Can you past your config here to see what you did?

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On Sep 28, 2013, at 11:40 AM, Bashar Aziz <bashar1a...@gmail.com> wrote:

> 
> Why I am getting 0% in Voice Gateway and Signalling for the 6th time, 100% 
> tested and worked, what is the trick ?
> 
> 
> Regards,
> 
> 
> 
> On Fri, Sep 27, 2013 at 4:29 PM, sanity insanity 
> <networksanitytoinsan...@gmail.com> wrote:
> Hi Guys,
> 
> Thanks once again for your replies.
> 
> @Lakshmish using your method of creating a seperate partition for RDP  ( on 
> the left side)  and not having the SB PH1 have access to it .  I noticed that 
> when a call is made from PSTN ( with calling number 5252222)  to 3300  and if 
> we enter the pin and dial a number say 2001 ( internal)  . The  2001 phone 
> rings and the call can be answered.
> 
> However the SBPH1 ( physical phone)  is unable show that  the  3001 line is 
> active by showing a red light  and therefore this does not appear to the 
> requirement for MVA is achieved . What do you think?
> 
> -MJ
> 
> 
> 
> On Fri, Sep 27, 2013 at 2:11 AM, Lakshmish NS <lakshmish...@gmail.com> wrote:
> Hi MJ, 
> 
> Martin is right, I had issues with SNR after configuring the RD to 7 digits 
> and setting the service parameter to complete match, MVA and SNR wouldn't go 
> together. Martin however has proposed a new fix, you could try it. The 
> workaround I used for this was to create an "Application Dial Rule", which 
> would certainly solve the issue.
> 
> Cheers, 
> 
> Laksh
> 
> 
> On Tue, Sep 24, 2013 at 8:39 PM, Martin Sloan <martinsloa...@gmail.com> wrote:
> Hi MJ,
> 
> 1) If you set the partial match to 7 digits and then configure your remote 
> destination as a 10 digit number, you'll get a match if the ANI is either 7 
> or 10 digits since the match rule takes 'X' partial-match digits from the RD 
> starting with the last number (2 in this case) and compares it to the ANI of 
> the calling number, but the calling party number must be equal to or shorter 
> in length than the configured remote destination, which is why it's good to 
> just set your RD at 10+ digits if you're using partial match.  Here are some 
> scenarios and the outcome for partial match:
> 
> Partial Match = True 
> Number of Digits For Match = 7 digits
> Remote Destination = 9725252222
> Calling Party Number = 5252222
> Result = Match
> 
> Partial Match = True 
> Number of Digits For Match = 7 digits
> Remote Destination = 9725252222
> Calling Party Number = 9725252222
> Result = Match
> 
> Partial Match = True 
> Number of Digits For Match = 7 digits
> Remote Destination = 5252222
> Calling Party Number = 5252222
> Result = Match
> 
> Partial Match = True 
> Number of Digits For Match = 7 digits
> Remote Destination = 5252222
> Calling Party Number = 9725252222
> Result = No Match (ANI is longer than RD)
> 
> When using Complete match, the ANI and RD have to be exactly the same.  I 
> like to make a call into SB from the PSTN phone prior to configuring SNR and 
> I can quickly see what the ANI is, which is what I then make my RD.
> 
> I had mentioned some buggy behavior with SNR though I never spent time 
> working with partial match since when I heard about that issue I just stuck 
> with complete match but I wanted to test my info above to make sure I wasn't 
> sending incorrect info. It wasn't too hard to run into this buggy behavior.  
> I found a workaround as well so I thought I'd share.
> 
> When changing the Complete Match service parameter to Partial Match you get a 
> screen pop that says to remember and set the "Number of Digits for Caller ID 
> Partial Match" service parameter.  The default for that parameter is 10 and 
> the bug that I found is that on the initial change from default 10 to 7, the 
> new setting does not take effect.  After changing from 10->7 I started to 
> make test calls and my CLID to SB PH1 was showing as the 7 digit ANI of the 
> PSTN phone and not "SB PHONE 2 3002" like it should.  I dug around for a bit 
> and tweaked a couple parameters and re-tested.  The deal is that you have 
> change Complete Match to Partial Match -> Save then change Partial Match 
> digits from 10 to 7 and Save again.
> 
> 2) For this one if your service parameter is set to Complete Match and your 
> ANI is 7 digits, just set your RD to the 7 digit number then use route 
> patterns/xlations to manipulate as needed.
> 
> 3) Not sure about that one.  I've definitely seen conflicting information on 
> certain things but I've realized that some of the training material is years 
> in the making and when things are discovered or updated, maybe the old 
> information is not or it's just floating out there.  I can confirm that based 
> on some recent experience with trusted trainers it was reiterated not to use 
> partial match, maybe in part because of the issue that I hit today.
> 
> Marty
> 
> 
> On Tue, Sep 24, 2013 at 8:19 AM, sanity insanity 
> <networksanitytoinsan...@gmail.com> wrote:
> Hi Guys ,
> 
> Thanks a lot for taking time out to reply to my  question. It was really 
> helpful.
> 
>  I was trying to understand the difference between full match  with  10 
> digits   and partial match with 7 digits.   Here are my scenarios...
> 
> 1) If I use partial match with 7 digits   then this will satisfy the 
> condition where my calling number is 7 digits  ( in this instance it is 
> 5252222)   but what happens if my calling
> number is in the form  9725252222 in this case it is 10 digits whereas my 
> service parameter indicates just 7 digits ?
> 
> 
> 2) If I use complete match with 10 digits then  will satisfy the condition 
> where my calling number is 10 digits but not when 7 digits .  I am not sure 
> where complete
> match means it includes the condition of the calling number with 7 digits as 
> well.  Would you be able to throw some light on this?
> 
> 
> 3)In some of the IPexpert walk through videos I see the instructor seems to 
> prefer partial match with 7 digits . However this may be for a specific 
> condition.  I am I correct on this ?
> 
> MJ
> 
> 
> 
> 
> On Wed, Sep 18, 2013 at 8:55 PM, Martin Sloan <martinsloa...@gmail.com> wrote:
> Hi MJ,
> 
> I did some research on this since I've been configuring MVA for a while but 
> have had some questions about underlying architecture.  Here's some responses 
> to your info plus some of my findings.
> 
> 1)  If the MVA DID is in line with your standard DID range for the site, why 
> not just piggy back on the existing CUCM dial-peers instead of creating a new 
> one just for MVA.  Say Site B for example with a 3XXX extension range, you 
> could use the CUCM dial-peer:
> 
> dial-peer voice 3000 voip
>  destination pattern 3...$
>  session target ipv4:10.10.210.11
>  no vad
>  voice-class codec 1
>  voice-class h323 1
>  dtmf-relay h245-alpha
>  incoming called-number .
> 
> 2)  Looks good.  I change my service name to MVA since I think there's a typo 
> somewhere in the CUCM pages where I copy/paste from but as long as the names 
> match up between the service and dial-peer, no worries.
> 
> 3) Right, I use the same to chop DID's to local extensions:
> 
>  voice translation-rule 1
>    rule 1 /.+\(....\)/ /\1/
> 
>  voice translation-profile PSTN
>    translate called 1
> 
>  voice-port 0/0/0:23
>   translation-prof in PSTN
> 
> 4) Here, I do not use partial match.  I've heard from a truly reliable source 
> that there is some buggi-ness with this particular version of CUCM and 
> partial matches.  In the end, I think it's less thinking and moving parts if 
> you just use a full match anyway.  Just my POV on this one.  Also, the 
> 'Mobile Voice Access Number' in the CCM service parameters isn't used for 
> VXML MVA.  From what I understand, this parameter is for Mobile Communicator. 
>  I've been through the SRND and several other pages and cannot pin the exact 
> meaning of the parameter, but in the SRND configuration guide for VXML MVA, 
> it cruises right over this parameter so I believe it's safe to leave at 
> default (blank).
> 
> 5) I've never had a specific requirement for this.  I'd say don't waste the 
> time setting it up if it's not required but if anyone has good reason to 
> think it should be configured, lemme know.
> 
> 6) Agreed
> 
> 7) Be sure to set the re-routing CSS on the RDP (if SNR is required).
>      CSS = MVA dialing
>      Rerouting CSS = SNR dialing
> 
>    Also, just as a heads up you shouldn't use SLRL for SNR as it will use the 
> RG of the calling party (say HQ phone 2) so the call would try to go out HQ 
> GW.  Make sure to create a route list for SB (if SNR is at site B) and point 
> the SNR pattern to it so it goes out the SB gateway as a local call.
> 
> 8) I use the full number here.
> 
> 9) I never set this and have not had any issues with MVA/SNR.  The CUCM help 
> file says its for CDR usage.  Anyone know how/if this setting impacts MVA/SNR?
> 
> 10) Agreed
> 
> About your questions, I'm not clear on #1.  Like I mentioned, I use full 
> match and don't do any manipulation of the calling number for SNR/MVA 
> questions.  For #2, you haven't mentioned the Media Resources->Mobile Voice 
> Access->Mobile Voice Access Directory Number.  Unlike the Service Parameter 
> Setting, this is the number that's used for calls from the H323 GW to CUCM.  
> Here are some debugs from a call into MVA from my lab.  The process is that 
> the CUCM instructs the GW to play prompts and collect digits based on the 
> DTMF input from the caller.  The call was placed from my configured Remote 
> Destination so I'm not prompted to enter my RD Number:
> 
> -------GET PIN------------
> Here the gateway prompts to enter my pin to authenticate
> 
> <vxml version="2.0">
> 
> 
> 
> 
> 
> <form id="Pin">
> 
>     <grammar type="application/grammar+regex">.</grammar>
>     <field name="pin" type="digits?minlength=1;maxlength=20">
> 
>       <prompt>
>          <audio s
> 
> 
> -------GET FUNCTION------------
> 
> Here the GW asks what I'd like to do ("Press 1 to place a call")
> 
> <vxml version="2.0">
> 
> 
> 
> 
>  <form id="GetFunctionSel">
>     <grammar type="application/grammar+regex">.</grammar>
>     <field name="funcsel" type="digits?length=1">
> 
> 
>              <pro
> 
> 
> -------GET DIALED DIGITS------------
> 
> Here the GW plays the prompt to enter the digits followed by pound
> 
> <vxml version="2.0">
> 
> 
> 
> 
> <form id="Getdialno">
> 
>     <grammar type="application/grammar+regex">.</grammar>
>     <field name="dialno" type="digits?minlength=1;maxlength=50">
> 
>       <prompt>
> 
> 
> -------TRANSFER CALL------------
> 
> Here is where the gateway hands the call to CUCM.
> 
> <vxml version="2.0">
> 
> 
> 
> 
>  <form id="Transfer">
> 
>       <transfer name="mycall"
>                 dest="phone://3300"  <<<<<<<<<<< This is the number that is 
> set within the CUCM Media Resource->MVA section
>                 cisco-ani="phone://4088397263"  <<<<<<<<<<<< ANI AKA Remote 
> Destination number
>                 cisco-rdn="phone://2002" <<<<<<<<<< My dialed digits
>                 cisco-rdntype="0" 
>                 cisco-rdnp
> 
> So the lesson for me was the the Media Resource->MVA Dir Num does not have to 
> match up with the actual MVA DID whatsoever.  The requirement is that the 
> H323 GW has a dial-peer with a destination-pattern that matches the number 
> provided by CUCM to the GW in the dest="phone://3300" field and also that the 
> H323 GW CSS within CUCM has access to the PT that the MVA Dirn was assigned 
> to.  In the case of using the existing CUCM outbound dial-peer from above, 
> it's a match.  Here's an example of a failed transfer:
> 
> 
> -------FAILED TRANSFER------------
> 
> <vxml version="2.0">
> 
> 
> 
> 
>  <form id="Transfer">
> 
>       <transfer name="mycall"
>                 dest="phone://33"
>                 cisco-ani="phone://4088397263"
>                 cisco-rdn="phone://2002"
>                 cisco-rdntype="0"
>                 cisco-rdnpla
> 
> 
> In this case the 'dest' field was changed to just '33' within CUCM Media 
> Resource->MVA and the call failed.  But, after I copied my CUCM dial-peer on 
> the GW and changed the destination-pattern to 33$, it worked.
> 
> I hope this helps, and thanks for prompting me to stop procrastinating on 
> researching MVA!
> 
> Marty
> 
> 
> On Wed, Sep 18, 2013 at 12:25 AM, sanity insanity 
> <networksanitytoinsan...@gmail.com> wrote:
> 
> Hi Guys,
> 
> 
> 
> I have been trying to find the right way of configuring MVA. Below is my 
> configuration....
> 
> 
> Details:
> =============
> 
> My config is following....
> 
> 1) The dial-peers are set in the following way
> 
> dial-peer voice 102 voip
>  preference 2
>  destination-pattern 3300
>  session target ipv4:<ip address of the CUCM Pub>  dtmf-relay 
> h245-alphanumeric 
>  codec g711ulaw  no vad !
> dial-peer voice 3300 pots
>  service cmm
>  incoming called-number 3300
>  no digit-strip
> 
> 
> 2) here is the MVA service url
> !
> application
> service cmm http://<ip address of the CUCM
> Pub>:8080/ccmivr/pages/IVRMainpage.vxml
> !
> 
> 
> 3) I am stripping 3033300 coming from pstn to last  4 digits  using a 
> translation-rule on the voice-port level . That is 3033300 becomes 3300 when 
> it 
> reaches CUCM.
> 
> 
> 4) On CUCM in the service parameters...
> 
> Enable Mobile Voice access is set to True Mobile voice access number is  3300 
> Matching caller id with Remote Destination is Partial Match Number of digits 
> of 
> Caller ID Partial Match is 7
> 
> 5) The Mobility softkey has been added for "on hold" and "connected" at the 
> softkey template level and applied to the phone ( SB PH1)
> 
> 
> 6)At the User  SB phone 1  I have enabled "Enable Mobility" and "Enable 
> Mobile 
> Voice Access"
> also selected the MAC address of the phone
> 
> 
> 7) Created a Remote Dest profile and selected user id of sb ph1 and the 
> correct 
> calling search space for the phone
> 
> 
> 8) Added a Remoted Destination number of 5252222
> 
> 
> 9) Also went to device > phone  and selected the Owner User ID of SB Ph1
> 
> 
> 10) Cisco Unified Mobile Voice Access Service is running on both Sub and Pub 
> on 
> CUCM
> 
> 
> 
> Questions :
> ====================  
> 
> 
> 1) Do I need to change my incoming calling number (coming from pstn)  from 
> 5252222 to 95252222  because the busy trigger on 3001 (phone)
> is set to 1  and therefore any other calling coming to this number will head 
> to Voicemail?
> 
> 
> 2) Anything else you find incorrect with my configuration?
> 
> 
> -MJ
> 
> _______________________________________________
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com
> 
> 
> 
> 
> _______________________________________________
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com
> 
> 
> 
> _______________________________________________
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com
> 
> _______________________________________________
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com
_______________________________________________
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