Re: [OSL | CCIE_Voice] Testing SRST

[VanBenschoten, Brian]

An easier way to test for THEO and such is to just shut down the voice-port 
(not the controller or serial).
Quick and easy and perhaps not as easy to overlook when troubleshooting.
I've left my null routes in a couple of times without realizing it.

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill Hatcher
Sent: Thursday, October 17, 2013 11:53 AM
To: ccievoice
Subject: Re: [OSL | CCIE_Voice] Testing SRST

If you are talking about testing redundancy I'll do the same thing on the 
gateway I want to simulate as being down. For example when doing TEHO where if 
the remote gateway is down we want to fail to the local gateway, I'll go to the 
remote gateway and put in the static routes.

On Thu, Oct 17, 2013 at 11:46 AM, Alex Mendoza 
mailto:aa.mend...@icloud.com>> wrote:
Hi Bill

I use your way to test SRST, I'm wondering what are you using for test Route 
List when they have 2 route groups.

best regards.
Alex

On Oct 17, 2013, at 09:23 AM, Bill Hatcher 
mailto:wchatc...@gmail.com>> wrote:
I have been looking for quick and easy ways to test SRST, and I've found many 
different waqys of doing this.  With the exception of pulling the WAN 
interface, they all seem to take a lot of time and effort to accomplish.  
Anything from creating access-lists to block the traffic to creating new call 
manager groups and shutting down one of the CallManager services.
I have found that a couple of simple static routes to the null 0 interface 
works very well.
ip route 10.10.210.10 255.255.255.255 null 0
ip route 10.10.210.11 255.255.255.255 null 0

no ip route 10.10.210.10 255.255.255.255 null 0
no ip route 10.10.210.11 255.255.255.255 null 0
Add them to a notepad and the no statements as well and you can quickly send 
your devices into srst mode. Now if you have any VoIP dial-peers that point to 
other addresses across your WAN you may have to add those as well.
What do you guys use?
HTH
Bill.
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Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping

[VanBenschoten, Brian]

I've found the QoS questions are very specific to test a certain area of 
knowledge.  They are not looking for what we would consider a "best practice" 
system wide.  I think we could skip setting the DSCP values in CUCM.
If you think the question calls for it you can have your class-map match both 
AF31 and CS3 for signaling.

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill Hatcher
Sent: Thursday, October 17, 2013 3:14 PM
To: probert...@gmail.com
Cc: ccievoice
Subject: Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping

I agree to setting the service parameters to default first.  I was planning on 
doing that myself.  As to changing the DSCP values, it all depends on what they 
ask for in the QoS section of the test is all.

On Thu, Oct 17, 2013 at 2:24 PM, 
probert...@gmail.com 
mailto:probert...@gmail.com>> wrote:
Hi,
I think my strategy will be to set all Service Parameters to default before 
making changes. This way I can avoid and undesirable presets.
Let me know your thoughts on this.
Also why are you setting DSCP for Phone Configuration and DSCP for Cisco 
CallManager to Device Interface to AF31? Default CS3 should be good, let me 
know if I'm wrong on this?

Rob

On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher 
mailto:wchatc...@gmail.com>> wrote:
My test is just a couple of weeks away, and I've been reading different blogs 
on how to maximize your time.  The one thing I'm really struggling with is 
mapping out my dial-plan during my read through of the lab.  I would love to 
hear what others are doing.

I have also been building base router configs for h323, gatekeeper, mgcp, 
srst,sip, etc so that I can practice quickly configuring those on the routers.

One of the things I haven't really been keeping track of are some of the 
service parameters that I should adjust out of habit. Here are a few that I can 
think of off the top of my head that I plan on tweaking at the start of the 
exam.  Please feel free to add to them.
Enterprise Parameters
DSCP for Phone Configuration - Set to AF31
DSCP for Cisco CallManager to Device Interface - Set to AF31
Change the Phone URL's to IP's
Organization Top Level Domain
Cluster Fully Qualified Domain Name

Service Parameters - CallManager
T302 Time - Know where it is if you need ot change interdigit timeout
Call Classification - Offnet
Builtin Bridge Enabled - True
Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed.
Transfer On-hook Enabled - True (Also a great thing to do in production when 
migrating users from other phone systems)
Block offnet to offnet transfers - Know where it's at.
Auto Call Pickup Enabled - True
Call Back Enabled Flag - True (Verify)
Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed.
Stop routing on Unallocated Number Flag - False - H323 redundancy
Preferred G.711 Millisecond Packet Size - 20 (Verify)
Preferred G.729 Millisecond Packet Size - 20 (Verify)
G722 Codec Enabled - Disabled (Unless otherwise directed)
Intraregion Audio Codec Default - G711/G722 (Verify)
Interregion Audio Codec Default - G729 (Verify)
Automated Alternate Routing Enabled - True (This one gets me every time on AAR 
so I turn it on by default now)
Enable Mobile Voice Access - Set as required
Mobile Voice Access Number - Set as required
System Remote Access Blocked Numbers - Set as required
Service Parameters -Cisco IP Voice Media Streaming App
Supported MOH Codecs - G711 mulaw and G729 Annex A

HTH
Bill Hatcher

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Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension

[Somphol Boonjing]

Sorry for revisiting this old thread.   The Calling Party Transformation at
the Device Pool level would come in handy for this particular need.

In the document starting 7.1.2, this is stated explicitly,
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fscallpn.html#wp1325305
.

*Cisco Unity/Cisco Unity Connection*


Because no calling party transformation options exist in Cisco Unified
Communications Manager Administration for voice-messaging ports, make sure
that you configure the calling party number transformations in the device
pool that is associated with the voice-messaging ports.
...

Table 7-8 Configuring the Calling Party Transformation CSS to Localize the
Calling Party Number
Also mentioned "Use Device Pool Calling Party Transformation CSS" as a
method to Localize the Calling Party Number.
...
...

The same document for 7.0.1 contained the table 7-8, but somehow doesn't
have that explicit section on Cisco Unity/Cisco Unity Connection's calling
party localization.  (REF:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html#wp1276877)
   So, I am not so sure whether this is possible in CUCM 7.0.1/CUC 7.0.1.

I don't have lab access to test this now, but would appreciate if anyone
can help testing this.

Note: I recall seeing some sort of Technotes outlining the strategy to
perform Calling Party transformation for Call Manager 4.x or something that
doesn't rely on Gateway's Calling Party Transformation.I can't locate
it now, but if anyone could point me to the URL that would be great.

Regards,
--Somphol.





On Sat, Mar 23, 2013 at 12:34 AM, Leslie Meade wrote:

>  Easy way of doing this is to copy the hunt pilot and give it another
> number.. set user caller ID and mask it to 
>
> Then in the call-manager-fallback change the voicemail to the new hunt
> pilot and your done
>
> ** **
>
> ** **
>
> *Leslie Meade* 
>
> .. *
> Mobile:778.228.4339* | *Main:* *604.676.5239*
> *Email:* leslie.me...@lvs1.com
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chadi H Hassoune
> (chassoun)
> *Sent:* Thursday, March 21, 2013 7:10 PM
> *To:* Pixar Perfect; Mark Thrash (marthras); Steve Keller
> *Cc:* CCIE Voice OSL
>
> *Subject:* Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate
> Extension
>
>  ** **
>
> Calling Party Xform and assign it to the CUC Device Pool works fine for
> me. 
>
> ** **
>
> HTH
>
> ** **
>
> *From: *Pixar Perfect 
> *Date: *Wednesday, March 20, 2013 11:43 PM
> *To: *"Mark Thrash (marthras)" , Steve Keller <
> skeller...@gmail.com>
> *Cc: *CCIE Voice OSL 
> *Subject: *Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate
> Extension
>
> ** **
>
>   the requirement is always for SiteB calling into SiteA voicemail by
> hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform
> isnt any use on MGCP gateway for incoming calls.  
>
> ** **
>
> here is another way of doing it ... 
>
> ** **
>
> Voicemail Pilot for CUC is 2200
>
> ** **
>
> call-manager-fallback
>
> voicemail 2777   ---> siteB specific 
>
> ** **
>
> translation-pattern on CUCM to convert 2777 into 2200 and mask calling
> number . The CSS of the translation pattern should have access to 2200.
> 
>
> ** **
>
> ** **
>
> ** **
>
> there is no definitive answer as to which solution is graded positively.
> there is a reason why many leading CCIE instructors say this is not a test
> of best practices but a test of how like able is your solution to the
> script. .. :) 
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>  --
>
> From: marth...@cisco.com
> To: skeller...@gmail.com
> Date: Thu, 21 Mar 2013 03:59:48 +
> CC: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate
> Extension
>
> What about a calling party transform mask on the incoming gateway?
>
> Sent from my iPhone
>
>
> On Mar 20, 2013, at 10:43 PM, "Steve Keller"  wrote:
> 
>
>  Thanks Bill, I like this option pretty well as it seems to limit
> treatment of calls this way to CUC when site B is in SRST mode only.  I
> will try to lab this up tomorrow morning. Question for you, will this only
> solve my issue of pressing the VM button to access my mailbox to retrieve a
> message. Meaning when PSTN calls in to site B phone and then gets
> forward(redirected) to voicemail, I use a dial-peer that provides RDNIS
> capabilites to route the caller to the correct mailbox and not the opening
> greeting. So with this would i still want to use the following to get the
> caller into my mailbox?
>
>  
>
> dial-peer voice 2600 pots
>
> destination-pattern 2600
>
> port 0/0/0:23
>
> no digit-strip
>
> prefix 202555 ( assuming no LD code at this site )
>
>  
>
> this is

Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping

[Somphol Boonjing]

Hi,

On service parameters, you may also want to check Vik's article
http://blog.ipexpert.com/2010/10/13/common-ucm-service-parameters-to-change/
.

On the comment section, Trinifox also mentioned "Please add: Intraregion
Audio Codec Default to G729 to avoid CSCsl74701 Bug".

In my checklist, I also tweak "Conference" section esp. "Drop Ad Hoc
Conference".

Regards,
--Somphol.




On Fri, Oct 18, 2013 at 7:14 AM, Bill Hatcher  wrote:

> I agree to setting the service parameters to default first.  I was
> planning on doing that myself.  As to changing the DSCP values, it all
> depends on what they ask for in the QoS section of the test is all.
>
>
> On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com <
> probert...@gmail.com> wrote:
>
>> Hi,
>>
>> I think my strategy will be to set all Service Parameters to default
>> before making changes. This way I can avoid and undesirable presets.
>> Let me know your thoughts on this.
>>
>> Also why are you setting DSCP for Phone Configuration and DSCP for Cisco
>> CallManager to Device Interface to AF31? Default CS3 should be good, let me
>> know if I'm wrong on this?
>>
>> Rob
>>
>>
>> On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wrote:
>>
>>> My test is just a couple of weeks away, and I've been reading different
>>> blogs on how to maximize your time.  The one thing I'm really struggling
>>> with is mapping out my dial-plan during my read through of the lab.  I
>>> would love to hear what others are doing.
>>>
>>> I have also been building base router configs for h323, gatekeeper,
>>> mgcp, srst,sip, etc so that I can practice quickly configuring those on the
>>> routers.
>>>
>>> One of the things I haven't really been keeping track of are some of the
>>> service parameters that I should adjust out of habit. Here are a few that I
>>> can think of off the top of my head that I plan on tweaking at the start of
>>> the exam.  Please feel free to add to them.
>>>
>>> Enterprise Parameters
>>> DSCP for Phone Configuration - Set to AF31
>>> DSCP for Cisco CallManager to Device Interface - Set to AF31
>>> Change the Phone URL's to IP's
>>> Organization Top Level Domain
>>> Cluster Fully Qualified Domain Name
>>>
>>> Service Parameters - CallManager
>>> T302 Time - Know where it is if you need ot change interdigit timeout
>>> Call Classification - Offnet
>>> Builtin Bridge Enabled - True
>>> Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as
>>> needed.
>>> Transfer On-hook Enabled - True (Also a great thing to do in production
>>> when migrating users from other phone systems)
>>> Block offnet to offnet transfers - Know where it's at.
>>> Auto Call Pickup Enabled - True
>>> Call Back Enabled Flag - True (Verify)
>>> Single Button Barge/CBarge Policy - Set to Barge unless otherwise
>>> directed.
>>> Stop routing on Unallocated Number Flag - False - H323 redundancy
>>> Preferred G.711 Millisecond Packet Size - 20 (Verify)
>>> Preferred G.729 Millisecond Packet Size - 20 (Verify)
>>> G722 Codec Enabled - Disabled (Unless otherwise directed)
>>> Intraregion Audio Codec Default - G711/G722 (Verify)
>>> Interregion Audio Codec Default - G729 (Verify)
>>> Automated Alternate Routing Enabled - True (This one gets me every time
>>> on AAR so I turn it on by default now)
>>> Enable Mobile Voice Access - Set as required
>>> Mobile Voice Access Number - Set as required
>>> System Remote Access Blocked Numbers - Set as required
>>>
>>> Service Parameters -Cisco IP Voice Media Streaming App
>>> Supported MOH Codecs - G711 mulaw and G729 Annex A
>>>
>>> HTH
>>>
>>> Bill Hatcher
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com
>>>
>>
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping

[Bill Hatcher]

I agree to setting the service parameters to default first.  I was planning
on doing that myself.  As to changing the DSCP values, it all depends on
what they ask for in the QoS section of the test is all.


On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com
wrote:

> Hi,
>
> I think my strategy will be to set all Service Parameters to default
> before making changes. This way I can avoid and undesirable presets.
> Let me know your thoughts on this.
>
> Also why are you setting DSCP for Phone Configuration and DSCP for Cisco
> CallManager to Device Interface to AF31? Default CS3 should be good, let me
> know if I'm wrong on this?
>
> Rob
>
>
> On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher  wrote:
>
>> My test is just a couple of weeks away, and I've been reading different
>> blogs on how to maximize your time.  The one thing I'm really struggling
>> with is mapping out my dial-plan during my read through of the lab.  I
>> would love to hear what others are doing.
>>
>> I have also been building base router configs for h323, gatekeeper, mgcp,
>> srst,sip, etc so that I can practice quickly configuring those on the
>> routers.
>>
>> One of the things I haven't really been keeping track of are some of the
>> service parameters that I should adjust out of habit. Here are a few that I
>> can think of off the top of my head that I plan on tweaking at the start of
>> the exam.  Please feel free to add to them.
>>
>> Enterprise Parameters
>> DSCP for Phone Configuration - Set to AF31
>> DSCP for Cisco CallManager to Device Interface - Set to AF31
>> Change the Phone URL's to IP's
>> Organization Top Level Domain
>> Cluster Fully Qualified Domain Name
>>
>> Service Parameters - CallManager
>> T302 Time - Know where it is if you need ot change interdigit timeout
>> Call Classification - Offnet
>> Builtin Bridge Enabled - True
>> Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as
>> needed.
>> Transfer On-hook Enabled - True (Also a great thing to do in production
>> when migrating users from other phone systems)
>> Block offnet to offnet transfers - Know where it's at.
>> Auto Call Pickup Enabled - True
>> Call Back Enabled Flag - True (Verify)
>> Single Button Barge/CBarge Policy - Set to Barge unless otherwise
>> directed.
>> Stop routing on Unallocated Number Flag - False - H323 redundancy
>> Preferred G.711 Millisecond Packet Size - 20 (Verify)
>> Preferred G.729 Millisecond Packet Size - 20 (Verify)
>> G722 Codec Enabled - Disabled (Unless otherwise directed)
>> Intraregion Audio Codec Default - G711/G722 (Verify)
>> Interregion Audio Codec Default - G729 (Verify)
>> Automated Alternate Routing Enabled - True (This one gets me every time
>> on AAR so I turn it on by default now)
>> Enable Mobile Voice Access - Set as required
>> Mobile Voice Access Number - Set as required
>> System Remote Access Blocked Numbers - Set as required
>>
>> Service Parameters -Cisco IP Voice Media Streaming App
>> Supported MOH Codecs - G711 mulaw and G729 Annex A
>>
>> HTH
>>
>> Bill Hatcher
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping

[probert...@gmail.com]

Hi,

I think my strategy will be to set all Service Parameters to default before
making changes. This way I can avoid and undesirable presets.
Let me know your thoughts on this.

Also why are you setting DSCP for Phone Configuration and DSCP for Cisco
CallManager to Device Interface to AF31? Default CS3 should be good, let me
know if I'm wrong on this?

Rob


On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher  wrote:

> My test is just a couple of weeks away, and I've been reading different
> blogs on how to maximize your time.  The one thing I'm really struggling
> with is mapping out my dial-plan during my read through of the lab.  I
> would love to hear what others are doing.
>
> I have also been building base router configs for h323, gatekeeper, mgcp,
> srst,sip, etc so that I can practice quickly configuring those on the
> routers.
>
> One of the things I haven't really been keeping track of are some of the
> service parameters that I should adjust out of habit. Here are a few that I
> can think of off the top of my head that I plan on tweaking at the start of
> the exam.  Please feel free to add to them.
>
> Enterprise Parameters
> DSCP for Phone Configuration - Set to AF31
> DSCP for Cisco CallManager to Device Interface - Set to AF31
> Change the Phone URL's to IP's
> Organization Top Level Domain
> Cluster Fully Qualified Domain Name
>
> Service Parameters - CallManager
> T302 Time - Know where it is if you need ot change interdigit timeout
> Call Classification - Offnet
> Builtin Bridge Enabled - True
> Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed.
> Transfer On-hook Enabled - True (Also a great thing to do in production
> when migrating users from other phone systems)
> Block offnet to offnet transfers - Know where it's at.
> Auto Call Pickup Enabled - True
> Call Back Enabled Flag - True (Verify)
> Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed.
> Stop routing on Unallocated Number Flag - False - H323 redundancy
> Preferred G.711 Millisecond Packet Size - 20 (Verify)
> Preferred G.729 Millisecond Packet Size - 20 (Verify)
> G722 Codec Enabled - Disabled (Unless otherwise directed)
> Intraregion Audio Codec Default - G711/G722 (Verify)
> Interregion Audio Codec Default - G729 (Verify)
> Automated Alternate Routing Enabled - True (This one gets me every time on
> AAR so I turn it on by default now)
> Enable Mobile Voice Access - Set as required
> Mobile Voice Access Number - Set as required
> System Remote Access Blocked Numbers - Set as required
>
> Service Parameters -Cisco IP Voice Media Streaming App
> Supported MOH Codecs - G711 mulaw and G729 Annex A
>
> HTH
>
> Bill Hatcher
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

[OSL | CCIE_Voice] Base configs and Dial Plan mapping

[Bill Hatcher]

My test is just a couple of weeks away, and I've been reading different
blogs on how to maximize your time.  The one thing I'm really struggling
with is mapping out my dial-plan during my read through of the lab.  I
would love to hear what others are doing.

I have also been building base router configs for h323, gatekeeper, mgcp,
srst,sip, etc so that I can practice quickly configuring those on the
routers.

One of the things I haven't really been keeping track of are some of the
service parameters that I should adjust out of habit. Here are a few that I
can think of off the top of my head that I plan on tweaking at the start of
the exam.  Please feel free to add to them.

Enterprise Parameters
DSCP for Phone Configuration - Set to AF31
DSCP for Cisco CallManager to Device Interface - Set to AF31
Change the Phone URL's to IP's
Organization Top Level Domain
Cluster Fully Qualified Domain Name

Service Parameters - CallManager
T302 Time - Know where it is if you need ot change interdigit timeout
Call Classification - Offnet
Builtin Bridge Enabled - True
Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed.
Transfer On-hook Enabled - True (Also a great thing to do in production
when migrating users from other phone systems)
Block offnet to offnet transfers - Know where it's at.
Auto Call Pickup Enabled - True
Call Back Enabled Flag - True (Verify)
Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed.
Stop routing on Unallocated Number Flag - False - H323 redundancy
Preferred G.711 Millisecond Packet Size - 20 (Verify)
Preferred G.729 Millisecond Packet Size - 20 (Verify)
G722 Codec Enabled - Disabled (Unless otherwise directed)
Intraregion Audio Codec Default - G711/G722 (Verify)
Interregion Audio Codec Default - G729 (Verify)
Automated Alternate Routing Enabled - True (This one gets me every time on
AAR so I turn it on by default now)
Enable Mobile Voice Access - Set as required
Mobile Voice Access Number - Set as required
System Remote Access Blocked Numbers - Set as required

Service Parameters -Cisco IP Voice Media Streaming App
Supported MOH Codecs - G711 mulaw and G729 Annex A

HTH

Bill Hatcher
___
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Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.

[Bill Hatcher]

Thank you everyone for your help on this.  I was bald already, but if I had
hair I would have been pulling it out on this one.


On Thu, Oct 17, 2013 at 11:23 AM, Ramcharan Arya
wrote:

> This is one of the best blog written by Vik on SRST it covers all details.
>
>
>
> http://blog.ipexpert.com/2012/03/12/high-availability-series-3-unified-cme-for-srst-gotchas/
>
> Thanks,
>
> Ramcharan Arya CCIE # 28926 (Voice/Routing & Switching)
>
>
>
> On Thu, Oct 17, 2013 at 11:05 AM, Martin Sloan wrote:
>
>> Bill,
>>
>> Here's a great reference for CME MWI:
>>
>> http://ciscovoiceguru.com/518/cue-mwi-notification-methods/
>>
>> I've used this a lot through my studies.
>>
>> @Ramcharan - good call.  I think the symptom with that bug is that the
>> phones will register but display no DN's and if you issue 'show ephone reg'
>> it will show the DN's as 'invalid' or something like that.  I have also hit
>> the other bug with SC MGCP after coming out of SRST the dial-peers are
>> still chosen on inbound calls from the PSTN.  I believe the fix for that is
>> to globally set 'voice hunt 2' and under the ephone-dn's assign a higher
>> preference (I use 9).
>>
>>
>> On Thu, Oct 17, 2013 at 11:53 AM, Bill Hatcher wrote:
>>
>>> Ok, I figured out my issue with the MWI not coming on in SRST more.
>>> Need to ass the key word unsolicited to the mwi-server command.
>>>
>>> Now to get the CME-SRST working.
>>>
>>>
>>>
>>> On Thu, Oct 17, 2013 at 10:36 AM, Bill Hatcher wrote:
>>>
 Seifeddine,

 I've run that debug, but there is absolutly no output when I'm using
 CME-SRST.

 Bill


 On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili <
 seifeddine.tl...@lvs1.com> wrote:

>  Can you send the output of debug ephone register?
>
> ** **
>
> Thx
>
> ** **
>
> *Kindly***
>
> * *
>
> *Seifeddine Tlili*  
>
> [image: Description: Description: Long View Systems]
>
> M.Eng CCIE # 26440
> Systems Consultant 
>
> .. *
> Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900
>
> [image: Description: Description: 
> Linkedin]
>  [image: Description: Description: 
> Twitter]
>  [image: Description: Description: 
> Facebook] [image:
> Description: Description: 
> Facebook]
> www.longviewsystems.com
> This message and any attached documents are only for the use of
> the intended recipient(s), are confidential and may contain privileged
> information. Any unauthorized review, use, retransmission, or other
> disclosure is strictly prohibited. If you have received this message
> in
> error, notify the sender immediately, and delete the original message.
> 
>
> ** **
>
> ** **
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher
> *Sent:* Thursday, October 17, 2013 9:29 AM
> *To:* ccievoice
> *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help.
>
> ** **
>
> Marty,
>
> The weird thing is they work when I use call-manager-fallback.
> Looking at the cnf file, all seems well.
>
> ** **
>
> On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan <
> martinsloa...@gmail.com> wrote:
>
> I had a similar issue recently which ended up being a DB replication
> problem.  You could check the phones config file:
>
> 10.10.210.11:6970/SEP123456789123.cnf.xml> Check
> subscribers copy of the file
>
> Right-click, select view source and search for 'srst' and see what it
> has there.  I could be missing something but it looks like you have enough
> to get them registered.
>
> Marty
>
> ** **
>
> On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher 
> wrote:
>
>   It’s not working!!  Can anyone see something I may be doing wrong?
> My PRI and CUE register, I can even see SIP MWI being sent, but my phones
> will not register.  They worked when I was using call-manager-fallback
> though so I know my SRST configuration is correct on the CallManager.*
> ***
>
>  
>
> telephony-service
>
>  srst mode auto-provision all
>
>  srst ephone description auto provisioned ephone  : Oct 17 2013
> 14:47:07
>
>  srst dn line-mode octo
>
>  max-ephones 4
>
>  max-dn 4
>
>  ip source-address 10.10.202.1 port 2000
>
>  max-conferences 12 gain -6
>
>  transfer-system full-consult
>
>  create cnf-files version-stamp 7960

Re: [OSL | CCIE_Voice] CME as SRST help.

[Alex Mendoza]

HiFor telephony-service I used:srst mode auto-provision allsrst dn line-mode dualAnd for call-manager-fallback I used:max-ephone 4max-dn 8 dual-lineAm I wrong using these configurations?I avoid octo-lines for the bug mentioned.best regards!AlexOn Oct 17, 2013, at 10:18 AM, Ramcharan Arya  wrote:Hi Bill,I believe this might be related to a bug with using octo lines in CME SRST.Come out of SRST and reload the router this might resolve the issue.Regards,Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher  wrote:It’s not working!!  Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register.  They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com  Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com  Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Testing SRST

[Bill Hatcher]

If you are talking about testing redundancy I'll do the same thing on the
gateway I want to simulate as being down. For example when doing TEHO where
if the remote gateway is down we want to fail to the local gateway, I'll go
to the remote gateway and put in the static routes.


On Thu, Oct 17, 2013 at 11:46 AM, Alex Mendoza wrote:

> Hi Bill
>
> I use your way to test SRST, I'm wondering what are you using for test
> Route List when they have 2 route groups.
>
> best regards.
> Alex
>
> On Oct 17, 2013, at 09:23 AM, Bill Hatcher  wrote:
>
> I have been looking for quick and easy ways to test SRST, and I've found
> many different waqys of doing this.  With the exception of pulling the WAN
> interface, they all seem to take a lot of time and effort to accomplish.
> Anything from creating access-lists to block the traffic to creating new
> call manager groups and shutting down one of the CallManager services.
>
> I have found that a couple of simple static routes to the null 0 interface
> works very well.
> ip route 10.10.210.10 255.255.255.255 null 0
> ip route 10.10.210.11 255.255.255.255 null 0
>
> no ip route 10.10.210.10 255.255.255.255 null 0
> no ip route 10.10.210.11 255.255.255.255 null 0
>
> Add them to a notepad and the no statements as well and you can quickly
> send your devices into srst mode. Now if you have any VoIP dial-peers that
> point to other addresses across your WAN you may have to add those as well.
>
> What do you guys use?
>
> HTH
>
> Bill.
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.

[Ramcharan Arya]

This is one of the best blog written by Vik on SRST it covers all details.


http://blog.ipexpert.com/2012/03/12/high-availability-series-3-unified-cme-for-srst-gotchas/

Thanks,
Ramcharan Arya CCIE # 28926 (Voice/Routing & Switching)



On Thu, Oct 17, 2013 at 11:05 AM, Martin Sloan wrote:

> Bill,
>
> Here's a great reference for CME MWI:
>
> http://ciscovoiceguru.com/518/cue-mwi-notification-methods/
>
> I've used this a lot through my studies.
>
> @Ramcharan - good call.  I think the symptom with that bug is that the
> phones will register but display no DN's and if you issue 'show ephone reg'
> it will show the DN's as 'invalid' or something like that.  I have also hit
> the other bug with SC MGCP after coming out of SRST the dial-peers are
> still chosen on inbound calls from the PSTN.  I believe the fix for that is
> to globally set 'voice hunt 2' and under the ephone-dn's assign a higher
> preference (I use 9).
>
>
> On Thu, Oct 17, 2013 at 11:53 AM, Bill Hatcher wrote:
>
>> Ok, I figured out my issue with the MWI not coming on in SRST more.  Need
>> to ass the key word unsolicited to the mwi-server command.
>>
>> Now to get the CME-SRST working.
>>
>>
>>
>> On Thu, Oct 17, 2013 at 10:36 AM, Bill Hatcher wrote:
>>
>>> Seifeddine,
>>>
>>> I've run that debug, but there is absolutly no output when I'm using
>>> CME-SRST.
>>>
>>> Bill
>>>
>>>
>>> On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili <
>>> seifeddine.tl...@lvs1.com> wrote:
>>>
  Can you send the output of debug ephone register?

 ** **

 Thx

 ** **

 *Kindly***

 * *

 *Seifeddine Tlili*  

 [image: Description: Description: Long View Systems]

 M.Eng CCIE # 26440
 Systems Consultant 

 .. *
 Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900

 [image: Description: Description: 
 Linkedin]
  [image: Description: Description: 
 Twitter]
  [image: Description: Description: 
 Facebook] [image:
 Description: Description: Facebook]
 www.longviewsystems.com
 This message and any attached documents are only for the use of
 the intended recipient(s), are confidential and may contain privileged
 information. Any unauthorized review, use, retransmission, or other
 disclosure is strictly prohibited. If you have received this message in
 error, notify the sender immediately, and delete the original message.
 

 ** **

 ** **

 ** **

 *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
 ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher
 *Sent:* Thursday, October 17, 2013 9:29 AM
 *To:* ccievoice
 *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help.

 ** **

 Marty,

 The weird thing is they work when I use call-manager-fallback.  Looking
 at the cnf file, all seems well.

 ** **

 On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan 
 wrote:

 I had a similar issue recently which ended up being a DB replication
 problem.  You could check the phones config file:

 10.10.210.11:6970/SEP123456789123.cnf.xml> Check
 subscribers copy of the file

 Right-click, select view source and search for 'srst' and see what it
 has there.  I could be missing something but it looks like you have enough
 to get them registered.

 Marty

 ** **

 On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher 
 wrote:

   It’s not working!!  Can anyone see something I may be doing wrong?
 My PRI and CUE register, I can even see SIP MWI being sent, but my phones
 will not register.  They worked when I was using call-manager-fallback
 though so I know my SRST configuration is correct on the CallManager.**
 **

  

 telephony-service

  srst mode auto-provision all

  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07
 

  srst dn line-mode octo

  max-ephones 4

  max-dn 4

  ip source-address 10.10.202.1 port 2000

  max-conferences 12 gain -6

  transfer-system full-consult

  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18

 ** **

 ** **

 ___
 For more information regarding industry leading CCIE Lab training,
 please visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com

  ** **

 ** **

 ** **

>>>
>>>
>>

Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.

[Martin Sloan]

Bill,

Here's a great reference for CME MWI:

http://ciscovoiceguru.com/518/cue-mwi-notification-methods/

I've used this a lot through my studies.

@Ramcharan - good call.  I think the symptom with that bug is that the
phones will register but display no DN's and if you issue 'show ephone reg'
it will show the DN's as 'invalid' or something like that.  I have also hit
the other bug with SC MGCP after coming out of SRST the dial-peers are
still chosen on inbound calls from the PSTN.  I believe the fix for that is
to globally set 'voice hunt 2' and under the ephone-dn's assign a higher
preference (I use 9).


On Thu, Oct 17, 2013 at 11:53 AM, Bill Hatcher  wrote:

> Ok, I figured out my issue with the MWI not coming on in SRST more.  Need
> to ass the key word unsolicited to the mwi-server command.
>
> Now to get the CME-SRST working.
>
>
>
> On Thu, Oct 17, 2013 at 10:36 AM, Bill Hatcher wrote:
>
>> Seifeddine,
>>
>> I've run that debug, but there is absolutly no output when I'm using
>> CME-SRST.
>>
>> Bill
>>
>>
>> On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili <
>> seifeddine.tl...@lvs1.com> wrote:
>>
>>>  Can you send the output of debug ephone register?
>>>
>>> ** **
>>>
>>> Thx
>>>
>>> ** **
>>>
>>> *Kindly***
>>>
>>> * *
>>>
>>> *Seifeddine Tlili*  
>>>
>>> [image: Description: Description: Long View Systems]
>>>
>>> M.Eng CCIE # 26440
>>> Systems Consultant 
>>>
>>> .. *
>>> Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900
>>>
>>> [image: Description: Description: 
>>> Linkedin]
>>>  [image: Description: Description: 
>>> Twitter]
>>>  [image: Description: Description: 
>>> Facebook] [image:
>>> Description: Description: Facebook]
>>> www.longviewsystems.com
>>> This message and any attached documents are only for the use of
>>> the intended recipient(s), are confidential and may contain privileged
>>> information. Any unauthorized review, use, retransmission, or other
>>> disclosure is strictly prohibited. If you have received this message in
>>> error, notify the sender immediately, and delete the original message. *
>>> ***
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher
>>> *Sent:* Thursday, October 17, 2013 9:29 AM
>>> *To:* ccievoice
>>> *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help.
>>>
>>> ** **
>>>
>>> Marty,
>>>
>>> The weird thing is they work when I use call-manager-fallback.  Looking
>>> at the cnf file, all seems well.
>>>
>>> ** **
>>>
>>> On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan 
>>> wrote:
>>>
>>> I had a similar issue recently which ended up being a DB replication
>>> problem.  You could check the phones config file:
>>>
>>> 10.10.210.11:6970/SEP123456789123.cnf.xml> Check
>>> subscribers copy of the file
>>>
>>> Right-click, select view source and search for 'srst' and see what it
>>> has there.  I could be missing something but it looks like you have enough
>>> to get them registered.
>>>
>>> Marty
>>>
>>> ** **
>>>
>>> On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher 
>>> wrote:
>>>
>>>   It’s not working!!  Can anyone see something I may be doing wrong? My
>>> PRI and CUE register, I can even see SIP MWI being sent, but my phones will
>>> not register.  They worked when I was using call-manager-fallback though so
>>> I know my SRST configuration is correct on the CallManager.
>>>
>>>  
>>>
>>> telephony-service
>>>
>>>  srst mode auto-provision all
>>>
>>>  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07
>>> 
>>>
>>>  srst dn line-mode octo
>>>
>>>  max-ephones 4
>>>
>>>  max-dn 4
>>>
>>>  ip source-address 10.10.202.1 port 2000
>>>
>>>  max-conferences 12 gain -6
>>>
>>>  transfer-system full-consult
>>>
>>>  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com
>>>
>>>  ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.

[Bill Hatcher]

Ok the re-boot worked and now I can register my phones to CME SRST on a WAN
failure, and the CUE MWI works!!

I did have to go in and edit the call-foward parameters on the ephone-dn
with voicemail, and add my cor-list info to all the lines.


On Thu, Oct 17, 2013 at 10:53 AM, Bill Hatcher  wrote:

> Ok, I figured out my issue with the MWI not coming on in SRST more.  Need
> to ass the key word unsolicited to the mwi-server command.
>
> Now to get the CME-SRST working.
>
>
>
> On Thu, Oct 17, 2013 at 10:36 AM, Bill Hatcher wrote:
>
>> Seifeddine,
>>
>> I've run that debug, but there is absolutly no output when I'm using
>> CME-SRST.
>>
>> Bill
>>
>>
>> On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili <
>> seifeddine.tl...@lvs1.com> wrote:
>>
>>>  Can you send the output of debug ephone register?
>>>
>>> ** **
>>>
>>> Thx
>>>
>>> ** **
>>>
>>> *Kindly***
>>>
>>> * *
>>>
>>> *Seifeddine Tlili*  
>>>
>>> [image: Description: Description: Long View Systems]
>>>
>>> M.Eng CCIE # 26440
>>> Systems Consultant 
>>>
>>> .. *
>>> Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900
>>>
>>> [image: Description: Description: 
>>> Linkedin]
>>>  [image: Description: Description: 
>>> Twitter]
>>>  [image: Description: Description: 
>>> Facebook] [image:
>>> Description: Description: Facebook]
>>> www.longviewsystems.com
>>> This message and any attached documents are only for the use of
>>> the intended recipient(s), are confidential and may contain privileged
>>> information. Any unauthorized review, use, retransmission, or other
>>> disclosure is strictly prohibited. If you have received this message in
>>> error, notify the sender immediately, and delete the original message. *
>>> ***
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher
>>> *Sent:* Thursday, October 17, 2013 9:29 AM
>>> *To:* ccievoice
>>> *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help.
>>>
>>> ** **
>>>
>>> Marty,
>>>
>>> The weird thing is they work when I use call-manager-fallback.  Looking
>>> at the cnf file, all seems well.
>>>
>>> ** **
>>>
>>> On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan 
>>> wrote:
>>>
>>> I had a similar issue recently which ended up being a DB replication
>>> problem.  You could check the phones config file:
>>>
>>> 10.10.210.11:6970/SEP123456789123.cnf.xml> Check
>>> subscribers copy of the file
>>>
>>> Right-click, select view source and search for 'srst' and see what it
>>> has there.  I could be missing something but it looks like you have enough
>>> to get them registered.
>>>
>>> Marty
>>>
>>> ** **
>>>
>>> On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher 
>>> wrote:
>>>
>>>   It’s not working!!  Can anyone see something I may be doing wrong? My
>>> PRI and CUE register, I can even see SIP MWI being sent, but my phones will
>>> not register.  They worked when I was using call-manager-fallback though so
>>> I know my SRST configuration is correct on the CallManager.
>>>
>>>  
>>>
>>> telephony-service
>>>
>>>  srst mode auto-provision all
>>>
>>>  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07
>>> 
>>>
>>>  srst dn line-mode octo
>>>
>>>  max-ephones 4
>>>
>>>  max-dn 4
>>>
>>>  ip source-address 10.10.202.1 port 2000
>>>
>>>  max-conferences 12 gain -6
>>>
>>>  transfer-system full-consult
>>>
>>>  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com
>>>
>>>  ** **
>>>
>>> ** **
>>>
>>> ** **
>>>
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.

[Bill Hatcher]

Ok, I figured out my issue with the MWI not coming on in SRST more.  Need
to ass the key word unsolicited to the mwi-server command.

Now to get the CME-SRST working.



On Thu, Oct 17, 2013 at 10:36 AM, Bill Hatcher  wrote:

> Seifeddine,
>
> I've run that debug, but there is absolutly no output when I'm using
> CME-SRST.
>
> Bill
>
>
> On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili <
> seifeddine.tl...@lvs1.com> wrote:
>
>>  Can you send the output of debug ephone register?
>>
>> ** **
>>
>> Thx
>>
>> ** **
>>
>> *Kindly***
>>
>> * *
>>
>> *Seifeddine Tlili*  
>>
>> [image: Description: Description: Long View Systems]
>>
>> M.Eng CCIE # 26440
>> Systems Consultant 
>>
>> .. *
>> Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900
>>
>> [image: Description: Description: 
>> Linkedin]
>>  [image: Description: Description: 
>> Twitter]
>>  [image: Description: Description: 
>> Facebook] [image:
>> Description: Description: Facebook]
>> www.longviewsystems.com
>> This message and any attached documents are only for the use of
>> the intended recipient(s), are confidential and may contain privileged
>> information. Any unauthorized review, use, retransmission, or other
>> disclosure is strictly prohibited. If you have received this message in
>> error, notify the sender immediately, and delete the original message. **
>> **
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher
>> *Sent:* Thursday, October 17, 2013 9:29 AM
>> *To:* ccievoice
>> *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help.
>>
>> ** **
>>
>> Marty,
>>
>> The weird thing is they work when I use call-manager-fallback.  Looking
>> at the cnf file, all seems well.
>>
>> ** **
>>
>> On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan 
>> wrote:
>>
>> I had a similar issue recently which ended up being a DB replication
>> problem.  You could check the phones config file:
>>
>> 10.10.210.11:6970/SEP123456789123.cnf.xml> Check subscribers
>> copy of the file
>>
>> Right-click, select view source and search for 'srst' and see what it has
>> there.  I could be missing something but it looks like you have enough to
>> get them registered.
>>
>> Marty
>>
>> ** **
>>
>> On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher 
>> wrote:
>>
>>   It’s not working!!  Can anyone see something I may be doing wrong? My
>> PRI and CUE register, I can even see SIP MWI being sent, but my phones will
>> not register.  They worked when I was using call-manager-fallback though so
>> I know my SRST configuration is correct on the CallManager.
>>
>>  
>>
>> telephony-service
>>
>>  srst mode auto-provision all
>>
>>  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07*
>> ***
>>
>>  srst dn line-mode octo
>>
>>  max-ephones 4
>>
>>  max-dn 4
>>
>>  ip source-address 10.10.202.1 port 2000
>>
>>  max-conferences 12 gain -6
>>
>>  transfer-system full-consult
>>
>>  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18
>>
>> ** **
>>
>> ** **
>>
>> ___
>> 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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www.ipexpert.com

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Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.

[Bill Hatcher]

Seifeddine,

I've run that debug, but there is absolutly no output when I'm using
CME-SRST.

Bill


On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili <
seifeddine.tl...@lvs1.com> wrote:

>  Can you send the output of debug ephone register?
>
> ** **
>
> Thx
>
> ** **
>
> *Kindly***
>
> * *
>
> *Seifeddine Tlili*  
>
> [image: Description: Description: Long View Systems]
>
> M.Eng CCIE # 26440
> Systems Consultant 
>
> .. *
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> This message and any attached documents are only for the use of
> the intended recipient(s), are confidential and may contain privileged
> information. Any unauthorized review, use, retransmission, or other
> disclosure is strictly prohibited. If you have received this message in
> error, notify the sender immediately, and delete the original message. ***
> *
>
> ** **
>
> ** **
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher
> *Sent:* Thursday, October 17, 2013 9:29 AM
> *To:* ccievoice
> *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help.
>
> ** **
>
> Marty,
>
> The weird thing is they work when I use call-manager-fallback.  Looking at
> the cnf file, all seems well.
>
> ** **
>
> On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan 
> wrote:
>
> I had a similar issue recently which ended up being a DB replication
> problem.  You could check the phones config file:
>
> 10.10.210.11:6970/SEP123456789123.cnf.xml> Check subscribers
> copy of the file
>
> Right-click, select view source and search for 'srst' and see what it has
> there.  I could be missing something but it looks like you have enough to
> get them registered.
>
> Marty
>
> ** **
>
> On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher 
> wrote:
>
>   It’s not working!!  Can anyone see something I may be doing wrong? My
> PRI and CUE register, I can even see SIP MWI being sent, but my phones will
> not register.  They worked when I was using call-manager-fallback though so
> I know my SRST configuration is correct on the CallManager.
>
>  
>
> telephony-service
>
>  srst mode auto-provision all
>
>  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07**
> **
>
>  srst dn line-mode octo
>
>  max-ephones 4
>
>  max-dn 4
>
>  ip source-address 10.10.202.1 port 2000
>
>  max-conferences 12 gain -6
>
>  transfer-system full-consult
>
>  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18
>
> ** **
>
> ** **
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>  ** **
>
> ** **
>
> ** **
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] CME as SRST help.

[Bill Hatcher]

The reason I was even trying to register my phones in CME SRST is begause I
cann't seem to get the CUE MWI to work in SRST and I wanted to see if I was
missing something.  I get a SIP 481 Call Leg/Transaction Does Not Exist
when my CME sends mwi.  In researching this issue I found that if the voice
port does not have the MWI command on it, you'll get this error.  Looking
at the ephone-dn's created by SRST I see mwi sip on the DN.  Below is the
config I'm using.

Result of show call-manager-fallback
ephone-dn 3 octo-line
number 1010 no-reg primary
name 6178631010
description 6178631010
preference 0 secondary 9
huntstop
huntstop channel 0
call-forward busy 3600
call-forward noan 3600 timeout 12
call-waiting beep
cor incoming css-inter
mwi sip
no cti notify
no cti watch

Relevant config
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
!
ccm-manager fallback-mgcp
!
dial-peer voice 3600 voip
 destination-pattern 3[16]00
 session protocol sipv2
 session target ipv4:10.10.202.2
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
!
sip-ua
 mwi-server ipv4:10.10.202.2 expires 3600 port 5060 transport udp
!
call-manager-fallback
 max-conferences 12 gain -6
 transfer-system full-consult
 ip source-address 10.10.202.1 port 2000
 max-ephones 4
 max-dn 4 octo-line no-reg
 voicemail 3600
 call-forward busy 3600
 call-forward noan 3600 timeout 12
 mwi relay
 moh music-on-hold.au
 cor incoming css-inter default


On Thu, Oct 17, 2013 at 10:29 AM, Bill Hatcher  wrote:

> Ramcharan,
>
> I'll give that a try.
>
>
> On Thu, Oct 17, 2013 at 10:18 AM, Ramcharan Arya  > wrote:
>
>> Hi Bill,
>>
>> I believe this might be related to a bug with using octo lines in CME
>> SRST.
>>
>> Come out of SRST and reload the router this might resolve the issue.
>>
>> Regards,
>> Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)
>>
>>
>>
>>
>> On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher wrote:
>>
>>> It’s not working!!  Can anyone see something I may be doing wrong? My
>>> PRI and CUE register, I can even see SIP MWI being sent, but my phones will
>>> not register.  They worked when I was using call-manager-fallback
>>> though so I know my SRST configuration is correct on the CallManager.
>>>
>>>
>>>
>>> telephony-service
>>>
>>>  srst mode auto-provision all
>>>
>>>  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07
>>>
>>>  srst dn line-mode octo
>>>
>>>  max-ephones 4
>>>
>>>  max-dn 4
>>>
>>>  ip source-address 10.10.202.1 port 2000
>>>
>>>  max-conferences 12 gain -6
>>>
>>>  transfer-system full-consult
>>>
>>>  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18
>>>
>>>
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training,
>>> please visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out
>>> www.PlatinumPlacement.com
>>>
>>
>>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

[OSL | CCIE_Voice] Fwd: CME as SRST help.

[Bill Hatcher]

Marty,

The weird thing is they work when I use call-manager-fallback.  Looking at
the cnf file, all seems well.


On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan wrote:

> I had a similar issue recently which ended up being a DB replication
> problem.  You could check the phones config file:
>
> 10.10.210.11:6970/SEP123456789123.cnf.xml> Check subscribers
> copy of the file
>
> Right-click, select view source and search for 'srst' and see what it has
> there.  I could be missing something but it looks like you have enough to
> get them registered.
>
> Marty
>
>
> On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher wrote:
>
>>  It’s not working!!  Can anyone see something I may be doing wrong? My
>> PRI and CUE register, I can even see SIP MWI being sent, but my phones will
>> not register.  They worked when I was using call-manager-fallback though
>> so I know my SRST configuration is correct on the CallManager.
>>
>>
>>
>> telephony-service
>>
>>  srst mode auto-provision all
>>
>>  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07
>>
>>  srst dn line-mode octo
>>
>>  max-ephones 4
>>
>>  max-dn 4
>>
>>  ip source-address 10.10.202.1 port 2000
>>
>>  max-conferences 12 gain -6
>>
>>  transfer-system full-consult
>>
>>  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] CME as SRST help.

[Bill Hatcher]

Ramcharan,

I'll give that a try.


On Thu, Oct 17, 2013 at 10:18 AM, Ramcharan Arya
wrote:

> Hi Bill,
>
> I believe this might be related to a bug with using octo lines in CME SRST.
>
> Come out of SRST and reload the router this might resolve the issue.
>
> Regards,
> Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)
>
>
>
>
> On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher  wrote:
>
>> It’s not working!!  Can anyone see something I may be doing wrong? My
>> PRI and CUE register, I can even see SIP MWI being sent, but my phones will
>> not register.  They worked when I was using call-manager-fallback though
>> so I know my SRST configuration is correct on the CallManager.
>>
>>
>>
>> telephony-service
>>
>>  srst mode auto-provision all
>>
>>  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07
>>
>>  srst dn line-mode octo
>>
>>  max-ephones 4
>>
>>  max-dn 4
>>
>>  ip source-address 10.10.202.1 port 2000
>>
>>  max-conferences 12 gain -6
>>
>>  transfer-system full-consult
>>
>>  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] CME as SRST help.

[Ramcharan Arya]

Hi Bill,

I believe this might be related to a bug with using octo lines in CME SRST.

Come out of SRST and reload the router this might resolve the issue.

Regards,
Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)




On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher  wrote:

> It’s not working!!  Can anyone see something I may be doing wrong? My PRI
> and CUE register, I can even see SIP MWI being sent, but my phones will not
> register.  They worked when I was using call-manager-fallback though so I
> know my SRST configuration is correct on the CallManager.
>
>
>
> telephony-service
>
>  srst mode auto-provision all
>
>  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07
>
>  srst dn line-mode octo
>
>  max-ephones 4
>
>  max-dn 4
>
>  ip source-address 10.10.202.1 port 2000
>
>  max-conferences 12 gain -6
>
>  transfer-system full-consult
>
>  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] CME as SRST help.

[Martin Sloan]

I had a similar issue recently which ended up being a DB replication
problem.  You could check the phones config file:

10.10.210.11:6970/SEP123456789123.cnf.xml> Check subscribers
copy of the file

Right-click, select view source and search for 'srst' and see what it has
there.  I could be missing something but it looks like you have enough to
get them registered.

Marty


On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher  wrote:

> It’s not working!!  Can anyone see something I may be doing wrong? My PRI
> and CUE register, I can even see SIP MWI being sent, but my phones will not
> register.  They worked when I was using call-manager-fallback though so I
> know my SRST configuration is correct on the CallManager.
>
>
>
> telephony-service
>
>  srst mode auto-provision all
>
>  srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07
>
>  srst dn line-mode octo
>
>  max-ephones 4
>
>  max-dn 4
>
>  ip source-address 10.10.202.1 port 2000
>
>  max-conferences 12 gain -6
>
>  transfer-system full-consult
>
>  create cnf-files version-stamp 7960 Oct 17 2013 16:07:18
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

[OSL | CCIE_Voice] CME as SRST help.

[Bill Hatcher]

It’s not working!!  Can anyone see something I may be doing wrong? My PRI
and CUE register, I can even see SIP MWI being sent, but my phones will not
register.  They worked when I was using call-manager-fallback though so I
know my SRST configuration is correct on the CallManager.



telephony-service

 srst mode auto-provision all

 srst ephone description auto provisioned ephone  : Oct 17 2013 14:47:07

 srst dn line-mode octo

 max-ephones 4

 max-dn 4

 ip source-address 10.10.202.1 port 2000

 max-conferences 12 gain -6

 transfer-system full-consult

 create cnf-files version-stamp 7960 Oct 17 2013 16:07:18
___
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www.PlatinumPlacement.com

[OSL | CCIE_Voice] Testing SRST

[Bill Hatcher]

I have been looking for quick and easy ways to test SRST, and I've found
many different waqys of doing this.  With the exception of pulling the WAN
interface, they all seem to take a lot of time and effort to accomplish.
Anything from creating access-lists to block the traffic to creating new
call manager groups and shutting down one of the CallManager services.

I have found that a couple of simple static routes to the null 0 interface
works very well.
ip route 10.10.210.10 255.255.255.255 null 0
ip route 10.10.210.11 255.255.255.255 null 0

no ip route 10.10.210.10 255.255.255.255 null 0
no ip route 10.10.210.11 255.255.255.255 null 0

Add them to a notepad and the no statements as well and you can quickly
send your devices into srst mode. Now if you have any VoIP dial-peers that
point to other addresses across your WAN you may have to add those as well.

What do you guys use?

HTH

Bill.
___
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