user has the Enable Mobile Voice Access checked? Thanks, On Mon, Mar 15, 2010 at 10:45 AM, wrote: > > Hi all, > > I was working on lab 5c -task 5.12(MVA). > > After I set-up the MVA, when I call 2123945999 and authenticate with the RD > and password, > I get only two options, 2 to enable Mobile voice connect, 3 to disable > Mobile voice connect. > I thought I should also get option 1 to call someone out from that number > to make it appear as if I called from that DN. > > Can anyone tell me why I did not receive that option? > > Thanks > Kalyan > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA
Would you please post your gw config? thanks, On Sun, Mar 14, 2010 at 11:13 PM, anupam TYAGI wrote: > it is configured , i am able to reach MVA prompts , but when i am dialing > the external when in MVA my call disconnects > > > On Mon, Mar 15, 2010 at 8:37 AM, Otto Sanchez wrote: > >> Hi, >> >> I'm referring to the mva number you have configured in the ucm->media >> resources->mobile voice access config, do you have a voip dial-peer matching >> this number as destination and pointing to the ucm? >> >> >> On Sun, Mar 14, 2010 at 10:33 PM, anupam TYAGI wrote: >> >>> I have the dial-peer on the gateway which matches the external number >>> when i am dialing through MVA . >>> >>> i am able to dial this external number when i am not in MVA >>> >>> >>> On Mon, Mar 15, 2010 at 2:11 AM, Otto Sanchez wrote: >>> >>>> Hi, >>>> >>>> Did you make sure that you have a dial peer in the hq rtr which >>>> destination number match the mva number you have in the ucm configuration?, >>>> this is needed to route calls outbound calls from the remote devices >>>> connected to the mva service, >>>> >>>> Thanks, >>>> >>>> On Fri, Mar 12, 2010 at 9:25 PM, anupam TYAGI wrote: >>>> >>>>> i have the route pattern partion assigned to the CSS and this CSS is >>>>> assigned to RDP >but still the call disconnect when i dial the external >>>>> number in MVZ >>>>> >>>>> On Fri, Mar 12, 2010 at 11:06 PM, Omotayo wrote: >>>>> >>>>>> Hello, >>>>>> >>>>>> Berry is right. >>>>>> >>>>>> create a partition called pt-mva >>>>>> >>>>>> crease a CSS called css-mva >>>>>> >>>>>> put the partition in the css >>>>>> >>>>>> create a route pattern like 9.011! in partition pt-mva. the gateway >>>>>> can be the hq gateway if you wish >>>>>> discard predot >>>>>> >>>>>> assign the css to the remote destination profile >>>>>> >>>>>> this will work for you >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI >>>>>> wrote: >>>>>> >>>>>>> if i dial that external number without MVA it goes through ,but when >>>>>>> in MVA i get a disconnect when calling this external number ( so don't >>>>>>> seems >>>>>>> to be codec issue ) >>>>>>> >>>>>>> >>>>>>> On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer < >>>>>>> myciscov...@gmail.com> wrote: >>>>>>> >>>>>>>> are you maybe calling to a remote location (g.729) and therefore a >>>>>>>> xcoder is required, but not set up correctly? >>>>>>>> >>>>>>>> 2010/3/12 anupam TYAGI >>>>>>>> >>>>>>>>> i saw the call hit the gateway . RDP is having the same CSS as >>>>>>>>> phone CSS >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. < >>>>>>>>> mjbe...@krollontrack.com> wrote: >>>>>>>>> >>>>>>>>>> Check the CSS on the remote destination profile you’re calling >>>>>>>>>> from. >>>>>>>>>> >>>>>>>>>> If you do a “debug isdn q931” on the PSTN gateway, do you see the >>>>>>>>>> call hit the gateway? >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Your rerouting CSS on the RDP is used for calls out to your RD. >>>>>>>>>> >>>>>>>>>> Your CSS on the RDP is used for calls through MVA that are routed >>>>>>>>>> out through your PSTN gateway. >>>>>>>>>> >>>>>>>>>> >>>>&g
Re: [OSL | CCIE_Voice] MVA
Hi, I'm referring to the mva number you have configured in the ucm->media resources->mobile voice access config, do you have a voip dial-peer matching this number as destination and pointing to the ucm? On Sun, Mar 14, 2010 at 10:33 PM, anupam TYAGI wrote: > I have the dial-peer on the gateway which matches the external number when > i am dialing through MVA . > > i am able to dial this external number when i am not in MVA > > > On Mon, Mar 15, 2010 at 2:11 AM, Otto Sanchez wrote: > >> Hi, >> >> Did you make sure that you have a dial peer in the hq rtr which >> destination number match the mva number you have in the ucm configuration?, >> this is needed to route calls outbound calls from the remote devices >> connected to the mva service, >> >> Thanks, >> >> On Fri, Mar 12, 2010 at 9:25 PM, anupam TYAGI wrote: >> >>> i have the route pattern partion assigned to the CSS and this CSS is >>> assigned to RDP >but still the call disconnect when i dial the external >>> number in MVZ >>> >>> On Fri, Mar 12, 2010 at 11:06 PM, Omotayo wrote: >>> >>>> Hello, >>>> >>>> Berry is right. >>>> >>>> create a partition called pt-mva >>>> >>>> crease a CSS called css-mva >>>> >>>> put the partition in the css >>>> >>>> create a route pattern like 9.011! in partition pt-mva. the gateway can >>>> be the hq gateway if you wish >>>> discard predot >>>> >>>> assign the css to the remote destination profile >>>> >>>> this will work for you >>>> >>>> >>>> >>>> >>>>On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI wrote: >>>> >>>>> if i dial that external number without MVA it goes through ,but when in >>>>> MVA i get a disconnect when calling this external number ( so don't seems >>>>> to >>>>> be codec issue ) >>>>> >>>>> >>>>> On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer < >>>>> myciscov...@gmail.com> wrote: >>>>> >>>>>> are you maybe calling to a remote location (g.729) and therefore a >>>>>> xcoder is required, but not set up correctly? >>>>>> >>>>>> 2010/3/12 anupam TYAGI >>>>>> >>>>>>> i saw the call hit the gateway . RDP is having the same CSS as >>>>>>> phone CSS >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. < >>>>>>> mjbe...@krollontrack.com> wrote: >>>>>>> >>>>>>>> Check the CSS on the remote destination profile you’re calling >>>>>>>> from. >>>>>>>> >>>>>>>> If you do a “debug isdn q931” on the PSTN gateway, do you see the >>>>>>>> call hit the gateway? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Your rerouting CSS on the RDP is used for calls out to your RD. >>>>>>>> >>>>>>>> Your CSS on the RDP is used for calls through MVA that are routed >>>>>>>> out through your PSTN gateway. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto: >>>>>>>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI >>>>>>>> *Sent:* Friday, March 12, 2010 9:27 AM >>>>>>>> *To:* ccie_voice-requ...@onlinestudylist.com; >>>>>>>> ccie_voice@onlinestudylist.com >>>>>>>> *Subject:* [OSL | CCIE_Voice] MVA >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Hi Folks >>>>>>>> >>>>>>>> I am doing MVA , When i dial the MVA number , I am able to hear >>>>>>>> the prompt. I dial a PSTN number , but the call disconnect . Can >>>>>>>> any body >>>>>>>> suggest me what can be the reason . >>>>>>>> >>>>>>>> >>>>>>>> Rgds >>>>>>>> Anu. >>>>>>>> >>>>>>> >>>>>>> >>>>>>> ___ >>>>>>> For more information regarding industry leading CCIE Lab training, >>>>>>> please visit www.ipexpert.com >>>>>>> >>>>>>> >>>>>> >>>>> >>>>> ___ >>>>> For more information regarding industry leading CCIE Lab training, >>>>> please visit www.ipexpert.com >>>>> >>>>> >>>> >>> >>> ___ >>> For more information regarding industry leading CCIE Lab training, please >>> visit www.ipexpert.com >>> >>> >> >> >> -- >> Regards, >> >> Otto Sanchez >> CCIE #25592 (Voice) >> Support Engineer - IPexpert, Inc. >> URL: http://www.IPexpert.com <http://www.ipexpert.com/> >> > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1
Hi Jason, Thanks for the information, what router/ interfaces are you using? Thanks! On Fri, Mar 12, 2010 at 1:04 PM, Jason Granat wrote: > Hi Otto, > > > > Thanks for the advice. In your second paragraph the opposite was actually > the case. The E1 voice-ports were originally showing a-law, and had > distortion. I hard set u-law on the E1 ports between the gateway and PSTN > router and the distortion went away. Perhaps that is what you meant? > > > > I took a look at the link you included. I’ll have to do some testing but my > main question is how is this handled in the real world at the provider > level? > > > > Thanks, > > > > Jason > > > > *From:* Otto Sanchez [mailto:o...@ipexpert.com] > *Sent:* Friday, March 12, 2010 4:59 AM > *To:* Jason Granat > *Cc:* > *Subject:* Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1 > > > > Hello Jason, > > E1's and T1's will always use a-law and u-law companding mechanism > respectively, this is used to give more "resolution" to low voice > frequencies when digitizing an analog signal (the mechanism is also used in > the other end for digital to analogue conversion), each mechanism is > designed exclusively to work with its voice digital standard and cannot be > used conversely, > > In that sense, my guess is that before applying that command in your E1 > port, the companding type was u-law, you can verify this using the sh voice > port command (perhaps the default configuration of a-law was somehow > overwritten by a cptone command in the same port configuration), and when > you hardcoded the a-law companding type everything worked as expected, > > I also found a note in the Cisco IOS Voice Port Configuration Guide, which > says that the command is used when cross-connecting in a local router, > > > http://www.cisco.com/en/US/partner/docs/ios/voice/voiceport/configuration/guide/vp_cfg_digital_vps_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1009871 > > > HTH, > > On Thu, Mar 11, 2010 at 2:37 PM, Jason Granat wrote: > > So I’ve got this partially figured out. It had to do with the compand-type. > E1 was a-law and T1 was u-law. I set the E1 side for u-law and it sounds > correct now. > > > > The final thing I am trying to figure out is how to ‘trans-compand’ (if > that is the correct term) on the PSTN gateway. As it sits I had to change > the compand-type between the PSTN and E1 gateway. I don’t have experience > with foreign connectivity so maybe this is the way it is done in the real > world but I am thinking that perhaps the E1 site may not want or be able to > change their compand-type, so can it be changed at the PSTN level between > a-law and u-law locations? > > > > Thanks, > > > > Jason > > > > *From:* Jason Granat > > *Sent:* Thursday, March 11, 2010 9:46 AM > *To:* > > *Subject:* PSTN Call Distortion Between T1/E1 > > > > Perhaps this is something simple that I am overlooking but I have the > generic setup running in my home lab with 3 gateways and one PSTN router. 2 > of the gateways are T1 and one is E1. The PSTN router is also running CME > with a 7960 to simulate PSTN destinations. Calls from any site to the PSTN > phone are fine. Calls between T1 sites are fine. Calls between T1 and E1 > sites are distorted, like the gain is way too high. I tried playing with the > gain on the voice-port but no luck. I’m not finding much online or in Cisco > docs. Any suggestions? > > > > Thanks, > > > > Jason > > > -- > > > > http://slash128.com > > > _______ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com <http://www.ipexpert.com/> > > -- > > > http://slash128.com > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA
Hi, Did you make sure that you have a dial peer in the hq rtr which destination number match the mva number you have in the ucm configuration?, this is needed to route calls outbound calls from the remote devices connected to the mva service, Thanks, On Fri, Mar 12, 2010 at 9:25 PM, anupam TYAGI wrote: > i have the route pattern partion assigned to the CSS and this CSS is > assigned to RDP >but still the call disconnect when i dial the external > number in MVZ > > On Fri, Mar 12, 2010 at 11:06 PM, Omotayo wrote: > >> Hello, >> >> Berry is right. >> >> create a partition called pt-mva >> >> crease a CSS called css-mva >> >> put the partition in the css >> >> create a route pattern like 9.011! in partition pt-mva. the gateway can be >> the hq gateway if you wish >> discard predot >> >> assign the css to the remote destination profile >> >> this will work for you >> >> >> >> >>On Fri, Mar 12, 2010 at 6:22 PM, anupam TYAGI wrote: >> >>> if i dial that external number without MVA it goes through ,but when in >>> MVA i get a disconnect when calling this external number ( so don't seems to >>> be codec issue ) >>> >>> >>> On Fri, Mar 12, 2010 at 10:30 PM, Patrick Fischer >> > wrote: >>> >>>> are you maybe calling to a remote location (g.729) and therefore a >>>> xcoder is required, but not set up correctly? >>>> >>>> 2010/3/12 anupam TYAGI >>>> >>>>> i saw the call hit the gateway . RDP is having the same CSS as phone >>>>> CSS >>>>> >>>>> >>>>> >>>>> On Fri, Mar 12, 2010 at 9:08 PM, Berry, Matthew J. < >>>>> mjbe...@krollontrack.com> wrote: >>>>> >>>>>> Check the CSS on the remote destination profile you’re calling >>>>>> from. >>>>>> >>>>>> If you do a “debug isdn q931” on the PSTN gateway, do you see the call >>>>>> hit the gateway? >>>>>> >>>>>> >>>>>> >>>>>> Your rerouting CSS on the RDP is used for calls out to your RD. >>>>>> >>>>>> Your CSS on the RDP is used for calls through MVA that are routed out >>>>>> through your PSTN gateway. >>>>>> >>>>>> >>>>>> >>>>>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto: >>>>>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anupam TYAGI >>>>>> *Sent:* Friday, March 12, 2010 9:27 AM >>>>>> *To:* ccie_voice-requ...@onlinestudylist.com; >>>>>> ccie_voice@onlinestudylist.com >>>>>> *Subject:* [OSL | CCIE_Voice] MVA >>>>>> >>>>>> >>>>>> >>>>>> Hi Folks >>>>>> >>>>>> I am doing MVA , When i dial the MVA number , I am able to hear the >>>>>> prompt. I dial a PSTN number , but the call disconnect . Can any body >>>>>> suggest me what can be the reason . >>>>>> >>>>>> >>>>>> Rgds >>>>>> Anu. >>>>>> >>>>> >>>>> >>>>> ___ >>>>> For more information regarding industry leading CCIE Lab training, >>>>> please visit www.ipexpert.com >>>>> >>>>> >>>> >>> >>> ___ >>> For more information regarding industry leading CCIE Lab training, please >>> visit www.ipexpert.com >>> >>> >> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hi, I think you are referring to the lab5c task 5.2, have you take a look at the pg solution?, there's a very good explanation there on how the hq resources are invoked, but always keep in mind that the transcoder resources are invoked where the codec mismatch occurs, Take a look and let us know, Thanks, On Fri, Mar 12, 2010 at 8:34 PM, Omotayo wrote: > Hello Jeff, > > All calls worked when i configure the xcoder on the cme > > The question says use the hq router resources- that is where i have issues > > thanks > > On Fri, Mar 12, 2010 at 11:33 PM, Jeff Cotter wrote: > >> FYI, I was only able to get this to work using transcoder on CME. Had >> to match the codec between UCM trunk and incoming dial-peer on CME…then >> xcoder would engage on CME for the SIP phone. I have a hardware limitation >> in my home lab so I am not able to configure a xcoder on both UCM and CME >> simultaneously. >> >> >> >> >> >> >> >> >> >> *From:* Omotayo [mailto:adefilabi...@gmail.com] >> *Sent:* Friday, March 12, 2010 6:33 AM >> *To:* Otto Sanchez >> *Cc:* Jeff Cotter; ccie_voice@onlinestudylist.com >> >> *Subject:* Re: [OSL | CCIE_Voice] SIP Hardware Transcoder >> >> >> >> Hello Otto, >> >> >> >> i had same issue >> >> >> >> The transcoder can be on the trunk? >> >> >> >> When i did the transcoder on the br2 router, i get a busy tone when the >> sip phone is being called from the hq phone >> >> >> >> REgards >> >> On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez wrote: >> >> Hi Jeff, >> >> Would you please tell us more about the call flow and the end to end codec >> requirements for this call. If doing g.729 over the wan, and your sip phone >> is using g.711 you should transcode at br2, >> >> Please let us know, >> >> Thanks, >> >> On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter wrote: >> >> Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder >> on UCM. Can’t seem to get a call from Call Manager to CME sip phone >> working. I can call from CME to UCM but not the other way around. Rings but >> disconnects when answered. Transcoder shows registered in Call manager. >> Thanks >> >> >> >> >> >> Jeff >> >> >> >> ___ >> For more information regarding industry leading CCIE Lab training, please >> visit www.ipexpert.com >> >> >> >> >> -- >> Regards, >> >> Otto Sanchez >> CCIE #25592 (Voice) >> Support Engineer - IPexpert, Inc. >> URL: http://www.IPexpert.com <http://www.ipexpert.com/> >> >> >> ___ >> For more information regarding industry leading CCIE Lab training, please >> visit www.ipexpert.com >> >> >> > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hi, You want to transcode at the br2 rtr as I suppose your requirement is to transport the call using g.729 over the wan right?, if that's the case, make sure the incoming dial-peer codec is set to g.729, in that case the transcoder at br2 shoud be invoked if the sip phone codec is set to g.711, On Fri, Mar 12, 2010 at 10:03 AM, Omotayo wrote: > Hello Otto, > > i had same issue > > The transcoder can be on the trunk? > > When i did the transcoder on the br2 router, i get a busy tone when the sip > phone is being called from the hq phone > > REgards > > On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez wrote: > >> Hi Jeff, >> >> Would you please tell us more about the call flow and the end to end codec >> requirements for this call. If doing g.729 over the wan, and your sip phone >> is using g.711 you should transcode at br2, >> >> Please let us know, >> >> Thanks, >> >> On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter wrote: >> >>> Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder >>> on UCM. Can’t seem to get a call from Call Manager to CME sip phone >>> working. I can call from CME to UCM but not the other way around. Rings but >>> disconnects when answered. Transcoder shows registered in Call manager. >>> Thanks >>> >>> >>> >>> >>> >>> Jeff >>> >>> ___ >>> For more information regarding industry leading CCIE Lab training, please >>> visit www.ipexpert.com >>> >>> >> >> >> -- >> Regards, >> >> Otto Sanchez >> CCIE #25592 (Voice) >> Support Engineer - IPexpert, Inc. >> URL: http://www.IPexpert.com <http://www.ipexpert.com/> >> >> ___ >> For more information regarding industry leading CCIE Lab training, please >> visit www.ipexpert.com >> >> > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] via gatekeeper "invia" key word
Hi Jeff, According to your lab results, you are describing the expected behavior, more information at: http://www.cisco.com/en/US/partner/docs/ios/voice/cubegk/configuration/guide/ve-gk-config_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1225776 Thanks!, On Thu, Mar 11, 2010 at 9:55 AM, Jeff Cotter wrote: > I am struggling a bit with the invia concept. I think I understand the > “outvia”. When I lab this up I find the following. > > > > Invia only applies to calls coming from a remote GK. In order for call to > use cube I had to configure the invia key word on the actual remote > zone…..not on the destination zone. Sample config of my invia GK > > > > gk zone local ucm cisco.com 1.1.1.1 > > gk zone local cube > > gk zone local cme > > gk zone remote gk2 lab.com 2.2.2.2 invia cube > > zone prefixs omitted > > > > So calls coming FROM gk2 destined for either ucm or cme zone used the > cube. If I applied the invia key word on either ucm or cme zone directly, > the cube was not invoked. This seems to conflict with the proctor guide > mock lab 1 statement “invia command when defining the UCME zone would invoke > the cube for calls coming in from a remote zone”. In my lab applying invia > directly to destination zone had no affect and cube was not invoked. > > > > Am I missing something. > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 4 AB GK registration problem
Hi Mike,
I'm noticing from your initial debugs that the 156.26.1.70:1719 ip
address/port the one confirming the GRQ message from BR2-RTR
***
value RasMessage ::= gatekeeperConfirm :
{
requestSeqNum 126
protocolIdentifier { 0 0 8 2250 0 4 }
gatekeeperIdentifier {"PL"}
rasAddress ipAddress :
{
ip '*9C1A0146*'H
port *1719*
}
}
After the Angel's suggestion this should have been corrected, so would you
please send the same debugs now from both routers? plus a show gatek zone
status, also I see that you are not pinging from the br2 l0 interface but
the closest to hq l0 interface (which might be the serial interface), please
try to use ping with options to verify that loopbacks can see each other,
Thanks!,
On Thu, Mar 11, 2010 at 2:24 PM, Mike Peterson wrote:
> Hi Angel,
>
> Thanks for helping me out with this GK issue. Yes indeed the GW doesn't
> receive the message , that is why we are seeing GRQ and GCF.
> I do have full connectivity b/w HQ/BR2/PUB/SUB .
> Below are the ping's you sugested to post.
>
> Thanks a lot in advance for your time and help.
>
> HQ#ping 192.21.66.254->from HQ to loopback of BR2
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.21.66.254, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 1/5/12 ms
> HQ#
>
>
> BR2-RTR#ping 192.21.64.254-->from BR2 to loopback of HQ
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.21.64.254, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 8/14/32 ms
> BR2-RTR#
>
>
>
>
>
> HQ#ping 192.168.0.11 >from HQ to CUCM PUB
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.168.0.11, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 1/4/8 ms
>
>
>
> HQ#ping 192.168.0.12 -> from HQ to CUCM SUB
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.168.0.12, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 16/25/40 ms
> HQ#
>
>
> BR2-RTR#ping 192.168.0.11 ->from BR2 CUCM PUB
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.168.0.11, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 16/27/44 ms
>
>
> BR2-RTR#ping 192.168.0.12---> from BR2 to CUCM SUB
>
> Type escape sequence to abort.
> Sending 5, 100-byte ICMP Echos to 192.168.0.12, timeout is 2 seconds:
> !
> Success rate is 100 percent (5/5), round-trip min/avg/max = 20/40/52 ms
> BR2-RTR#
>
>
>
> --
> *From:* Angel Perez **
> *Sent:* Thu, March 11, 2010 1:17:17 PM
> *Subject:* RE: [OSL | CCIE_Voice] Lab 4 AB GK registration problem
>
> Just to verify, can you ping hq loo 0 192.21.64.254 from br2? And br2 loop
> 192.21.66.254 from hq?
>
> It looks like br2 gw ask for registration GRQ, and then gk try to confirm
> GCF but the gw can't recieve the message
>
> hth
> --
> Date: Thu, 11 Mar 2010 08:54:30 -0800
>
>
> Subject: Re: [OSL | CCIE_Voice] Lab 4 AB GK registration problem
>
> Hi All,
>
> I did tried your suggestion (to add loopback IP address :
> zone local PL cisco.com 192.21.64.254 ) which does make sense but it
> doesn't work.
> I took a look at "deb ras" and I am seeing only GRQ (a message sent by
> endpoint to GK ) and GCF (A reply from gatekeeper to endpoint
> which indicates the transport address of the gatekeeper RAS channel) and I
> am not seeing GRJ (the reject the endpoint request for registration) so
> something I am missing or I am hitting a BUG!
> The "deb gatekeeper main 19" or "deb h225 asn1" still doesn't give me a
> clue of why GK is failing to register.
>
> Once again thanks for your time and help.
>
> Kind Regards,
>
> Mike
>
>
> Note: This is the change I made:
>
> gatekeeper
> zone local PL cisco.com 192.21.64.254
> zone prefix PL 1... gw-priority 10 gk-trunk_2
> zone prefix PL 1... gw-priority 9 gk-trunk_1
> zone prefix PL 1... gw-priority 0 BR2-RTR
> zone prefix PL 5... gw-priority 10 gk-trunk_2
> zone prefix PL 5... gw-priority 9 gk-trunk_1
> zone prefix PL 5... gw-priority 0 BR2-RTR
> no shutdown
> !
>
>
>
>
> --
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Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1
Hello Jason, E1's and T1's will always use a-law and u-law companding mechanism respectively, this is used to give more "resolution" to low voice frequencies when digitizing an analog signal (the mechanism is also used in the other end for digital to analogue conversion), each mechanism is designed exclusively to work with its voice digital standard and cannot be used conversely, In that sense, my guess is that before applying that command in your E1 port, the companding type was u-law, you can verify this using the sh voice port command (perhaps the default configuration of a-law was somehow overwritten by a cptone command in the same port configuration), and when you hardcoded the a-law companding type everything worked as expected, I also found a note in the Cisco IOS Voice Port Configuration Guide, which says that the command is used when cross-connecting in a local router, http://www.cisco.com/en/US/partner/docs/ios/voice/voiceport/configuration/guide/vp_cfg_digital_vps_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1009871 HTH, On Thu, Mar 11, 2010 at 2:37 PM, Jason Granat wrote: > So I’ve got this partially figured out. It had to do with the > compand-type. E1 was a-law and T1 was u-law. I set the E1 side for u-law and > it sounds correct now. > > > > The final thing I am trying to figure out is how to ‘trans-compand’ (if > that is the correct term) on the PSTN gateway. As it sits I had to change > the compand-type between the PSTN and E1 gateway. I don’t have experience > with foreign connectivity so maybe this is the way it is done in the real > world but I am thinking that perhaps the E1 site may not want or be able to > change their compand-type, so can it be changed at the PSTN level between > a-law and u-law locations? > > > > Thanks, > > > > Jason > > > > *From:* Jason Granat > *Sent:* Thursday, March 11, 2010 9:46 AM > *To:* > *Subject:* PSTN Call Distortion Between T1/E1 > > > > Perhaps this is something simple that I am overlooking but I have the > generic setup running in my home lab with 3 gateways and one PSTN router. 2 > of the gateways are T1 and one is E1. The PSTN router is also running CME > with a 7960 to simulate PSTN destinations. Calls from any site to the PSTN > phone are fine. Calls between T1 sites are fine. Calls between T1 and E1 > sites are distorted, like the gain is way too high. I tried playing with the > gain on the voice-port but no luck. I’m not finding much online or in Cisco > docs. Any suggestions? > > > > Thanks, > > > > Jason > > -- > > > http://slash128.com > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter wrote: > Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on > UCM. Can’t seem to get a call from Call Manager to CME sip phone working. > I can call from CME to UCM but not the other way around. Rings but > disconnects when answered. Transcoder shows registered in Call manager. > Thanks > > > > > > Jeff > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP WITH MLPoFR
Hi Angel, You are right, the "ip rsvp bandwidth" has to be configured wherever the outbound ip interface configuration is, as it will carry the reservation messages from one rsvp agent to the other, according to the routing protocol in place. In your case this new ip outbound interface is virtual-template200 HTH, On Wed, Mar 10, 2010 at 2:06 PM, Angel Perez wrote: > Hello: > > I was configurin MLPoFR and LFI on a link between hq and br1, on the serial > interface I had: > > *interface Serial0/2/0.202 point-to-point > ip rsvp bandwidth 64 * > > Calls where progressing as configured (two g729 calls) > > Then after apply *auto qos voip trust fr-atm *new virtual templates and > virtual access interfaces are created > > Then trying to test the policy-map just created and tuned I noticed that I > could not make calls from hq to br1 (rsvp was rejecting the call) > > So I added the following at hq and br1: > > *interface Virtual-Template200 > ip rsvp bandwidth 64* > ** > And the problem get solved > ** > Is this the normal situation? I suppose it is but not 100% sure > > Thanks > > > > -- > ¿Te gustaría tener Hotmail en tu móvil Movistar? ¡Es > gratis!<http://serviciosmoviles.es.msn.com/hotmail/movistar-particulares.aspx> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] BARGE
Hi Angel, This setting will enable or disable the built-in bridge for all phones that support the barge feature (you still have the phone configuration that may override this configuration), I think it's outdated in the help, For a comprehensive list of phones that support the barge feature please take a look at the SRND (endpoints chapter): http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/endpnts.html On Thu, Mar 11, 2010 at 4:48 AM, Angel Perez wrote: > Hello: > > It seems to be a limitation on barge feature on phones other than 7940, > 7960, and 7970 > > From ccm help (service parameter): > *Built in brige: This parameter determines whether the bridge that is > built in to Cisco IP Phone models 7940, 7960, and 7970 is enabled* > ** > So I suppose that for model 7961/7941 and higher the only option is cBarge > instead of Barge > > Anyone has seen this before > > Thanks > > > > -- > ¿Quieres saber qué móvil eres? ¡Descúbrelo aquí!<http://www.quemovileres.com/> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Issues With CME BLF Call Lists
Hi Scott, As Angel, I remember having this issue before and a router reload seemed to fix it, if this still doesn't work, please post your config, Thanks, On Tue, Mar 9, 2010 at 4:54 AM, Angel Perez wrote: > Hello: > > something similar happend to me, if you think that your config is ok, try > to reset all phones, if this don´t help just reload the router, this worked > for me > > hth > > -- > From: scarruthe...@hotmail.com > To: naoufal.kerbo...@cbi.ma > Date: Tue, 9 Mar 2010 01:01:29 -0800 > > CC: ccie_voice@onlinestudylist.com > Subject: Re: [OSL | CCIE_Voice] Issues With CME BLF Call Lists > > Thanks Naoufal, > > I actually already had the presence call-list, presence-enable, and allow > watch commands in place - the latter two had to be there as the BLF SD was > working for the same lines I was attempting to get presence for in the call > list. That's why it was odd - everything seemed to be in place. Anyone > else have trouble with this? > > Thanks > Scott > > > Date: Tue, 9 Mar 2010 08:55:23 + > > From: naoufal.kerbo...@cbi.ma > > To: scarruthe...@hotmail.com > > CC: ccie_voice@onlinestudylist.com > > Subject: Re: [OSL | CCIE_Voice] Issues With CME BLF Call Lists > > > > Hi scott, > > > > try to add this lines on your config: > > > > presence > > presence call-list > > ! > > sip-ua > > presence enable > > ! > > > > also "allow watch" under ephone-dn for all directory numbers. > > > > Regards > > Naoufal > > > > > > scott carruthers wrote: > > > I have review the configuration a dozen times - I have BLF speed dials > > > between two phones properly showing presence status on the line > > > appearance/SD but I cannot get the call list presence functionality to > > > work. I have presence call-list specified under presence config mode > > > - obviously the allow watch, etc is configured on the phones properly > > > as the BLFs are working. Any ideas? > > > > > > Thanks > > > Scott > > > > > > > > > > > > > > > > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up > > > now. <http://clk.atdmt.com/GBL/go/201469229/direct/01/> > > > > > > > > > > ___ > > > For more information regarding industry leading CCIE Lab training, > please visit www.ipexpert.com > > > > > > > -- > Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up > now. <http://clk.atdmt.com/GBL/go/201469226/direct/01/> > -- > ¿Sabes que la Videollamada de Messenger es GRATIS > ¡Descúbrela!<http://events.es.msn.com/windows-live/redes-sociales/default.aspx> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper Endpoint Max-Calls Command
Hi Scott, Your gk config is good, So, you have to work a little from your ucm cluster. A workaround to your requirement is to configure 2 separate trunks, each with only one server within its ucm group ie ccm-trunk-pub, ucm group consisting on the ucm pub only ccm-trunk-sub, ucm group consisting on the ucm sub only The output from the sh gatek endpoint may look like this: * HQ-RTR#sh gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.3 1720 10.10.110.3 59738 CME VOIP-GW H323-ID: BR2-RTR Voice Capacity Max.= Avail.= Current.= 0 10.10.210.1036392 10.10.210.1032797 HQVOIP-GW H323-ID: gk-trunkpub_1 Voice Capacity Max.= 1000 Avail.= 1000 Current.= 0 10.10.210.1134175 10.10.210.1132790 HQVOIP-GW H323-ID: gk-trunksub_2 Voice Capacity Max.= 1000 Avail.= 1000 Current.= 0 Total number of active registrations = 3 * Then, configure a route group with the two new gw with gk-trunksub_2 in the first place and the distribution algorithm top down, then the rl and rp, in that way your calls will always be sent from the sub node and if they fail the pub node will take care, Your gk config may have to change a bit to accomodate this changes, At the end you will see an output like this, *** HQ-RTR#sh gatek call Total number of active calls = 2. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 42-32984 15 0(Kbps) Endpt(s): Alias E.164Addr src EP: gk-trunksub_25002 CallSignalAddr Port RASSignalAddr Port 10.10.210.1134175 10.10.210.1132790 Endpt(s): Alias E.164Addr dst EP: BR2-RTR 1#3001 CallSignalAddr Port RASSignalAddr Port 10.10.110.3 1720 10.10.110.3 59738 LocalCallIDAge(secs) BW 43-32986 10 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk-trunkpub_1 5003 CallSignalAddr Port RASSignalAddr Port 10.10.210.1036392 10.10.210.1032797 Endpt(s): Alias E.164Addr dst EP: BR2-RTR 1#3002 CallSignalAddr Port RASSignalAddr Port 10.10.110.3 1720 10.10.110.3 59738 *** My gk config was the following *** gatekeeper zone local HQ ipexpert.com 10.10.110.1 zone local CME ipexpert.com zone prefix CME 3... zone prefix HQ 5... gw-priority 10 gk-trunksub_2 zone prefix HQ 5... gw-priority 9 gk-trunkpub_1 no shutdown endpoint resource-threshold endpoint max-calls h323id gk-trunksub_2 1 endpoint max-calls h323id gk-trunkpub_1 1000 !*** There's also a good reading on how the outbound calls are handled by the h.225 trunks in ucm, http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.html#wp1044813 HTH, ,On Sat, Mar 6, 2010 at 9:04 PM, scott carruthers wrote: > With the gatekeeper max-calls command - I have the following config to > only allow one active call to the CM sub and all subsequent calls will route > to the pub: > > no shutdown > endpoint resource-threshold > endpoint max-calls h323id CM-Trunk_2 1 > endpoint max-calls h323id CM-Trunk_1 1000 > > This works fine for call from CME to CM thru the GK. But in the CM to CME > direction one call succeeds and any subsequent calls fail. It seems CM will > not re-initiate the call attempt from the pub. I thought there was a > service parameter needed in conjunction with the max call command on the GK > but I am not able to find. Am I missing a simple service parameter or is > something else needed? > > > > -- > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up > now. <http://clk.atdmt.com/GBL/go/201469229/direct/01/> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Theory questions from Vol 1, Labs 5a-c
> > > 4. When it comes to mobility,I know that I can associate an end user with > a phone via End Users. I can also set an Owner User ID via the Device > page. I can also associate the user with a line. What is the big > difference between these features and what account association is required > for what service? > > > > Those are my questions for now. :) > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP phone registered with CUCME
Kalyan, Please try to use a lower sip firmware load version for you cme, the one that corresponds to your cme version 7.0(1), you may find this information in the cme compatibility matrix, Thanks, On Thu, Mar 4, 2010 at 12:06 PM, wrote: > > Hi Guys, > > I was wondering if any one else had the same problem that I have with a SIP > phone registered with CME > > CME version 7.0(1) on router 2821 running IOS 12.4(22) T3 > Phone - 7962 running SIP load SIP42.8-5-3-4S > > The SIP phone is registered with the CME. I can make calls /receive calls, > > I call from a SCCP/SIP phone registered with CUCM or from a PSTN phone, > call rings on the SIP phone and if I answer the call, everything is good. > However if I don't answer the phone and if I dont hang up the phone I made > the call from, the SIP phone keeps ringing on that line and never > disconnects. I have to reset the phone by unplugging the ethernet and reboot > the phone, > > Thanks > Kalyan > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 5A - 5.10
Kalyan, Did you make sure the ip configured in the h323-gateway voip bind srcaddr x.x.x.x command matches the ip configured in the ucm h.323 gw device name field?, Also, that your outoging hq-rtr voip outgoing dial-peers point to the ucm addresses, GK should be used to route ucm calls to ucme as in the case of the 3002 dn, On Wed, Mar 3, 2010 at 6:16 PM, wrote: > > Hi Guys, > > I was working on lab 5A and this specific task was to set up the support > hunt group that only works during a certain time period. Outside of the work > hours calls are routed to 3002. > > When I call from the PSTN, call comes into the HQ G/W then routed over to > the CallManager for call treatment. The HQ G/W also acts as the Gatekeeper, > so basically there are two paths from the HQ G/W to the CallManager - (1) a > GK trunk between CallManager and HQ G/W (2) The HQ G/W as a H323 G/W > administered on the CallManager. > > Now to the question, which path does the call go over? The reason I am > asking is because this has an impact of where I need to change the inbound > CSS. When I made the call, it came over the GK trunk. I noticed this because > I was changing the CSS on the H323 G/W and the call was always routed to the > 3002 number (because of partition of the translation pattern 5000). I > was expecting it to come over to the H323 G/W. > > Can someone help me understand why the call came over the GK trunk? > > > Thanks > Kalyan > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol2 Lab 4 - Presense - CUPS not working
Hi Wael, Please follow the guidelines in this blog to control your phone cti from cupc and let us know the results: http://blog.ipexpert.com/cti-phone-control-from-cupc/#more-2272 Thanks, On Wed, Mar 3, 2010 at 4:24 AM, Wael Agina wrote: > Dear All, > >I've integerate the cups - 10.10.210.12 with cucm. > I assigned the user gwashington as per PG to line 5002 of HQ PH 2 / My CIPC > phone. > > I am login using my CUPC client on my machine to the CUPS 10.10.210.12 , > however it is not working fine. > I can login, but not monitoring or dialing via phone hq ph 2 as supposed. > > User gwashington has device associated hq phe 2 and primary line 5002. > > > Any idea ? > > > -- > > Thanks and Best Regards, > Wael Agina > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] DTMF PROBLEM UNITY CON AND SIP PHONE
i, 26 Feb 2010 09:43:32 -0430 > Subject: Re: [OSL | CCIE_Voice] DTMF PROBLEM UNITY CON AND SIP PHONE > From: o...@ipexpert.com > To: gorr...@hotmail.com > CC: ccie_voice@onlinestudylist.com > > > Hi Angel, > > What sip phone types are you using for br1 and hq sites?, > > Thanks!! > > On Fri, Feb 26, 2010 at 6:57 AM, Angel Perez wrote: > > Hello: > > I've the following issue: > > When I call uc from br1 sip phone the DTMF are getting distorted, I can > call uc from the phone then the password is asked, but then uc never gets > the correct password (I can see the Failed Logon Attempt at user page in UC > are increasing) > ** > From a sccp phone at br1 I've no problems neither from sip phone at HQ > > Looks like the DTMF's were losed crossing the WAN... > > Any suggestion? Thanks > > -- > ¡Nuevo MSN Entretenimiento! Todos los trailers, series de tv y videoclips, > los mejores juegos online y lo último sobre tus estrellas > favoritas.<http://entretenimiento.es.msn.com/> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com <http://www.ipexpert.com/> > > -- > ¡Nuevo MSN Entretenimiento! Todos los trailers, series de tv y videoclips, > los mejores juegos online y lo último sobre tus estrellas > favoritas.<http://entretenimiento.es.msn.com/> > > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com <http://www.ipexpert.com/> > > -- > ¿Quieres tener a tus amigos de Facebook en Messenger? ¡Clic > AQUÍ!<http://vivelive.com/feedfacebook/> > > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com <http://www.ipexpert.com/> > > -- > ¿Quieres tener a tus amigos de Facebook en Messenger? ¡Clic > AQUÍ!<http://vivelive.com/feedfacebook/> > > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com <http://www.ipexpert.com/> > > -- > Actualízate gratis al nuevo Internet Explorer 8 y navega más > seguro<http://www.microsoft.com/spain/windows/internet-explorer/default.aspx> > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 5c Task 5.3 - 911 calls from HQ disconnecting
Hi Steve, What's the output for the following commands: -. sh controllers t1 -. sh isdn status -. sh voice port summary Also, please configure an incoming dial-peer in hq-rtr to receive h.323 calls from ucm, dial-peer voice x voip incoming called-number . Thanks, On Tue, Mar 2, 2010 at 1:22 PM, Steve Denney (stdenney) wrote: > With attachment this time :) > > > > > > *From:* Steve Denney (stdenney) > *Sent:* Tuesday, March 02, 2010 12:52 PM > *To:* ccie_voice@onlinestudylist.com > *Subject:* Lab 5c Task 5.3 - 911 calls from HQ disconnecting > > > > Having trouble with calls from HQ Ph2 (SIP CIPC DN 5002) to 911 (PSTN > phone). > > This is similar to what happened last time I did this lab (on a different > pod). > > > > The call hits the GW fine, but I get a reorder tone on the CIPC. > > The call actually gets as far as the PSTN router, then disconnects as per > the debug ISDN q931 output (below). > > The PSTN phone alerts as if it’s trying to accept the call, but the call > can’t be picked up. > > > > Calls from BR1 to 911 work fine. > > Debug shows Dial peer matching on HQ router is correct. > > Have restarted **everything** (all routers including PSTN, phones, > pub/sub, HQ 3750). > > > > HQ router config attached. > > > > Any ideas? About to chalk it up to a bug and move on... > > > > cheers, sd > > > > > > HQ-RTR Debug: > > > > Mar 2 17:41:20.667: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid > type/plan 0x0 0x0 may be overriden; sw-type 13 > > Mar 2 17:41:20.667: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type > 0xD is 0x2 0x1, Calling num 2123945002 > > Mar 2 17:41:20.667: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type > 0xD is 0x0 0x0, Called num 911 > > Mar 2 17:41:20.671: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = > 0x0087 > > Bearer Capability i = 0x8090A2 > > Standard = CCITT > > Transfer Capability = Speech > > Transfer Mode = Circuit > > Transfer Rate = 64 kbit/s > > Channel ID i = 0xA98383 > > Exclusive, Channel 3 > > Calling Party Number i = 0x2181, '2123945002' > > Plan:ISDN, Type:National > > Called Party Number i = 0x80, '911' > > Plan:Unknown, Type:Unknown > > Mar 2 17:41:20.703: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8 callref > = 0x8087 > > Channel ID i = 0xA98383 > > Exclusive, Channel 3 > > Mar 2 17:41:20.715: ISDN Se0/0/0:23 Q931: RX <- ALERTING pd = 8 callref = > 0x8087 > > Progress Ind i = 0x8188 - In-band info or appropriate now available > > > Mar 2 17:41:20.755: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref > = 0x0087 > > Cause i = 0x80AC - Requested circuit/channel not available > > Mar 2 17:41:20.767: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = > 0x8087 > > Mar 2 17:41:20.771: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 > callref = 0x0087 > > > > > > PSTN-WAN Router Debug: > > > > Mar 2 17:41:20.675: ISDN Se0/3/0:23 Q931: RX <- SETUP pd = 8 callref = > 0x0087 > > Bearer Capability i = 0x8090A2 > > Standard = CCITT > > Transfer Capability = Speech > > Transfer Mode = Circuit > > Transfer Rate = 64 kbit/s > > Channel ID i = 0xA98383 > > Exclusive, Channel 3 > > Calling Party Number i = 0x2181, '2123945002' > > Plan:ISDN, Type:National > > Called Party Number i = 0x80, '911' > > Plan:Unknown, Type:Unknown > > Mar 2 17:41:20.699: ISDN Se0/3/0:23 Q931: TX -> CALL_PROC pd = 8 callref > = 0x8087 > > Channel ID i = 0xA98383 > > Exclusive, Channel 3 > > > > Mar 2 17:41:20.711: ISDN Se0/3/0:23 Q931: TX -> ALERTING pd = 8 callref = > 0x8087 > > Progress Ind i = 0x8188 - In-band info or appropriate now available > > > Mar 2 17:41:20.759: ISDN Se0/3/0:23 Q931: RX <- DISCONNECT pd = 8 callref > = 0x0087 > > Cause i = 0x80AC - Requested circuit/channel not available > > Mar 2 17:41:20.763: ISDN Se0/3/0:23 Q931: TX -> RELEASE pd = 8 callref = > 0x8087 > > Mar 2 17:41:20.775: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8 > callref = 0x0087 > > > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Display IE and MGCP interaction
Hi, When using mgcp the isdn outgoing display-ie command is not needed in the rtr's serial interface as this information will be controlled from the ucm itself (mgcp controls l3), so only checking or unchecking the Display IE Delivery check-box will allow or restrict the name presentation to the pstn respectively, If you were using a h.323 gw, that command will be needed along with the ucm display ie configuration for the name to be sent. So it's recommended to always configure the command as your mgcp gw may fall into srts and that configuration will be needed for the name to be sent, Thanks, On Wed, Feb 24, 2010 at 9:56 PM, t n wrote: > Display IE Delivery > > Check the check box to enable delivery of the display information element > (IE) in SETUP and NOTIFY messages (for DMS protocol) for the calling and > connected party name delivery service. > > I guess it would be necessary to check that box if one is connected to a > DMS switch. It's dependent upon the ISDN switchtype. With primary-ni, those > must get sent automatically. Yet what about the MGCP q931 backhaul portion? > > On Wed, Feb 24, 2010 at 9:16 PM, t n wrote: > >> >> I've observed in the debugs that you only need to configure "isdn outgoing >> display-ie" on the D-channel of the gateway in order to the the calling name >> sent to the PSTN. It's not necessary to check "Display IE Delivery" in the >> gateway config of CUCM. >> >> Doing so we get the following is debug isdn q931: >> >> Feb 25 01:53:34.512: ISDN Se1/0:23 Q931: TX -> SETUP pd = 8 callref = >> 0x0086 >> Bearer Capability i = 0x8090A2 >> Standard = CCITT >> Transfer Capability = Speech >> Transfer Mode = Circuit >> Transfer Rate = 64 kbit/s >> Channel ID i = 0xA98383 >> Exclusive, Channel 3 >> *Display i = 'HQ 2 SIP 7965'* >> Calling Party Number i = 0x2181, '2123945002' >> Plan:ISDN, Type:National >> Called Party Number i = 0xA1, '2123942123' >> Plan:ISDN, Type:National >> Feb 25 01:53:34.632: ISDN Se1/0:23 Q931: RX <- CALL_PROC pd = 8 callref = >> 0x8086 >> Channel ID i = 0xA98383 >> Exclusive, Channel 3 >> Feb 25 01:53:34.664: ISDN Se1/0:23 Q931: RX <- ALERTING pd = 8 callref = >> 0x8086 >> >> >> However with MGCP, even with "isdn outgoing display-ie" removed, the >> calling name information element would always get sent. I always see it in >> my debugs. Is that how it's supposed to work? >> >> >> -- >> Thanks. >> >> tnn314.wordpress.com >> > > > > -- > Thanks. > > tnn314.wordpress.com > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] class based shaping and cRTP
e 0 bps > > > > Mark Nigh > *Systems Engineer* > > *mn...@netelligent.com* > > (p) 314.392.6926 > [image: img_moreThan] <http://www.netelligent.com/> > > > > > -- > > This transmission and any attached files are privileged, confidential or > otherwise the exclusive property of the intended recipient or Netelligent > Corporation. If you are not the intended recipient, any disclosure, copying, > distribution or use of any of the information contained in or attached to > this transmission is strictly prohibited. If you have received this > transmission in error, please contact us immediately by responding to this > message or by telephone (314-392-6900) and promptly destroy the original > transmission and its attachments. > > -- > This transmission and any attached files are privileged, confidential or > otherwise the exclusive property of the intended recipient or Netelligent > Corporation. If you are not the intended recipient, any disclosure, copying, > distribution or use of any of the information contained in or attached to > this transmission is strictly prohibited. If you have received this > transmission in error, please contact us immediately by responding to this > message or by telephone (314-392-6900) and promptly destroy the original > transmission and its attachments. > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] DTMF PROBLEM UNITY CON AND SIP PHONE
Enable All Phone Device Trace, MTP Trace and SIP Call Processing Trace, Thanks, On Mon, Mar 1, 2010 at 11:22 AM, Angel Perez wrote: > Hello: > > Otto, wich ccm traces do suggest me? "All Phone Device Trace" or trace by > device > > Thanks > > -- > Date: Fri, 26 Feb 2010 17:31:54 -0430 > > Subject: Re: [OSL | CCIE_Voice] DTMF PROBLEM UNITY CON AND SIP PHONE > From: o...@ipexpert.com > To: gorr...@hotmail.com > CC: ccie_voice@onlinestudylist.com > > Hi Angel, > > Have you tried to move the br1 phone to the hq site and dp?, did you get > the same result?, br1 phone should be sending dtmf digits after the call is > answered by cuc as kpml, did you take a look at the ucm traces to see if ucm > is getting those digits? > > thanks, > > On Fri, Feb 26, 2010 at 12:02 PM, Angel Perez wrote: > > Hello: > > All sip phones are 7961 with SIP41.8-4-1s loads (the one at br1 and the one > at hq) > br1 is a h323 gw registered to ucm > > gracias > > -- > Date: Fri, 26 Feb 2010 09:43:32 -0430 > Subject: Re: [OSL | CCIE_Voice] DTMF PROBLEM UNITY CON AND SIP PHONE > From: o...@ipexpert.com > To: gorr...@hotmail.com > CC: ccie_voice@onlinestudylist.com > > > Hi Angel, > > What sip phone types are you using for br1 and hq sites?, > > Thanks!! > > On Fri, Feb 26, 2010 at 6:57 AM, Angel Perez wrote: > > Hello: > > I've the following issue: > > When I call uc from br1 sip phone the DTMF are getting distorted, I can > call uc from the phone then the password is asked, but then uc never gets > the correct password (I can see the Failed Logon Attempt at user page in UC > are increasing) > ** > From a sccp phone at br1 I've no problems neither from sip phone at HQ > > Looks like the DTMF's were losed crossing the WAN... > > Any suggestion? Thanks > > -- > ¡Nuevo MSN Entretenimiento! Todos los trailers, series de tv y videoclips, > los mejores juegos online y lo último sobre tus estrellas > favoritas.<http://entretenimiento.es.msn.com/> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com <http://www.ipexpert.com/> > > -- > ¡Nuevo MSN Entretenimiento! Todos los trailers, series de tv y videoclips, > los mejores juegos online y lo último sobre tus estrellas > favoritas.<http://entretenimiento.es.msn.com/> > > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com <http://www.ipexpert.com/> > > -- > ¿Quieres tener a tus amigos de Facebook en Messenger? ¡Clic > AQUÍ!<http://vivelive.com/feedfacebook/> > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Issue with UC Vol2 Lab7
Hi Mark, Did you change the UC default forwarded routing rules?, it seems like it tries to "attempt to sign in" instead of "attempt forward" when a forwarded call gets to UC and if a subscriber is not found (UK site) the opening greeting is played, Thanks, On Sun, Feb 28, 2010 at 2:31 PM, Mark Drucker wrote: > I'm having an issue with UC in this lab. I call from DN 5002 to 1002 and I > get a sign in when it his VM. The same thing happens the other way around > and when I call from UK to either US DN it end up at the opening greeting > instead of the users VM. Any ideas? > > Thanks for your help! > > -- > Mark A. Drucker > (925) 321-5791 > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP & SK phones show different time in CME
Pavan, Please double check the following: -. CME rtr is sincronyzed with ntp server (show ntp status, sh ntp associations) -. Phone type under ephone configuration exactly matches your real phone type -. Issue a no create cnf-files, create cnf-files and reset ephones If no success, please reload the router, Thanks, On Sat, Feb 27, 2010 at 7:06 PM, Pavan K wrote: > > I am running 8.3.2.27 on the SK phones. IOS is 12.4(20)T2 and CME 7.0(0) > The SK phone seems to be showing incorrect time. > Has anybody seen this before ? > > -Pavan > > > === > voice register global > mode cme > source-address 142.103.66.254 port 5060 > max-dn 10 > max-pool 2 > load 7961GE term61.default > load 7961 term61.default > authenticate register > timezone 43 > time-format 24 > tftp-path flash: > create profile sync 0007641738744237 > ntp-server 142.3.64.254 mode anycast > == > ntp server 142.3.64.254 > telephony-service > load 7970 term70.default > load 7971 term71.default > max-ephones 2 > max-dn 10 no-reg > ip source-address 142.103.66.254 port 2000 > system message Your current Options > time-zone 43 > time-format 24 > max-conferences 12 gain -6 > transfer-system full-consult > create cnf-files version-stamp 7960 Feb 27 2010 22:28:52 > === > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] bacd issue
t0/0 > no ip address > shutdown > duplex auto > speed auto > ! > interface Service-Engine0/0 > ip unnumbered Vlan400 > service-module ip address 10.10.202.2 255.255.255.0 > service-module ip default-gateway 10.10.202.1 > ! > interface FastEthernet0/1 > no ip address > shutdown > duplex auto > speed auto > ! > interface FastEthernet0/3/0 > switchport trunk native vlan 200 > switchport mode trunk > switchport voice vlan 400 > ! > interface FastEthernet0/3/1 > switchport trunk native vlan 200 > switchport mode trunk > switchport voice vlan 400 > ! > interface FastEthernet0/3/2 > shutdown > ! > interface FastEthernet0/3/3 > shutdown > ! > interface Serial0/0/0:15 > no ip address > encapsulation hdlc > isdn switch-type primary-net5 > isdn incoming-voice voice > isdn bchan-number-order ascending > isdn outgoing display-ie > no cdp enable > ! > interface Serial0/1/0:0 > no ip address > encapsulation frame-relay IETF > frame-relay lmi-type ansi > ! > interface Serial0/1/0:0.1 point-to-point > ip address 10.10.112.2 255.255.255.0 > ip ospf mtu-ignore > snmp trap link-status > frame-relay interface-dlci 102 > ! > interface Vlan1 > no ip address > ! > interface Vlan200 > ip address 10.10.102.1 255.255.255.0 > ! > interface Vlan400 > ip address 10.10.202.1 255.255.255.0 > ! > router ospf 1 > router-id 10.10.202.1 > log-adjacency-changes > network 10.10.0.0 0.0.255.255 area 0 > ! > ip forward-protocol nd > ip route 10.10.202.2 255.255.255.255 Service-Engine0/0 > ip http server > no ip http secure-server > ip http path flash:/GUI > ! > ! > ! > ! > ! > ! > ! > ! > ! > tftp-server flash:PHONE/7941-7961/apps41.8-3-2-27.sbn alias > apps41.8-3-2-27.sbn > tftp-server flash:PHONE/7941-7961/cnu41.8-3-2-27.sbn alias > cnu41.8-3-2-27.sbn > tftp-server flash:PHONE/7941-7961/cvm41sccp.8-3-2-27.sbn alias > cvm41sccp.8-3-2-27.sbn > tftp-server flash:PHONE/7941-7961/dsp41.8-3-2-27.sbn alias > dsp41.8-3-2-27.sbn > tftp-server flash:PHONE/7941-7961/jar41sccp.8-3-2-27.sbn alias > jar41sccp.8-3-2-27.sbn > tftp-server flash:PHONE/7941-7961/SCCP41.8-3-3S.loads alias > SCCP41.8-3-3S.loads > tftp-server flash:PHONE/7941-7961/term41.default.loads alias > term41.default.loads > tftp-server flash:PHONE/7941-7961/term61.default.loads alias > term61.default.loads > ! > control-plane > ! > ! > ! > voice-port 0/0/0:15 > translation-profile incoming IN > ! > ! > ! > ! > ! > dial-peer voice 100 pots > incoming called-number . > direct-inward-dial > ! > dial-peer voice 999 pots > translation-profile outgoing OUT > destination-pattern 999 > port 0/0/0:15 > forward-digits all > ! > dial-peer voice 300 voip > voice-class codec 1 > incoming called-number . > dtmf-relay h245-alphanumeric > no vad > ! > dial-peer voice 400 voip > destination-pattern [15]... > session target ras > tech-prefix 1# > dtmf-relay h245-alphanumeric > no vad > ! > dial-peer voice 8 pots > destination-pattern 9 > port 0/0/0:15 > forward-digits 8 > ! > dial-peer voice 3600 voip > destination-pattern 3600 > session protocol sipv2 > session target ipv4:10.10.202.2 > incoming called-number 199[89] > dtmf-relay sip-notify > codec g711ulaw > no vad > ! > dial-peer voice 222 voip > service aa > destination-pattern 3007 > session target ipv4:10.10.110.3 > incoming called-number 3007 > dtmf-relay h245-alphanumeric > codec g711ulaw > no vad > ! > ! > num-exp 34 9 > gateway > timer receive-rtp 1200 > ! > ! > ! > gatekeeper > shutdown > ! > ! > telephony-service > no auto-reg-ephone > authentication credential admin cisco > max-ephones 3 > max-dn 8 > ip source-address 10.10.202.1 port 2000 > system message ipexpert > url services http://10.10.202.2/voiceview/common/login.do > url authentication http://10.10.202.1/CCMCIP/authenticate.asp > time-zone 42 > voicemail 3600 > max-conferences 8 gain -6 > moh music-on-hold.au > web admin system name admin password cisco > dn-webedit > time-webedit > transfer-system full-consult > create cnf-files version-stamp 7960 Feb 28 2010 03:55:39 > ! > ! > ephone-dn 1 > number 3001 > description 32143001 > ! > ! > ephone-dn 2 dual-line > number 3002 > label SiteC Phone 2 > description 32143002 > name SiteC Phone 2 > call-forward busy 3600 > call-forward noan 3600 timeout 15 > ! > ! > ephone-dn 4 > number 1998.... no-reg primary > mwi on > ! > ! >
Re: [OSL | CCIE_Voice] Lab 4.4 Question
Hi Randall, In that case, 10 digits (2123942123) should be forwarded to the pstn, On Thu, Feb 25, 2010 at 3:35 AM, Randall Crumm < randall.cr...@harmonicinc.com> wrote: > It looks like the requirement is to have OB call to 212 area code, the gw > should fwd 10 digits(area code+7) > If so then should it be forward digits 10 and not 7? > > Am I reading it wrong? > > Thanks, > Randall > > -Original Message- > From: ccie_voice-boun...@onlinestudylist.com [mailto: > ccie_voice-boun...@onlinestudylist.com] On Behalf Of > ccie_voice-requ...@onlinestudylist.com > Sent: Wednesday, February 24, 2010 6:27 PM > To: ccie_voice@onlinestudylist.com > Subject: CCIE_Voice Digest, Vol 48, Issue 152 > > Send CCIE_Voice mailing list submissions to >ccie_voice@onlinestudylist.com > > To subscribe or unsubscribe via the World Wide Web, visit >http://onlinestudylist.com/mailman/listinfo/ccie_voice > or, via email, send a message with subject or body 'help' to >ccie_voice-requ...@onlinestudylist.com > > You can reach the person managing the list at >ccie_voice-ow...@onlinestudylist.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of CCIE_Voice digest..." > > > Today's Topics: > > 1. Re: SDI or CCM trace logs in CCM 7 (Tanner Ezell) > 2. Re: SDI or CCM trace logs in CCM 7 (t n) > 3. Re: + dialing (Otto Sanchez) > 4. Display IE and MGCP interaction (t n) > 5. Re: Display IE and MGCP interaction (t n) > > > -- > > Message: 1 > Date: Wed, 24 Feb 2010 17:02:42 -0500 > From: Tanner Ezell > Subject: Re: [OSL | CCIE_Voice] SDI or CCM trace logs in CCM 7 > To: t n > Cc: OSL Group > Message-ID: ><9c4f122d1002241402s50bd68b6mde142aa66b767...@mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Yea, RTMT pretty much sucks :) > > On Wed, Feb 24, 2010 at 4:59 PM, t n wrote: > > You can also from the shell: > > > > file list activelog / > > file list activelog /cm/trace/ccm/sdi/ > > then file view > > file view activelog /cm/trace/ccm/sdi/ccm0001.txt > > > > > > RTMT is better although it can be much better. > > > > > > > > > > -- > Regards, > Tanner Ezell > > > -- > > Message: 2 > Date: Wed, 24 Feb 2010 17:16:49 -0500 > From: t n > Subject: Re: [OSL | CCIE_Voice] SDI or CCM trace logs in CCM 7 > To: Tanner Ezell > Cc: OSL Group > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > Yup. :) > > > http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/cucos/5_1_1/iptpappa.html#wp1009409 > > admin:file search activelog cm/trace/ccm/sdi/ccm0004.txt > 10\.10\.200\.254 > > Searching path: /var/log/active/cm/trace/ccm/sdi/ccm0004.txt > Searching file: /var/log/active/cm/trace/ccm/sdi/ccm0004.txt > 02/10/2010 20:11:14.924 CCM|StationD:(002) DND settings from TSP: > status=0, option=0, > > ringSetting=5| > 02/10/2010 20:12:13.118 CCM|StationD:(003) DND settings from TSP: > status=0, option=0, > > ringSetting=5| > > | > 02/10/2010 20:14:52.557 CCM|ConnectionManager - wait_AuDisconnectRequest > ERROR:NO ENTRY FOUND IN > > TABLE,CI(30237231,30237232),dcType=1,IFCreated(0,0),PID(0-0,0-0),IFHandling(0,0),MCNode(0,0)| > 02/10/2010 20:14:52.558 > CCM|MatrixControl:updatePartyMediaCoordinatorNodeId: > party1 videoCapable=0, party 2 > > videocapable=0| > > | > -- next part -- > An HTML attachment was scrubbed... > URL: > http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100224/6e86e814/attachment-0001.htm > > -- > > Message: 3 > Date: Wed, 24 Feb 2010 19:52:43 -0430 > From: Otto Sanchez > Subject: Re: [OSL | CCIE_Voice] + dialing > To: Cristobal Priego > Cc: ccie_voice@onlinestudylist.com > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > Hi Cris, > > Please try first to globalize the calling number (TP) and have that number > on the phone display and call directories. When you are sure everything is > working as desired, use the cg xform patterns as discussed before to > localize the phone display, after that, you still should see the calling > number's globalized form in the phone directories. Use the 7941s you have > for testing this feature, > > Thanks, > > On Wed, Feb 24, 2010 at 3:53 PM, Cristobal Priego < > cristobalpri...@gmail.com > > wrote: &
Re: [OSL | CCIE_Voice] DTMF PROBLEM UNITY CON AND SIP PHONE
Hi Angel, Have you tried to move the br1 phone to the hq site and dp?, did you get the same result?, br1 phone should be sending dtmf digits after the call is answered by cuc as kpml, did you take a look at the ucm traces to see if ucm is getting those digits? thanks, On Fri, Feb 26, 2010 at 12:02 PM, Angel Perez wrote: > Hello: > > All sip phones are 7961 with SIP41.8-4-1s loads (the one at br1 and the one > at hq) > br1 is a h323 gw registered to ucm > > gracias > > -- > Date: Fri, 26 Feb 2010 09:43:32 -0430 > Subject: Re: [OSL | CCIE_Voice] DTMF PROBLEM UNITY CON AND SIP PHONE > From: o...@ipexpert.com > To: gorr...@hotmail.com > CC: ccie_voice@onlinestudylist.com > > > Hi Angel, > > What sip phone types are you using for br1 and hq sites?, > > Thanks!! > > On Fri, Feb 26, 2010 at 6:57 AM, Angel Perez wrote: > > Hello: > > I've the following issue: > > When I call uc from br1 sip phone the DTMF are getting distorted, I can > call uc from the phone then the password is asked, but then uc never gets > the correct password (I can see the Failed Logon Attempt at user page in UC > are increasing) > ** > From a sccp phone at br1 I've no problems neither from sip phone at HQ > > Looks like the DTMF's were losed crossing the WAN... > > Any suggestion? Thanks > > -- > ¡Nuevo MSN Entretenimiento! Todos los trailers, series de tv y videoclips, > los mejores juegos online y lo último sobre tus estrellas > favoritas.<http://entretenimiento.es.msn.com/> > > _______ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > > > > -- > Regards, > > Otto Sanchez > CCIE #25592 (Voice) > Support Engineer - IPexpert, Inc. > URL: http://www.IPexpert.com <http://www.ipexpert.com/> > > -- > ¡Nuevo MSN Entretenimiento! Todos los trailers, series de tv y videoclips, > los mejores juegos online y lo último sobre tus estrellas > favoritas.<http://entretenimiento.es.msn.com/> > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] DTMF PROBLEM UNITY CON AND SIP PHONE
Hi Angel, What sip phone types are you using for br1 and hq sites?, Thanks!! On Fri, Feb 26, 2010 at 6:57 AM, Angel Perez wrote: > Hello: > > I've the following issue: > > When I call uc from br1 sip phone the DTMF are getting distorted, I can > call uc from the phone then the password is asked, but then uc never gets > the correct password (I can see the Failed Logon Attempt at user page in UC > are increasing) > ** > From a sccp phone at br1 I've no problems neither from sip phone at HQ > > Looks like the DTMF's were losed crossing the WAN... > > Any suggestion? Thanks > > -- > ¡Nuevo MSN Entretenimiento! Todos los trailers, series de tv y videoclips, > los mejores juegos online y lo último sobre tus estrellas > favoritas.<http://entretenimiento.es.msn.com/> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Volume 2 Lab 5
Rebot, 1.- The subscriber hq gw incoming calling party for subscriber calls should be +44:1 in order to globalize local calls with the +44 20 5943 format (whitout the 0). Also all UK phones' should follow this format regarding the ext phone number mask and calling number, 2.- You are right, since no TON was required for emergency calls, it is ok to set the TON required for local calls for those type of calls and configure a single cg xform pattern for local and emergency calls, Thanks, On Wed, Feb 24, 2010 at 4:53 PM, Remon wrote: > Hi, > >I am doing vol2 lab 5 and I found a lot of weird things in that lab , I > hope any one can help me with: > > 1- The calling number globalized format for UK number should be > +442054934xxx while in the work book shows that the number that should be > displayed in the call list as globalized format is +44 0 2054934xxx and > the proctor guide follow the first format (+442054934xxx) in the > verification part . > > Also for the incoming call prefixes for HQ gateway the procotor guide > mentioned I should add +44 for Subscriber TON and +44:1 for national, and > the PSTN always send subscriber TON and the digits in the following format > (0205494xxx) as calling number that mean that globalized number sill be +44 > 0 205493 while in the verification part it shows something else.I am really > confused. > > 2- For emergency calls :nothing mentioned in eth workbook about > setting certain TON for the calling number while the proctor guide shows > that it should be set to National , in the mean while we have another > calling party transformation pattern in the same pt with the same range of > extensions (4xxx) and set the TON to subscriber . For sure there will be > overlap . > > > > Can some one help me with these issues ? > > > > Regards > > Rebot > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] + dialing
Hi Cris, Please try first to globalize the calling number (TP) and have that number on the phone display and call directories. When you are sure everything is working as desired, use the cg xform patterns as discussed before to localize the phone display, after that, you still should see the calling number's globalized form in the phone directories. Use the 7941s you have for testing this feature, Thanks, On Wed, Feb 24, 2010 at 3:53 PM, Cristobal Priego wrote: > The issue was the predot > > i have a question now, when i call i see on the display 4 digits, and if i > don't answer the call i only see the 4 digits, what do i need to do to see > internal missed calls to display the globalized number? > > thanks > > 2010/2/23 Cristobal Priego > > I didn't have the predot >> maybe that's why >> I'm using a couple of 7921's and 7941's and an IPC (i know this soft phone >> doesn't support the + dialing) >> >> 2010/2/22 Otto Sanchez >> >> Hi Cris, >>> >>> Excelente!, espero estes bien!, >>> >>> I assume you have a cg xform pattern in a separate pt that is reachable >>> from the "Calling Party Transformation CSS" css setting on the phone. Does >>> that pattern look like this: >>> >>> \+1916251. with Discard Digit Instructions predot or Calling Party >>> Transformation Mask , >>> >>> Please double check this, also, just curious what phone type are you >>> using?, >>> >>> Thanks, >>> >>> >>> >>> On Mon, Feb 22, 2010 at 4:31 PM, Cristobal Priego < >>> cristobalpri...@gmail.com> wrote: >>> >>>> Hi Otto, >>>> >>>> Como estas mi estimado amigo ? como te trata la vida ? >>>> >>>> this is what i did >>>> >>>> 1006 dials 1007 --> hits translation pat , prefix +1 and uses >>>> Calling party xform mask --> after translation we have +1916251 >>>> (+19162511006) >>>> >>>> on 1007 "Calling Party Transformation CSS" is set, device pool calling >>>> transformation party is UNchecked >>>> >>>> on 1007 this is what i see on the screen +19162511006 and then just >>>> above the softkeys i see: >>>> from 1006 >>>> >>>> is that the correct configuration ? is that what i'm supposed to see? >>>> >>>> thanks >>>> >>>> Cris >>>> >>>> ah i didn't reply earlier because i was playing with calling and called >>>> transofrmation patterns >>>> and i just realized that if you have a called and calling transf pattern >>>> assigned to the same partition with a different pattern on version 7.1.3 , >>>> when you try to make a call the callmanager service crashes and it creates >>>> a >>>> core dump >>>> >>>> >>>> >>>> 2010/2/18 Otto Sanchez >>>> >>>> Hi Cris, >>>>> >>>>> You have to configure a calling party xform pattern (call routing-> >>>>> transformation pattern-> calling party transformation pattern menu), and >>>>> manipulate the calling number you are globalizing into a 4 digit >>>>> extension, >>>>> i.e.: >>>>> >>>>> The globalizaed number is +16178631002, so your pattern should look >>>>> like \+1617863.1XXX, in a partition not being used for any other function >>>>> ( >>>>> as Mustafa stated) with discard digits instruction: predot (calling party >>>>> transformation section) >>>>> >>>>> Then, create a css which contains the already created partition and >>>>> configure that css in the device pool or device "Calling Party >>>>> Transformation CSS" option. By default device pool configuration will >>>>> override the phone, so if you want to configure that css at the device >>>>> level, make sure the "Use Device Pool Calling Party Transformation CSS" is >>>>> unchecked, >>>>> >>>>> Thanks, >>>>> >>>>> >>>>> On Thu, Feb 18, 2010 at 5:26 PM, Mustafa wrote: >>>>> >>>>>> The calling party xform will take place on the called device. So if >>>>>> phone A calls phone B, then you should have the calling party xform >>>>>
Re: [OSL | CCIE_Voice] BACD
Hi, 1.- Change the number-of-hunt-grps parameter to 3 in the queue appl, and reload the application, ref. http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1055309 2.- Configure the 888 mailbox in your voicemail system and make sure you can call directly to the 888 ephone hunt group and that calls are distributed among the members, On Wed, Feb 24, 2010 at 7:13 AM, Omotayo wrote: > Hello, > > Working on the BACD for volume 1 in my lab > > i have the following. when i dial 1 i get the messgae "you have entered an > invalid entry" > > when i dial 2 i get thereis no mail box associated with ths extension > > Only dial0ng by extension and operator works > > what could be the issue > > > voice service voip > allow-connection h t s > > > > telephony-service > moh music-on-hold.au > > application > service queue flash:app-b-acd-2.1.2.2.tcl > param number-of-hunt-grps 2 > param aa-hunt1 888 > param aa-hunt2 999 > param aa-hunt10 100 > param queue-len 15 > param queue-manager-debugs 1 > service aa flash:app-b-acd-aa-2.1.2.2.tcl > paramspace english index 1 > paramspace english language en > paramspace english location flash: > param service-name queue > param handoff-string aa > param aa-pilot 800 > param welcome-prompt _bacd_welcome.au > param number-of-hunt-grps 3 > param dial-by-extension-option 3 > param second-greeting-time 60 > param call-retry-timer 15 > param max-time-call-retry 90 > param max-time-vm-retry 2 > param voice-mail 333 > > dial-peer voice 800 voip > service aa > destination-pattern 800 > session target ipv4:172.168.10.1 > incoming called-number 800 > dtmf-relay h245-alphanumeric > codec g711ulaw > no vad > > > ephone-hunt 1 sequential > pilot 999 > list 104, 103 > timeout 10, 10 > ! > ! > ephone-hunt 2 peer > pilot 888 > list 109, 105, 107, 111 > ! > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Prsenece DeskPhone control with SIP Phone Load
Chad, Nothing special at all, just make sure your sip phone and line has "Allow Control of Device from CTI" checked, that the cupc user in ucm has the sip phone associated and is member of the CTI Enable user group, and finally that the primary extension for that user matches your sip phone dn. If no luck, try restarting the ucm CTI Manager and CUPS services, HTH, On Mon, Feb 22, 2010 at 11:37 PM, Chad Stachowicz wrote: > Guys, > > Is there anythign special that needs to be done with a SIP phone to allow > CUPS Desk Phone control? I followed the same steps for an SCCP Phone which > works, and my SIP phone doesn't seem to, any suggestions? > > Thanks! > Chad > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Blocking a router pattern and PrecedenceMessage in Vol 1 Lab 5c Task 5.6
Hello, Since sccp phones will send digits in one go (en block dialing) when calling from directories, the pattern that you might have been created will not be matched (I assume 91900 with urgent priority) under that circumstances, so in that case, please create another 91900! pattern with similar characteristics to match calls made from the directories and play the same precedence level exceeded message, HTH, On Tue, Feb 23, 2010 at 11:05 AM, CCIETalk.com wrote: > I Am thinking that since dialing from call history isn't really a digit > press may be that's why? Let Otto shed some light :D > > Steve - it appears we are on the same workbook lab... When are you > attempting the lab? > > > On Tue, Feb 23, 2010 at 10:30 AM, Steve Denney (stdenney) < > stden...@cisco.com> wrote: > >> Thanks Otto, I noticed that behavior as well. >> >> >> >> Hijacking the thread - :) I also noticed something else interesting while >> working on this same task – I was able to “bypass” the 91900 block if I >> dialed from the call history directories. This was true on both CICP (SIP) >> and IP Blue (SCCP) clients. >> >> Tried the 91900 RP both with and without Urgent Priority checked; no >> change. >> >> Maybe I needed to add the “?” at the end of the pattern to match the whole >> string? Ran out of lab time before trying that... >> >> >> >> cheers, sd >> >> >> >> *From:* ccie_voice-boun...@onlinestudylist.com [mailto: >> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Otto Sanchez >> *Sent:* Tuesday, February 23, 2010 10:20 AM >> *To:* CCIETalk.com >> *Cc:* ccie_voice@onlinestudylist.com >> *Subject:* Re: [OSL | CCIE_Voice] Blocking a router pattern and >> PrecedenceMessage in Vol 1 Lab 5c Task 5.6 >> >> >> >> Hi, >> >> That's the expected behavior since the annunciator does not support sip >> phones, >> >> >> On Tue, Feb 23, 2010 at 10:44 AM, CCIETalk.com >> wrote: >> >> So i was working Vol 1 Lab 5c task 5.6 where it asks to block 9-1-900 >> calls. Appeared straight forward and I was able to get this done. However my >> SIP phones dont play this message and just disconnect the call after I dial >> 9-1-900 while my SCCP phones play the message. Any reason? >> >> -- >> www.ccietalk.com >> >> _______ >> For more information regarding industry leading CCIE Lab training, please >> visit www.ipexpert.com >> >> >> >> >> -- >> Regards, >> >> Otto Sanchez >> CCIE #25592 (Voice) >> Support Engineer - IPexpert, Inc. >> URL: http://www.IPexpert.com >> > > > > -- > www.ccietalk.com > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Blocking a router pattern and Precedence Message in Vol 1 Lab 5c Task 5.6
Hi, That's the expected behavior since the annunciator does not support sip phones, On Tue, Feb 23, 2010 at 10:44 AM, CCIETalk.com wrote: > So i was working Vol 1 Lab 5c task 5.6 where it asks to block 9-1-900 > calls. Appeared straight forward and I was able to get this done. However my > SIP phones dont play this message and just disconnect the call after I dial > 9-1-900 while my SCCP phones play the message. Any reason? > > -- > www.ccietalk.com > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 5c - Calls from BR2 CME SCCP / SIP to UCM HQ / BR1 SCCP Fails
=z9hG4bK23150B >> Remote-Party-ID: "br2 phn2" >> >;party=calling;screen=no;privacy=off >> From: "br2 phn2" >> >;tag=649C24-25D1 >> To: > >> Date: Tue, 23 Feb 2010 10:11:15 GMT >> Call-ID: 9c9e71ef-1f9a11df-80bea08f-6f27...@10.10.112.2 >> Supported: 100rel,timer,resource-priority,replaces >> Min-SE: 1800 >> Cisco-Guid: 2620869952-530190815-2159517839-116553341 >> User-Agent: Cisco-SIPGateway/IOS-12.x >> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, >> NOTIFY, INFO, REGISTER >> CSeq: 101 INVITE >> Max-Forwards: 70 >> Timestamp: 1266919875 >> Contact: >> Expires: 180 >> Allow-Events: telephone-event >> Content-Type: application/sdp >> Content-Disposition: session;handling=required >> Content-Length: 210 >> >> v=0 >> o=CiscoSystemsSIP-GW-UserAgent 3890 535 IN IP4 10.10.112.2 >> s=SIP Call >> c=IN IP4 10.10.112.2 >> t=0 0 >> m=audio 17616 RTP/AVP 18 >> c=IN IP4 10.10.112.2 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=ptime:20 >> >> Feb 23 10:11:15.773: //-1//SIP/Msg/ccsipDisplayMsg: >> Received: >> SIP/2.0 100 Trying >> Date: Tue, 23 Feb 2010 10:11:15 GMT >> From: "br2 phn2" >> >;tag=649C24-25D1 >> Allow-Events: presence >> Content-Length: 0 >> To: > >> Call-ID: 9c9e71ef-1f9a11df-80bea08f-6f27...@10.10.112.2 >> Via: SIP/2.0/UDP 10.10.112.2:5060;branch=z9hG4bK23150B >> CSeq: 101 INVITE >> >> >> Feb 23 10:11:15.777: //-1//SIP/Msg/ccsipDisplayMsg: >> Received: >> *SIP/2.0 503 Service Unavailable* >> Date: Tue, 23 Feb 2010 10:11:15 GMT >> Warning: 399 "Routing failed: ccbid=47 socket=10.10.112.2:5060" >> From: "br2 phn2" >> >;tag=649C24-25D1 >> Allow-Events: presence >> Content-Length: 0 >> To: >;tag=287642110 >> Call-ID: 9c9e71ef-1f9a11df-80bea08f-6f27...@10.10.112.2 >> Via: SIP/2.0/UDP 10.10.112.2:5060;branch=z9hG4bK23150B >> CSeq: 101 INVITE >> >> >> Feb 23 10:11:15.789: //-1//SIP/Msg/ccsipDisplayMsg: >> Sent: >> ACK sip:5...@10.10.210.10:5060 SIP/2.0 >> Via: SIP/2.0/UDP 10.10.112.2:5060;branch=z9hG4bK23150B >> From: "br2 phn2" >> >;tag=649C24-25D1 >> To: >;tag=287642110 >> Date: Tue, 23 Feb 2010 10:11:15 GMT >> Call-ID: 9c9e71ef-1f9a11df-80bea08f-6f27...@10.10.112.2 >> Max-Forwards: 70 >> CSeq: 101 ACK >> Allow-Events: telephone-event >> Content-Length: 0 >> >> >> BR2-RTR# >> Feb 23 10:11:24.345: //-1//SIP/Msg/ccsipDisplayMsg: >> Received: >> >> >> * >> * >> *Feb 23 10:11:24.345: //-1//SIP/Error/HandleUdpSocketReads: >> SIP Message incomplete, trashed* >> BR2-RTR# >> -- >> >> Thanks and Best Regards, >> Wael Agina >> >> ___ >> For more information regarding industry leading CCIE Lab training, please >> visit www.ipexpert.com >> >> > > > -- > www.ccietalk.com > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] + dialing
Hi Cris, Excelente!, espero estes bien!, I assume you have a cg xform pattern in a separate pt that is reachable from the "Calling Party Transformation CSS" css setting on the phone. Does that pattern look like this: \+1916251. with Discard Digit Instructions predot or Calling Party Transformation Mask , Please double check this, also, just curious what phone type are you using?, Thanks, On Mon, Feb 22, 2010 at 4:31 PM, Cristobal Priego wrote: > Hi Otto, > > Como estas mi estimado amigo ? como te trata la vida ? > > this is what i did > > 1006 dials 1007 --> hits translation pat , prefix +1 and uses Calling > party xform mask --> after translation we have +1916251 (+19162511006) > > on 1007 "Calling Party Transformation CSS" is set, device pool calling > transformation party is UNchecked > > on 1007 this is what i see on the screen +19162511006 and then just above > the softkeys i see: > from 1006 > > is that the correct configuration ? is that what i'm supposed to see? > > thanks > > Cris > > ah i didn't reply earlier because i was playing with calling and called > transofrmation patterns > and i just realized that if you have a called and calling transf pattern > assigned to the same partition with a different pattern on version 7.1.3 , > when you try to make a call the callmanager service crashes and it creates a > core dump > > > > 2010/2/18 Otto Sanchez > > Hi Cris, >> >> You have to configure a calling party xform pattern (call routing-> >> transformation pattern-> calling party transformation pattern menu), and >> manipulate the calling number you are globalizing into a 4 digit extension, >> i.e.: >> >> The globalizaed number is +16178631002, so your pattern should look like >> \+1617863.1XXX, in a partition not being used for any other function ( as >> Mustafa stated) with discard digits instruction: predot (calling party >> transformation section) >> >> Then, create a css which contains the already created partition and >> configure that css in the device pool or device "Calling Party >> Transformation CSS" option. By default device pool configuration will >> override the phone, so if you want to configure that css at the device >> level, make sure the "Use Device Pool Calling Party Transformation CSS" is >> unchecked, >> >> Thanks, >> >> >> On Thu, Feb 18, 2010 at 5:26 PM, Mustafa wrote: >> >>> The calling party xform will take place on the called device. So if >>> phone A calls phone B, then you should have the calling party xform >>> configured on the phone B to manipulate the phone A calling number. It >>> will be invoked when phone B gets the call and looks into configured >>> xform css/pt for a match. Make sure that you are matching and the pt/css >>> is also exclusive. >>> >>> Device pool xform overrides xform directly configured on the phone. >>> >>> -- Mustafa >>> >>> >>> Cristobal Priego wrote: >>> > Hello, >>> > >>> > I know this has been posted quite some times. >>> > I'm configuring + dialing on my lab and I think I have a problem on >>> > internal calls, when I call a 4 digit extension, the called number >>> > shows the globalized number on the display, I'm not able to get it to >>> > localize the number to 4 digits. I see the +... on the missed calls. I >>> > have the calling party transformation set on the phone itself >>> > how does this actually work ? how is the calling party transformation >>> > invoked at the destination device ? >>> > >>> > thanks >>> > >>> >>> > >>> > ___ >>> > For more information regarding industry leading CCIE Lab training, >>> please visit www.ipexpert.com >>> > >>> >>> ___ >>> For more information regarding industry leading CCIE Lab training, please >>> visit www.ipexpert.com >>> >> >> >> >> -- >> Regards, >> >> Otto Sanchez >> CCIE #25592 (Voice) >> Support Engineer - IPexpert, Inc. >> URL: http://www.IPexpert.com >> > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE Voice Vol 1 Lab 5c - Transcoder
Hi, Would you please send us your 3725 configuration?, Thanks, On Sun, Feb 21, 2010 at 10:27 PM, CCIETalk.com wrote: > Well it appears that NM-HDV doesn't support transcoding either :\ > > > On Sun, Feb 21, 2010 at 9:55 PM, CCIETalk.com wrote: > >> I added NM-HDV with pvdm-12 but I still see the same issue. >> >> >> On Sun, Feb 14, 2010 at 3:51 PM, Otto Sanchez wrote: >> >>> Hi, >>> >>> I don't think the AIM-VOICE-30 supports transcoding or conferencing but >>> voice termination services only, so in this case you may need to install a >>> NM in your 3725 to move on, >>> >>> Thanks, >>> >>> On Sun, Feb 14, 2010 at 12:07 PM, CCIETalk.com wrote: >>> >>>> I was working through lab 5c and came across the task where I had to >>>> configure a transcoder. I am using a 3725 with AIM-30 >>>> >>>> - one voice pri with 3 channels >>>> - one data T1 >>>> >>>> I try to create the dspfarm profile and get this erro >>>> >>>> HQ-RTR(config-dspfarm-profile)#codec ? >>>> % Unrecognized command >>>> >>>> Any idea? >>>> >>>> -- >>>> www.ccietalk.com >>>> >>>> ___ >>>> For more information regarding industry leading CCIE Lab training, >>>> please visit www.ipexpert.com >>>> >>>> >>> >>> >>> -- >>> Regards, >>> >>> Otto Sanchez >>> CCIE #25592 (Voice) >>> Support Engineer - IPexpert, Inc. >>> URL: http://www.IPexpert.com >>> >> >> >> >> -- >> www.ccietalk.com >> > > > > -- > www.ccietalk.com > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Device Mobility with Local Route Groups
Hi Jeff, You are right, phones at the roaming location will get the LRG configuration from the roaming DP, we cannot avoid this. However, a workaround to your scenario may be to implement + dialing, in which you globalize the caller input (translation patterns) depending on his/her dialing habits and localize it for the outgoing gateway (cd xform patterns), in that way, a US user roaming to UK will be allowed to use the LRG route patterns (using US dialing habits) therefore using local UK resources. The drawback here is that this won't be a cost effective solution, in which case you will have to implement teho/location specific route patterns to route calls out and save costs, Hope this make sense, On Fri, Feb 19, 2010 at 6:23 PM, Jeff Cotter wrote: > Having trouble understanding how this is supposed to function without > Site Specific Route Patterns and Route List/Group. If using only non-site > specific route patterns pointed to local route group I see no value in this. > Of course I am probably mistaken…hence this message. Scenario below > > > > > > No location specific Route Patterns exist…all route patterns point to local > route group. > > > > Device roams from HQ to BR1 DMG is US for both. (CSS for both home and > Roaming device pool is CSS-LD). Roaming sensitive settings are applied as > well as Mobility Settings.which means device will use local route group > defined in BRI device pool for all calls based on CSS-LD. User dials > 95551212..No problem here since calls will now be sent out BR1-GW as > expected with correct digits assuming predot is applied via Called > Transformation. > > > > However take the same scenario above except the DMG is now changed to UK > for roaming phone. Mobility settings are no longer applied meaning CSS does > not change. The purpose for this is supposed to be that user does not want > to have to dial differently when in a new country. However Local Route Group > is still obtained from roaming Device Pool. Since all Route Patterns point > to local route group and local route group is now UK…. ALL calls will now be > directed to UK gw. Call will fail as digits PSTN is expecting will be > incorrect. > > > > What is the excepted way to get calls to NOT route out local gateway but > traverse the WAN and go out US gateway. I can’t think of how to do this > without using location specific Route Patterns and Route Lists which now > defeats the purpose of the Local Route Group concept. ARRRH > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Barge
If you are barging br1phn2 the br1 conference resources are used, not the hq ones. Also make sure that br1phn2 has privacy off by the use of the privacy button or has privacy off in the device configuration, Finally, make sure that when cbarging from the shared line button, your hq phone config has the cbarge setting configured, On Sat, Feb 20, 2010 at 4:24 PM, Omotayo wrote: > Hello, > working on Volume 1 lab 8 > IOS conference has been cnfigured on hq router > when i tried with question 8.1 i used hq phones as bri phone of the > question and vice versa > on pressing the button on br1 phone 2 when in In Use Remote, i still see > the barge softkey and it gives the message No Conference Bridge > CBarge was enabled on the service parameter > > Below is a proof the conference bridge on the hq router is working > > HQ-RTR#sh sccp connections > sess_idconn_idstype mode codec ripaddr rport sport > 33557433 33554446 conf sendrecv g729b 192.168.3.1616386 16524 > 33557433 3355 conf sendrecv g729b 192.168.3.1218674 17838 > 33557433 33554442 conf sendrecv g711u 192.168.3.1828554 18950 > Any ideas on what the issue is > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUC Not Licensed For VPIM
Hi Scott, Currently Proctorlabs gear is not licensed for VPIM, so we cannot make test with the feature for the time being, On Sat, Feb 20, 2010 at 7:46 PM, scott carruthers wrote: > When I attempt to add a VPIM location is Unity Connection I receive the > following license error. Are the proctorlabs servers not licensed for > VPIM? Anyone attempt VPIM in these labs yet? > > Status [image: error] The requested operation would result in a license > violation. [image: error] Unable to create VPIM Location > > *S*ave New VPIM Location Display Name* Dtmf Access ID* Partition cuc7-pub > Partition Domain Name* IP Address* Remote phone prefix > *S*ave Fields marked with an asterisk (*) are required. > > The Demo license info show nothing for VPIM: > > SERVER this_host ANY > VENDOR cisco > INCREMENT LicVoicePortsMax cisco 7.0 permanent 2 HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic0 \ > dummyPak" SIGN=A3DF5BBED8B0 > INCREMENT LicSubscribersMax cisco 7.0 permanent 10 HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic1 \ > dummyPak" SIGN=FA226A483396 > INCREMENT LicVMISubscribersMax cisco 7.0 permanent 10 HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic2 \ > dummyPak" SIGN=22D6A4F63854 > INCREMENT LicAdvancedUserMax cisco 7.0 permanent 10 HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic3 \ > dummyPak" SIGN=85B5BD2CDF32 > INCREMENT LicRealspeakSessionsMax cisco 7.0 permanent 2 HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic4 \ > dummyPak" SIGN=24848F662AEC > INCREMENT LicServerBackend cisco 7.0 permanent 1 HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic5 \ > dummyPak" SIGN=6750CF4C26B4 > INCREMENT LicIMAPSubscribersMax cisco 7.0 permanent 10 HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic6 \ > dummyPak" SIGN=0A5E3C90C67A > INCREMENT LicUnityVoiceRecSessionsMax cisco 7.0 permanent 2 \ > HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic7 \ > dummyPak" SIGN=12E962E6B592 > INCREMENT LicServerVoiceRec cisco 7.0 permanent 1 HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic8 \ > dummyPak" SIGN=5C6FF1C641AE > INCREMENT LicMaxMsgRecLenIsLicensed cisco 7.0 permanent 1 HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic9 \ > dummyPak" SIGN=573BA6B413B6 > INCREMENT LicRegionIsUnrestricted cisco 7.0 permanent 1 HOSTID=ANY \ > > NOTICE="FOR_DEMO_ONLY.lic10 \ > dummyPak" SIGN=40EBACAE87D8 > > > > -- > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up > now. <http://clk.atdmt.com/GBL/go/201469229/direct/01/> > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] LAB 3A intital config issues
Hi Randall, Please try with lab 4a initial load for br2 rtr in the meanwhile, you might see a little telephony-service configuration there, do a no telephony-service and move forward, Let me research this further and report it internally if needed, Thanks, On Sat, Feb 20, 2010 at 10:55 PM, Randall Crumm < randall.cr...@harmonicinc.com> wrote: > HI, > I have been racking my brain for a while now and give up. > I loaded the lab 3A initial config and it was missing OSPF config, VLAN > and interace configuration...And I am not sure what else. I reverted the > pod twice and added some config in but I am still not able to register my > phone and http to the router for GUI access. > The problem this is causing me is access from my PC so I can register my > phone on my laptop. > > Can someone look into this as why the config is missing so much? > Thanks, > Randall > > > > From: ccie_voice-boun...@onlinestudylist.com [ > ccie_voice-boun...@onlinestudylist.com] On Behalf Of > ccie_voice-requ...@onlinestudylist.com [ > ccie_voice-requ...@onlinestudylist.com] > Sent: Saturday, February 20, 2010 9:00 AM > To: ccie_voice@onlinestudylist.com > Subject: CCIE_Voice Digest, Vol 48, Issue 115 > > Send CCIE_Voice mailing list submissions to >ccie_voice@onlinestudylist.com > > To subscribe or unsubscribe via the World Wide Web, visit >http://onlinestudylist.com/mailman/listinfo/ccie_voice > or, via email, send a message with subject or body 'help' to >ccie_voice-requ...@onlinestudylist.com > > You can reach the person managing the list at >ccie_voice-ow...@onlinestudylist.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of CCIE_Voice digest..." > > > Today's Topics: > > 1. VOL2 LAB 6 (kavi ten) > 2. VOL2 LAB 6 UCME with UC (kavi ten) > > > -- > > Message: 1 > Date: Sat, 20 Feb 2010 18:55:30 +0400 > From: kavi ten > Subject: [OSL | CCIE_Voice] VOL2 LAB 6 > To: ccie_voice@onlinestudylist.com > Message-ID: ><2115142c1002200655v44a9349ak12f9e2ab1058...@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Guys, > > I got stuck at the Teho dialing Question 4.5 > I have the CME dial-peer configured & it works fine as the call to HQ/BR1 > hits the GK > I also have the translation pattern configured to match the incoming call > same as in the PG > 1#39001972XXXFOR 330 with called no xformation > Still the call is not going to the BR1 gw & same the case with HQ calls > > The call then dial from BR2 PRI as per the preference. > > I can not understand how this should work as I have all parameter > configured. > Can some one guide me through how to go about this. > > Thanks. > -- next part -- > An HTML attachment was scrubbed... > URL: > http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100220/28296b94/attachment-0001.htm > > -- > > Message: 2 > Date: Sat, 20 Feb 2010 19:58:10 +0400 > From: kavi ten > Subject: [OSL | CCIE_Voice] VOL2 LAB 6 UCME with UC > To: ccie_voice@onlinestudylist.com > Message-ID: ><2115142c1002200758j17a99b37t604b6ccdec7b8...@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi Guys, > > When have the done the configuration as per the Guide, but the BR2 Phn1 > phone on busy /noan , the incoming call gets disconnected. > I checked the debug voice ccapi inout , which says disconnect cause of 19 > , > means no answer from the UC. > > Has any one come across this problem. > > Thanks > -- next part -- > An HTML attachment was scrubbed... > URL: > http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100220/73fa1cc5/attachment-0001.htm > > -- > > ___ > CCIE_Voice mailing list > CCIE_Voice@onlinestudylist.com > http://onlinestudylist.com/mailman/listinfo/ccie_voice > > > End of CCIE_Voice Digest, Vol 48, Issue 115 > *** > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 5A Task 5.2 - \"Cannot reach thenumber\" display
Hi Wael, Please double check the following: -. br1ph2 is in br1 device pool -. mgcp br1 gw is correctly registered to the ucm -. br1 route group contains the br1 gw device -. br1 device pool points to br1 route group in the Local Route Group setting -. route list points to the standard local route group -. 911 route pattern points to the created rl, contains no called number transformation and in a partition reachable for all the phones I would also suggest to test your br1 gw functioning without any local route group configuration but a single route pattern pointing to br1 gw, in that way you make sure your gw is good and focus on the lrg settings, HTH, On Sun, Feb 21, 2010 at 8:35 AM, Wael Agina wrote: > Hi Steve, > > Yes i mean BR1 PH2. > Actually I restarted the CUCM both Pub and Sub but still not able to dial > 911. > I run DNA for that phone but everything normal and call fwd to the local RL > which points to standard local RG , meaning BR1 GW. > However on the GW I couldnt get the call - the debug isdn q931 doesnt > produce any output. > > The rack time ended and i may repeat it tomorrow and see. > > Thanks for your reply. > > Regards, > Wael Agina > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper Issue when calling India
e > > visit www.ipexpert.com > > > > > > -- next part -- > > An HTML attachment was scrubbed... > > URL: > > > http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100218/28a2d00 > > c/attachment-0001.htm > > > > -- > > > > Message: 4 > > Date: Thu, 18 Feb 2010 21:52:45 +1000 > > From: Roger Henderson > > Subject: [OSL | CCIE_Voice] UCCX Scripting > > To: ccie_voice@onlinestudylist.com > > Message-ID: > > > > Content-Type: text/plain; charset="iso-8859-1" > > > > Hey Everyone, > > > > What is the best resource(s) to learn the various UCCX scripting methods > > needed for the lab? Does anyone have any good resources online? How > > complicated is it likely to get for the lab and how much time should we > > dedicate to this? > > > > Thanks, > > > > Roger > > -- next part -- > > An HTML attachment was scrubbed... > > URL: > > > http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100218/8c8e3e0 > > 5/attachment-0001.htm > > > > -- > > > > Message: 5 > > Date: Thu, 18 Feb 2010 07:24:16 -0500 > > From: "Brian Valentine" > > Subject: Re: [OSL | CCIE_Voice] UCCX Scripting > > To: "'Roger Henderson'" , > > > > Message-ID: <001b01cab095$4abdf860$e039e9...@com> > > Content-Type: text/plain; charset="us-ascii" > > > > Roger, > > > > > > > > I think the simplest explanation of this is that 1) this exam is not a > CCIE > > in UCCX Scripting. And 2) some scripting could be on the exam. > > > > > > > > I know that's obscure. That's because no one can say exactly what you > will > > see in the lab. Let me also say that the IPExpert materials are > sufficient > > in preparing you for the lab exam. I would be prepared to edit an > existing > > script according to very specific requirements. I would not expect to > have > > to create a very detailed script from scratch. If you did get something > > like that, I would think you may consider skipping those 3 or 4 points. > > Spending a couple of hours for 3 or 4 points wouldn't be worth it in my > > opinion. > > > > > > > > I would also be familiar with the "built in" scripts that come on the > UCCX > > server out of the box. There is an AA script and an ICD script that > could > > be used as a starting point in case you need to build something simple > from > > scratch. This is my opinion, but I can't imagine you would see anything > > more time intensive than that on the exam. > > > > > > > > These are the things that IPExpert has you doing in their workbooks. > > > > > > > > Brian > > > > > > > > > > > > > > > > > > > > From: ccie_voice-boun...@onlinestudylist.com > > [mailto:ccie_voice-boun...@onlinestudylist.com] > On Behalf Of Roger Henderson > > Sent: Thursday, February 18, 2010 6:53 AM > > To: ccie_voice@onlinestudylist.com > > Subject: [OSL | CCIE_Voice] UCCX Scripting > > > > > > > > Hey Everyone, > > > > > > > > What is the best resource(s) to learn the various UCCX scripting methods > > needed for the lab? Does anyone have any good resources online? How > > complicated is it likely to get for the lab and how much time should we > > dedicate to this? > > > > > > > > Thanks, > > > > > > Roger > > > > No virus found in this incoming message. > > Checked by AVG - www.avg.com > > Version: 9.0.733 / Virus Database: 271.1.1/2693 - Release Date: 02/17/10 > > 02:35:00 > > > > -- next part -- > > An HTML attachment was scrubbed... > > URL: > > > http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100218/6e7c253 > > a/attachment-0001.htm > > > > -- > > > > Message: 6 > > Date: Thu, 18 Feb 2010 08:57:49 -0500 > > From: "Pulos, Greg" > > Subject: Re: [OSL | CCIE_Voice] UCCX Scripting > > To: Roger Henderson , > > "ccie_voice@onlinestudylist.com" > > > > Message-ID: > > > > > Content-Type: text/plain; charset="us-ascii" > > > > Knowing CCX scripting is vital if you acutally want to be able to provide > > solutions to customers and to troubleshoot CRS application problems. > > > > Please see the link below for more information on CCX 7.01 scripting. > > > > > http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/cr > > s/express_7_0/user/guide/uccx70edgs.pdf > > > > Thank you. > > > > greg > > > > > > -Original Message- > > From: ccie_voice-boun...@onlinestudylist.com > > [mailto:ccie_voice-boun...@onlinestudylist.com] > On Behalf Of Roger Henderson > > Sent: Thursday, February 18, 2010 6:53 AM > > To: ccie_voice@onlinestudylist.com > > Subject: [OSL | CCIE_Voice] UCCX Scripting > > > > Hey Everyone, > > > > What is the best resource(s) to learn the various UCCX scripting methods > > needed for the lab? Does anyone have any good resources online? How > > complicated is it likely to get for the lab and how much time should we > > dedicate to this? > > > > > > Thanks, > > > > Roger > > > > > > > > -- > > > > ___ > > CCIE_Voice mailing list > > CCIE_Voice@onlinestudylist.com > > http://onlinestudylist.com/mailman/listinfo/ccie_voice > > > > > > End of CCIE_Voice Digest, Vol 48, Issue 101 > > *** > > > > __ Information from ESET NOD32 Antivirus, version of virus > signature > > database 4872 (20100216) __ > > > > The message was checked by ESET NOD32 Antivirus. > > > > http://www.eset.com > > > > > > > > > > __ Information from ESET NOD32 Antivirus, version of virus > signature > > database 4872 (20100216) __ > > > > The message was checked by ESET NOD32 Antivirus. > > > > http://www.eset.com > > > > > > ___ > > For more information regarding industry leading CCIE Lab training, please > > visit www.ipexpert.com > > > > > > __ Information from ESET NOD32 Antivirus, version of virus > signature > > database 4872 (20100216) __ > > > > The message was checked by ESET NOD32 Antivirus. > > > > http://www.eset.com > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Help with IPExpert Material
Hi Chad, The vlans and ip addressing for all pods are the same, try downloading the documents 1.- and 2.- in the "CCIE Vocie 5-Day ILT Labs" section, On Thu, Feb 18, 2010 at 5:46 PM, Chad Stachowicz wrote: > I'm currently just starting my firswt CCIE V3 blueprint session, and I have > my proctor guide and workbook, and I am familiar with IP Expert and can log > into device and such. However the workbook does not have the IP Address and > VLan's for each POD laid out anywhere nor do i see it anywhere in the > proctor labs itself. It says in the workbook to get it from the > configuration files on pexpert.com, however when i try to download the > CCIE V3 Configuration files, it says it is unable to process that request. > Any ideas?! > > Thanks! > Chad > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] + dialing
Hi Cris, You have to configure a calling party xform pattern (call routing-> transformation pattern-> calling party transformation pattern menu), and manipulate the calling number you are globalizing into a 4 digit extension, i.e.: The globalizaed number is +16178631002, so your pattern should look like \+1617863.1XXX, in a partition not being used for any other function ( as Mustafa stated) with discard digits instruction: predot (calling party transformation section) Then, create a css which contains the already created partition and configure that css in the device pool or device "Calling Party Transformation CSS" option. By default device pool configuration will override the phone, so if you want to configure that css at the device level, make sure the "Use Device Pool Calling Party Transformation CSS" is unchecked, Thanks, On Thu, Feb 18, 2010 at 5:26 PM, Mustafa wrote: > The calling party xform will take place on the called device. So if > phone A calls phone B, then you should have the calling party xform > configured on the phone B to manipulate the phone A calling number. It > will be invoked when phone B gets the call and looks into configured > xform css/pt for a match. Make sure that you are matching and the pt/css > is also exclusive. > > Device pool xform overrides xform directly configured on the phone. > > -- Mustafa > > > Cristobal Priego wrote: > > Hello, > > > > I know this has been posted quite some times. > > I'm configuring + dialing on my lab and I think I have a problem on > > internal calls, when I call a 4 digit extension, the called number > > shows the globalized number on the display, I'm not able to get it to > > localize the number to 4 digits. I see the +... on the missed calls. I > > have the calling party transformation set on the phone itself > > how does this actually work ? how is the calling party transformation > > invoked at the destination device ? > > > > thanks > > > > > > ___ > > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HQ>>BR2>>>CUE getting fast busy
Hi, Would you please provide a sh sccp output, phone type and transcoding configuration from br2?, Thanks, On Sat, Feb 13, 2010 at 11:54 PM, vccie2010 wrote: > HQ>>BR2>>>CUE getting fast busy > > CUE is inte to CCM, it works when the region between HQ and BR2 is setup as > G711. I do have transcodes reg and in MRGL of HQ and BR2. Hq mrgl has tran > config on hq router and br2 mrgl has tran config on br2 router. CUE CTI RP > and Ports are in BR2 DP. Any clue pls? > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] sending CUCM traces to syslog
Hi, You have the option to export the trace files from the rtmt (and also from the cli to an sftp server with the file get command), but not to a syslog server in real time, which can in turn be used to send ucm alarms, Thanks, On Sun, Feb 14, 2010 at 6:33 PM, t n wrote: > Hello, > > Is there a way to send traces directly from CUCM to syslog? Looking at > traces via the CLI is really cumbersome. > > The syslog agent in enterprise parameters is not it. > -- > Thanks. > > tnn314.wordpress.com > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE Voice Vol 1 Lab 5c - Transcoder
Hi, I don't think the AIM-VOICE-30 supports transcoding or conferencing but voice termination services only, so in this case you may need to install a NM in your 3725 to move on, Thanks, On Sun, Feb 14, 2010 at 12:07 PM, CCIETalk.com wrote: > I was working through lab 5c and came across the task where I had to > configure a transcoder. I am using a 3725 with AIM-30 > > - one voice pri with 3 channels > - one data T1 > > I try to create the dspfarm profile and get this erro > > HQ-RTR(config-dspfarm-profile)#codec ? > % Unrecognized command > > Any idea? > > -- > www.ccietalk.com > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 - Lab 5 - Task 5.5 / Setting up TEHO for 212 calls from BR1 through HQ H.323 GW
Hey Matthew, That's the expected behavior since h.323 gateways don't support the + character sending, so if you still want to send that character out to the pstn you should handle it from the router itself (for example voice translation rules), You will find more information in: http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a03rp.html#wp1166491 Thanks, On Sun, Feb 14, 2010 at 2:38 PM, Berry, Matthew J. wrote: > Vol 1 - Lab 5 - Task 5.5 / Setting up TEHO for 212 calls from BR1 through > HQ H.323 GW > > I can get TEHO to work when dialing a 617 area code number from HQ Phone 2, > routing the call over the WAN, out the BR1 MGCP gateway. It works like a > charm. It appends the + which seems to come from the 9.1617XXX > translation pattern in PT-HQ-PSTN. > > Problem: I cannot get the + to be sent out when setting up TEHO for 212 > area code calls from BR1 through HQ's H.323 GW. All of my settings for the > BR1 site are identical to the HQ site. > > My only guess is that TEHO over WAN and out the BR1 MGCP gateway is using > MGCP and not H.323. > > I can append a + using a dial-peer on the H.323 gateway, but I'm not sure > if that is the best way to do it. > > It seems like Ben was saying that however you produce the end results in > the lab is all that matters. > > What do you guys think? Am I missing something? > > Digital Footprint: > Skype: ciscovoiceguru > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] wierd h323 gateway problem lab7
Hi Sean, Your incoming pots dial peer and voip outgoing dial peers (and perhaps your voice port) don't have any called number translation, so from your h323 gw config in ucm set significant digits to 4, also make sure the h.323 interface in your br2 gw has the h323-gateway voip interface and h323-gateway voip bind scraddr commands configured, Thanks, -. On Sat, Feb 13, 2010 at 12:35 PM, sean hurricane wrote: > Wierd H323 gateway problem PSTN inbound call to BR2 phones does not > ring phones.. calls come in and i verify using debug isdn q931 but phone > does not ring, if i use csim start from gateway to call phone, it rings, so > that eliminates partition and inbound CSS issue. phone can successfully make > outbound calls. > > BR1#sh run | s dial-peer > dial-peer voice 1000 voip > destination-pattern 1...$ > voice-class codec 1 > voice-class h323 1 > session target ipv4:10.10.210.11 > dtmf-relay h245-alphanumeric > ip qos dscp cs3 signaling > no vad > dial-peer voice 1005 voip > preference 1 > destination-pattern 1...$ > voice-class codec 1 > voice-class h323 1 > session target ipv4:10.10.210.10 > dtmf-relay h245-alphanumeric > ip qos dscp cs3 signaling > no vad > dial-peer voice 10 pots > destination-pattern 9%911 > incoming called-number . > no digit-strip > direct-inward-dial > port 0/1/1:23 > dial-peer voice 15 pots > translation-profile outgoing 7digitani > destination-pattern 9[2-9].. > port 0/1/1:23 > forward-digits 7 > dial-peer voice 20 pots > translation-profile outgoing TEHO-BR2 > destination-pattern 0114420T > port 0/1/1:23 > forward-digits all > dial-peer voice 25 pots > incoming called-number . > direct-inward-dial > BR1# > BR1# > BR1# > BR1#csim start 1001 > csim: called number = 1001, loop count = 1 ping count = 0 > > Feb 13 17:44:24.956: //-1//DPM/dpMatchPeersCore: >Calling Number=, Called Number=1001, Peer Info Type=DIALPEER_INFO_SPEECH > Feb 13 17:44:24.956: //-1//DPM/dpMatchPeersCore: >Match Rule=DP_MATCH_DEST; Called Number=1001 > Feb 13 17:44:24.956: //-1//DPM/dpMatchPeersCore: >Result=Success(0) after DP_MATCH_DEST > Feb 13 17:44:24.956: //-1//DPM/dpMatchPeers: >Result=SUCCESS(0) >List of Matched Outgoing Dial-peer(s): > 1: Dial-peer Tag=1000 > 2: Dial-peer Tag=1005 > csim err csimDisconnected recvd DISC cid(97) > csim: loop = 1, failed = 1 > csim: call attempted = 1, setup failed = 1, tone failed = 0 > > BR1# > BR1# > BR1# > BR1# > BR1# > BR1#csim start 1002 > csim: called number = 1002, loop count = 1 ping count = 0 > > Feb 13 17:44:43.460: //-1//DPM/dpMatchPeersCore: >Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH > Feb 13 17:44:43.460: //-1//DPM/dpMatchPeersCore: >Match Rule=DP_MATCH_DEST; Called Number=1002 > Feb 13 17:44:43.460: //-1//DPM/dpMatchPeersCore: >Result=Success(0) after DP_MATCH_DEST > Feb 13 17:44:43.460: //-1//DPM/dpMatchPeers: >Result=SUCCESS(0) >List of Matched Outgoing Dial-peer(s): > 1: Dial-peer Tag=1000 > 2: Dial-peer Tag=1005 > csim err csimDisconnected recvd DISC cid(98) > csim: loop = 1, failed = 1 > csim: call attempted = 1, setup failed = 1, tone failed = 0 > > BR1# > BR1# > BR1# > BR1# > BR1# > BR1#ping 10.10.210.11 > > Type escape sequence to abort. > Sending 5, 100-byte ICMP Echos to 10.10.210.11, timeout is 2 seconds: > ! > Success rate is 100 percent (5/5), round-trip min/avg/max = 56/56/60 ms > BR1#ping 10.10.210.10 > Type escape sequence to abort. > Sending 5, 100-byte ICMP Echos to 10.10.210.10, timeout is 2 seconds: > ! > Success rate is 100 percent (5/5), round-trip min/avg/max = 56/57/60 ms > BR1# > > any thots? > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Q5.1
Hi, I'd say yes if the negotiated codec between the gw and ucm is g.729r8 (default codec for the dial peer), so make sure the gw is using that codec when talking to ucm or at least it's in the list for the voice class codec used by the dial peer, Thanks, On Wed, Feb 10, 2010 at 2:07 PM, vccie2010 wrote: > Per the PG solutions the "dspfarm profile 1 transcode" does not show "codec > g729r8" it only shows "g279ar8 and g729abr8" don't we need "codec G729r8" > statement here since the traffic coming from UCM across GK will be G729r8 > ??? > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX SQL question
Hi Cris, When you backup the uccx using the native embedded backup utility in the appadmin page, you backup both configuration and historical data from uccx, Regarding the upgrade to the database, uccx 7.0X currently supports sql/msde 2000 only, Thanks, On Tue, Feb 9, 2010 at 5:37 PM, Cristobal Priego wrote: > Guys I have a question related to UCCX and SQL > > when you run a backup on UCCX in regards to SQL (backups / reindex) is that > included in the backup ? > > What are the steps that i need to follow to get the versions upgraded to > at least SQL05? > > thanks > > Cris > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call park – lab 8A task 8 .2
Hi Roger, I think you are referring to the transfer on hook feature, this will work only if you go on hook with your handset to complete the transfer and not when you press the end softkey, in that case you should complete the transfer using the transfer softkey again, Thanks, On Fri, Feb 12, 2010 at 5:31 AM, Roger Källberg wrote: > Hi guys, > > I have one question about the directed call park feature. I get different > behavior if I try to do a directed call park when I’m off hook by lifting > the hand set or if I’m off hook by simply pressing the “New Call” soft key, > ie using the speaker. > > When I go off hook by lifting the hand set all work as expected. I can do a > transfer, call 8555 (call park dn) and then go on-hook by placing the hand > set back, as the requirement in task 8.2 the lab tells. > > But when I go off hook by pressing the soft key “New Call”, to use the > speaker phone, and then do the same sort of directed call park, by pressing > Transfer, dial 8555 (call park dn) and then when the park slot answers I > try to go on hook by pressing the End Call soft key. That results in the > original call not being sent to the park slot dn, as I would expect it to > be. What happens instead is that the second calls, the one to the park slot > dn, will be dropped and then I can resume the original placed call by > pressing “Resume”. > > I have tried this on both my SCCP and SIP 7962. I’m not using my own HW, > I’m connected to PL with a HW vpn and have 5 7962 phones at my home lab > room. > > Anyone else that has seen this odd behavior or maybe it’s just as it’s > designed to work? > > Brgds, > *Roger Källberg* > Consultant > Cygate AB > Eric Perssons väg 21, SE-217 62 MALMÖ > > Direkt: +46108787498 > Växel: +46108787400 > roger.kallb...@cygate.se > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] BW Calculation Question
Hi Jeff, I would calculate and configure on the routers exactly the amount of voice bandwidth (including L2) required for the number of calls you planned to traverse across the whole path, in that way you will set what you will actually consume for each link. In UCM will always be the same L3 value, no matter the L2 overhead if you are using UCM CAC. If your topology includes multiple paths that the voice stream could use to get from one end to the other and different L2 technologies, I would do the same, but this time reserving the amount of bandwidth that could be used for voice for each link, and using rsvp + llq, you should take into consideration the srnd best practices in terms of the bw calculation in these cases, Thanks, On Mon, Feb 8, 2010 at 12:02 PM, Jeff Garvas wrote: > > I understand the calculation for ethernet / ppp / MLP / FR / ATM -- but > what about a call that traverses multiple paths? > > What if a call between two voip phones traverses an FRF.8 ATM to F/R > environment running MLPPP (with ethernet on each end). Do you simply > consider the call's bandwidth to be the worst case scenario along the path? > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Dial Peer Voice Pots Longest Match
Hi Kev, In that case yes because of the T at the end and if the dialed number begins with 001, if it wasn't there the dp with the 00 destination pattern will always be matched, Thanks, On Fri, Feb 5, 2010 at 2:59 PM, Ken Kov wrote: > Are pots dial-peers subject to longest match? > > So 001T matched will always be chosen ahead of 00T? > > Thanks, > Ken > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SNR Issue
Hi Arun, How long does it takes for the call to reach the rd number once you call the 5002 device? There shouldn't be any other delay other than the "delay before ringing timer" parameter to route the call to the rd number, Since the rerouting css is used to route both calls, the only difference I see here is that you could be using a rp with LRG configuration and the calls might be routed slightly different depending on the origin (whether 1002 calls 5002 or 5002 is sending the call to the rd number), if that's the case, please isolate the problem and configure a different rp with the corresponding rd number within a new partition, and point it to the hq gw or rl containing only hq rg gateways, in consequence change the rerouting css to a new one containing only the new created pt, Also, did you restart the servers?, configured everything from scratch?, etc, Thanks, On Fri, Feb 5, 2010 at 12:37 AM, Arun Kumar wrote: > Hi All, > > Any clue will be very helpful on this, during lab 5C I made it working but > after that its not working anymore every time same issue, please let me know > what to check. > > Thanks > Arun > > > On Thu, Feb 4, 2010 at 11:59 PM, Arun Kumar wrote: > >> Hi All, >> >> I run into this SNR issue: >> >> wired thing is that I can send calls to Mobile after hitting the mobility >> softkey >> and when I call from mobile line to BR1 Ph2 I see caller id as 5002 (HQ >> Phone 2) >> and when I hangup on mobile able to pickup from the desk phone >> but when I call the desk phone my mobile don't rings same time (Delay >> Before Ringing Timer = 0) every time I need to hit the mobility softkey >> only >> >> any clue to look into this ? >> >> Thanks >> Arun >> > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration rejected
Hi Jeff, Kavi, A cause for this error is that the auto registration number range has been exhausted (i.e. at some point in time the dn numbers have been assigned), please increase that range and try again, also, make sure your pub and sub have different ranges, If still with problems, I would think that the sub server is not getting the auto-registration settings from cisco unified cm configuration section, so as Roger mentioned this might be a replication issue, Then try to force a database replication from the pub server's cli, if it still doesn't work, please send us the sub call manager service traces when the phone is auto registering, Thanks, On Thu, Feb 4, 2010 at 3:04 PM, Jeff Price (jeffpric) wrote: > Anyone have any ideas? I’m still stuck here. I can’t find anything on > google relating to this error, other than IP communicator stuff. I don’t > believe its DB replication, because the Phones have replicated to the SUB > when I am logged into the CM Administration page of the SUB. However, I am > in a lab that has restricted access to CLI, OS Admin Page, and DRS page, so > I’m unable to verify other than logging into the SUB CM Admin page. > > > > Thanks, > > Jeff > > > > *From:* ccie_voice-boun...@onlinestudylist.com [mailto: > ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Jeff Price > (jeffpric) > *Sent:* Thursday, February 04, 2010 9:55 AM > *To:* Roger Källberg; kavi ten; ccie_voice@onlinestudylist.com > > *Subject:* Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration > rejected > > > > Kavi, > > > > I am having the same issue. I will let you know if I have any success in > finding a solution. I’m asking if you don’t mind doing the same. Thank you > all for your help. > > > > Jeff > > > > *From:* ccie_voice-boun...@onlinestudylist.com [mailto: > ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Roger Källberg > *Sent:* Thursday, February 04, 2010 9:39 AM > *To:* kavi ten; ccie_voice@onlinestudylist.com > *Subject:* Re: [OSL | CCIE_Voice] dhcp server - ip phone resgistration > rejected > > > > Sound like a db replication issue or possibly, but less likely, the order > of CPE in the call manager group. > > > > *Roger Källberg* > Unified Communication Consultant > Cygate AB > > > > *From:* kavi ten [mailto:kaucc...@gmail.com] > *Sent:* den 4 februari 2010 14:23 > *To:* ccie_voice@onlinestudylist.com > *Subject:* [OSL | CCIE_Voice] dhcp server - ip phone resgistration > rejected > > > > Hi Guys, > > > > I have my DHCP server as Publisher > > DHCP Server: PUB > > Primary TFTP : 10.10.210.11 > > Secondary TFTP : 10.10.210.10 > > > > > > Auto reguistration enabed for SUB with range specified. > > > > Now the phone shown in the Devices--> Phones page but Status Rejected > > On the Phone it shows Rejectration Rejected: Security Error > > > > When I auto register with PUB it registers properly. > > > > What could be the problem when Auto regiosteration is enabled in the SUB. > > > > Thanks, > > > > ___ > For more information regarding industry leading CCIE Lab training, please > visit www.ipexpert.com > > -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
