Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

[Hamu Ebiso]

Hello Anthony and all,

I have found more information about this.

3 call handlers are involved with this.

Location 1 Main call handler
Clasified department  call handler
Location Main call handler

call to location 1 Main call handler >> option 3 to Classified advertising >> 
option 1 to going back to Nortel phone system and works fine.
Call to location 2 Main call handler >> option 2 to classified advertising >> 
option 1 go back to Nortel phone system. Plays wait while I transfer and then 
fail.

Location on SIP trunk and CUCM
Location 1 is on PRI and Nortel phone system

some how Location 1 phone system is connected to Unity connection.

Dial-peer is configured with DTMF correctly

Everything in unity is configured correctly but the transfer doesn't work.

thanks
Hamu


From: Anthony Holloway 
Sent: Friday, April 24, 2020 9:32 AM
To: Hamu Ebiso 
Cc: cisco-voip@puck.nether.net 
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not 
working correctly.

Hamu,

Are you saying then, that the issue is not DTMF? Does the system take your 
input without error?

There's no magic bullet to fix transfers, you need to see what's happening, and 
prescribe a fix.  Can you share the SIP flow from the CUBE for the entire call 
duration?  Feel free to censor the sensitive bits.

On Fri, Apr 24, 2020 at 9:25 AM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
Thank you very much for your help here. Is there anyway you could share the few 
setting I could change on CUBE or CUCM if the issue is transffering?

thanks



From: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Friday, April 24, 2020 8:55 AM
To: Hamu Ebiso mailto:hebiso2...@hotmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not 
working correctly.

The reason I ask is that the troubleshooting is a little different for each 
issue.

DTMF

You would know if it's DTMF if for example, you push the button and the voice 
recording just keeps on going.  Most recordings are set such that if you barge 
in on them, the recording ends abruptly to process your input.

OR

You would know if it's a DTMF issue if for example, you press a button and CUC 
processes it twice, as in double digits.  This might be a little harder to tell 
from UX, but it might be easier if you setup a test number to the Opening 
Greeting and pressing * exists the app, versus taking you to Login.

Transfer

You would know if it's a Transfer issue, if it wasn't a DTMF issue.  I.e., You 
press the button, the recording stops, or even says, "Wait while I transfer 
your call" and then the failure happens.

Transfer failures could happen for a few different reasons, and there's a few 
settings on CUBE and within CUCM which can affect how a transfer functions, 
thus improving success with each knob turned.

On Fri, Apr 24, 2020 at 7:26 AM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
I was thinking it might be Transfer issue. What makes you ask that question 
Anthony?

thanks
Hamu


From: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, April 23, 2020 2:54 PM
To: Hamu Ebiso mailto:hebiso2...@hotmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not 
working correctly.

Is this a DTMF issue, or a transfer issue?

On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
Hello team,

I hope someone have come across this issue and can help me. We ported our 
numbers to SIP yesterday. Now, their main menu is not transferring numbers 
correctly.  For example, when you select classifieds, it is supposed to go to 
the LAC Classifieds call handler.  Selecting option 1 is not routing correctly. 
Calling the LAC numbers directly works.
What do you think might be causing this issue?

Thanks
Hamu
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Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

[Hamu Ebiso]

Thank you Jason for your questions. how can you setup rfc2833 In CUCM trunks?

thanks


From: Jason Aarons 
Sent: Friday, April 24, 2020 8:24 AM
To: Hamu Ebiso 
Cc: cisco-voip 
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not 
working correctly.

I am doubtful porting had anything to do with it. Was it tested fully before 
the port?

Under dial peers is dtmf-relay rtp-nte set? In CUCM trunks is rfc2833 set? How 
is Unity integrated with CUCM ? SIP? CXN Version?

Without some debugs /traces I suspect you won't find much.

On Thu, Apr 23, 2020, 3:52 PM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
Hello team,

I hope someone have come across this issue and can help me. We ported our 
numbers to SIP yesterday. Now, their main menu is not transferring numbers 
correctly.  For example, when you select classifieds, it is supposed to go to 
the LAC Classifieds call handler.  Selecting option 1 is not routing correctly. 
Calling the LAC numbers directly works.
What do you think might be causing this issue?

Thanks
Hamu
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Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

[Hamu Ebiso]

Thank you very much for your help here. Is there anyway you could share the few 
setting I could change on CUBE or CUCM if the issue is transffering?

thanks



From: Anthony Holloway 
Sent: Friday, April 24, 2020 8:55 AM
To: Hamu Ebiso 
Cc: cisco-voip@puck.nether.net 
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not 
working correctly.

The reason I ask is that the troubleshooting is a little different for each 
issue.

DTMF

You would know if it's DTMF if for example, you push the button and the voice 
recording just keeps on going.  Most recordings are set such that if you barge 
in on them, the recording ends abruptly to process your input.

OR

You would know if it's a DTMF issue if for example, you press a button and CUC 
processes it twice, as in double digits.  This might be a little harder to tell 
from UX, but it might be easier if you setup a test number to the Opening 
Greeting and pressing * exists the app, versus taking you to Login.

Transfer

You would know if it's a Transfer issue, if it wasn't a DTMF issue.  I.e., You 
press the button, the recording stops, or even says, "Wait while I transfer 
your call" and then the failure happens.

Transfer failures could happen for a few different reasons, and there's a few 
settings on CUBE and within CUCM which can affect how a transfer functions, 
thus improving success with each knob turned.

On Fri, Apr 24, 2020 at 7:26 AM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
I was thinking it might be Transfer issue. What makes you ask that question 
Anthony?

thanks
Hamu


From: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, April 23, 2020 2:54 PM
To: Hamu Ebiso mailto:hebiso2...@hotmail.com>>
Cc: cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net> 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not 
working correctly.

Is this a DTMF issue, or a transfer issue?

On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
Hello team,

I hope someone have come across this issue and can help me. We ported our 
numbers to SIP yesterday. Now, their main menu is not transferring numbers 
correctly.  For example, when you select classifieds, it is supposed to go to 
the LAC Classifieds call handler.  Selecting option 1 is not routing correctly. 
Calling the LAC numbers directly works.
What do you think might be causing this issue?

Thanks
Hamu
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Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

[Hamu Ebiso]

I was thinking it might be Transfer issue. What makes you ask that question 
Anthony?

thanks
Hamu


From: Anthony Holloway 
Sent: Thursday, April 23, 2020 2:54 PM
To: Hamu Ebiso 
Cc: cisco-voip@puck.nether.net 
Subject: Re: [cisco-voip] Ported Numbers to SIP call handler transfer is not 
working correctly.

Is this a DTMF issue, or a transfer issue?

On Thu, Apr 23, 2020 at 2:51 PM Hamu Ebiso 
mailto:hebiso2...@hotmail.com>> wrote:
Hello team,

I hope someone have come across this issue and can help me. We ported our 
numbers to SIP yesterday. Now, their main menu is not transferring numbers 
correctly.  For example, when you select classifieds, it is supposed to go to 
the LAC Classifieds call handler.  Selecting option 1 is not routing correctly. 
Calling the LAC numbers directly works.
What do you think might be causing this issue?

Thanks
Hamu
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[cisco-voip] Ported Numbers to SIP call handler transfer is not working correctly.

[Hamu Ebiso]

Hello team,

I hope someone have come across this issue and can help me. We ported our 
numbers to SIP yesterday. Now, their main menu is not transferring numbers 
correctly.  For example, when you select classifieds, it is supposed to go to 
the LAC Classifieds call handler.  Selecting option 1 is not routing correctly. 
Calling the LAC numbers directly works.
What do you think might be causing this issue?

Thanks
Hamu
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Re: [cisco-voip] 3. Cisco and Avaya dial plan integration. (Hamu Ebiso)

[Hamu Ebiso]

Thank you very much for your advice Amit!!


From: cisco-voip  on behalf of Amit Katyal 

Sent: Wednesday, April 17, 2019 11:30 AM
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 3. Cisco and Avaya dial plan integration. (Hamu Ebiso)

Hi Hamu,

In these cases, you will have to dig deep into dial plan and findout whether
1.what numbers are used for dialing PSTN ,
2.for internal calling
3.if there is any site codes on both sides.

after analysis, you can either opt for masking the extension with a prefix or 
using site codes between both systems.

thanks,
Amit

On Wed, Apr 17, 2019 at 9:40 PM 
mailto:cisco-voip-requ...@puck.nether.net>> 
wrote:
Send cisco-voip mailing list submissions to
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Today's Topics:

   1. Re: One of the silliest bugs ever (Anthony Holloway)
   2. Re: CCIE Collaboration V2 (Nick Britt)
   3. Cisco and Avaya dial plan integration. (Hamu Ebiso)


--

Message: 1
Date: Tue, 16 Apr 2019 11:19:14 -0500
From: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>
To: Ryan Huff mailto:ryanh...@outlook.com>>
Cc: "voyp list, cisco-voip" 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] One of the silliest bugs ever
Message-ID:

mailto:avsuw_m-i...@mail.gmail.com>>
Content-Type: text/plain; charset="utf-8"

And here I thought only UCCX had username case sensitivity issues.

Damn.  We're in 2019 now, and we're still being bamboozled by upper and
lower case letters.  It's too bad we can't extend the web design mantra of
"separate content from presentation" at a much lower level, such as the
case of letters.

On Mon, Apr 15, 2019 at 8:09 PM Ryan Huff 
mailto:ryanh...@outlook.com>> wrote:

> This is just crazy...
>
> https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvh72242
>
> -Ryan
> ___
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> cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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--

Message: 2
Date: Tue, 16 Apr 2019 09:29:01 -0700
From: Nick Britt mailto:nickolasjbr...@gmail.com>>
To: Benjamin Turner mailto:benmtur...@hotmail.com>>
Cc: Anthony Holloway 
mailto:avholloway%2bcisco-v...@gmail.com>>,  
cisco-voip
voyp list 
mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] CCIE Collaboration V2
Message-ID:

mailto:cakss23k9s8wcfvzvnyls-djaomep4mr6pr5zgvmqyeejvhh...@mail.gmail.com>>
Content-Type: text/plain; charset="utf-8"

Buy cheap, Buy twice.

If you are committed to getting the cert, go with Vik (in person if
possible), it will save you a huge amount of time/money in the long run.

On Thu, Apr 4, 2019 at 5:49 AM Benjamin Turner 
mailto:benmtur...@hotmail.com>>
wrote:

> Also,
>
>
>
> Kevin Wallace is soon to release the Collab V2 video and workbook training
> soon.
>
>
>
>
>
>
>
> Sent from Mail <https://go.microsoft.com/fwlink/?LinkId=550986> for
> Windows 10
>
>
> --
> *From:* Anthony Holloway 
> mailto:avholloway%2bcisco-v...@gmail.com>>
> *Sent:* Wednesday, April 3, 2019 11:02:47 PM
> *To:* Benjamin Turner
> *Cc:* Fares Alsaafani; cisco-voip voyp list
> *Subject:* Re: [cisco-voip] CCIE Collaboration V2
>
> Collab Cert use to sell a written study guide, and I wouldn't see why he
> wouldn't do the same for the V2.  Maybe it would be worth shooting them an
> email and asking about it.  FWIW, I never actually saw the product, I only
> saw it for sale on the site, so I cannot say if it was helpful or worth
> it.  Either way, email them, Vik has built a business and based his very
> long career in helping people attaining their CCIE; I'm sure he can help in
> some way.
>
> On Wed, Apr 3, 2019 at 10:58 AM Benjamin Turner 
> mailto:benmtur...@hotmail.com>>
> wrote:
>
>> Good luck. I'm in the same boat. The only site I can find 

[cisco-voip] Cisco and Avaya dial plan integration.

[Hamu Ebiso]

Hello everyone,

I work for the company that uses cisco phone system. We were merged with the 
company that uses Avaya phone system. Down the road we will be moving Avaya 
system to cisco phone system and that will be near future. In the mean time, we 
would like to integrate of the dial plan so that end users from both phone 
system will be able to call each other easily. Cisco sites have 10 digit dial 
plan and Avaya sites have 5 digit dial plan. I'm only familiar with Cisco phone 
system.

Has anyone done something like this and have some document to share with me?
I'm trying to understand what is required for the dial-plan integration of 2 
phone system.
CUCM is centralized cluster and Avaya is I believe Hub and Spoke for the most 
part.

thanks
Hamu
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[cisco-voip] Turn off LAN connection to Remote Location.

[Hamu Ebiso]

Hello everyone,

I have one question I hope someone will be able to answer.

We have main location and 1 remote location residing in different building. The 
remote location uses main location PRI and DID for both inbound and outbound 
calls. We have placed 2921 router for 911 calls with few POTS line and they 
also use POTS line for their few faxes.

The biggest question is, what will happen when we turn off the LAN connections 
back to the datacenter and place them on our WAN DMVPN cloud?  Datacenter is 
located in main building.
I hope someone will be able to help me answer this question.


thanks

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[cisco-voip] Spectralink 8440 phones registration status on CUCM

[Hamu Ebiso]

Hi Team,


Has anyone experienced this issue before and know how to fix it. I have been 
trying to figure out how to setup redundancy for spectralink wireless phones so 
that whenever one of the server down phones will failover. I was able finally 
figure out redundancy part, but I don't see phones status in CUCM. Even though 
hand held spectralink show registered and happy, CUCM shows unknown status. I 
don't know what I'm missing and or if CUCM is not cable of showing Status for 
redundancy phones? your help is greatly appreciated as usual.


thanks

Hamu
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Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

Thank you Anthony very much. I really appreciated your guidance. I was able to 
make work with the sip setup and incoming calls are now working fine but it 
doesn't have caller ID. I have attached the logs.


below is the config.


control-plane
!
!
voice-port 0/2/0
 trunk-group FXO_EM
 no battery-reversal
 input gain 1
 echo-cancel mode 1
 no vad
 no comfort-noise
 connection plar 12089588038
 impedance 900r
 description 208-343-0207
 caller-id enable
!
voice-port 0/2/1
 trunk-group FXO_EM
 connection plar 12089588038
 description 208-343-3497
 caller-id enable


dial-peer voice 2 voip
 preference 1
 destination-pattern 1208958
 session protocol sipv2
 session target ipv4:10.170.99.12
 voice-class codec 1
 voice-class sip options-keepalive up-interval 12 down-interval 65 retry 2
 dtmf-relay rtp-nte
 no vad



From: Anthony Holloway <avholloway+cisco-v...@gmail.com>
Sent: Friday, March 2, 2018 9:52 AM
To: Hamu Ebiso
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

Nope.  It's SIP or die dude.

You already have a SIP dial-peer pointing at CUCM:

dial-peer voice 2 voip
 description CUCM Dial-Peer - Inbound
 preference 1
 destination-pattern 12086858038<tel:(208)%20685-8038>
 session protocol sipv2
 session target ipv4:10.0.2.6
 incoming uri via 10
 voice-class codec 1
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 fax nsf 00
 ip qos dscp cs3 signaling
 no vad
!

You still have an issue with the number you're dialing and what's in your 
destination pattern.  But if you use the above dial peer config on dial-peer 5, 
then maybe you'll actually send a SIP INVITE to CUCM.

I just want to say, there are bigger issues here than just your POTS call to 
CIPC.  I work for a Cisco Partner in your city, and if you want, we can talk 
about having me come help you in person on this project. Let me know.

On Fri, Mar 2, 2018 at 9:12 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Hi Anthony,


do you think I should change the setting between CUCM and Gateway to H.323 
instead of SIP? do you think that makes it easy?


Thanks

Hamu



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, March 1, 2018 1:16 PM
To: Hamu Ebiso

Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
debug voip ccapi inout, and if you're using SIP, then debug ccsip messages as 
well.  the debug vpm signal is not very helpful, though it does show the port 
selected.

On Thu, Mar 1, 2018 at 1:05 PM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Thank you Anthony very much for your help.

The 19529473283<tel:(952)%20947-3283> for  connection plar 
19529473283<tel:(952)%20947-3283> I was testing with. After changing of few 
things on the gateway, I'm now seeing different messages when I run debug. The 
calls used to ring only one time and then fast busy, now it rings twice and 
then fax busy for incoming. for outgoing, it's says calls connot be completed 
as dialed.


Attached the new logs again. I really appreciate your help.


thanks again.

Hamu



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, March 1, 2018 11:05 AM

To: Hamu Ebiso
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
According to the debug, you're calling 19529473283<tel:(952)%20947-3283>.  Is 
that correct?


Called Number=19529473283<tel:(952)%20947-3283>(TON=Unknown, NPI=Unknown),

If so, then this debug is not for this gateway, because you're hitting Outgoing 
Dial-peer=5


Outgoing Dial-peer=5, Params=0x688FF3FC, Progress Indication=ORIGINATING SIDE 
IS NON ISDN(3)

And dial-peer 5, while it exists in your config you pasted, it doesn't match 
the called number.

dial-peer voice 5 voip
 description FXO test
 destination-pattern 12083437020<tel:(208)%20343-7020>
 session target ipv4:10.0.2.5
!

Even if it did match the destination pattern, you're missing some config on 
this dial-peer to make a SIP capable dial-peer, which means it's sending H323 
setup to CUCM.  If you built the gateway as H323, then I can see how it rang 
your CIPC, otherwise, if you built it as SIP, which you said you did, then it 
would never ring your CIPC.

So, something is not adding up here.

Also, you're hitting Incoming Dial-peer=0, which on a POTS leg is not too 
terrible, but it's also not good practice to be using dial-peer 0.  It's really 
VoIP DP 0 that is messy though.


Incoming Dial-peer=0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), 
Calli

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

let me change the config setting in CUCM to H.323 and let you know.


thank you for your help again.


thanks

Hamu



From: Anthony Holloway <avholloway+cisco-v...@gmail.com>
Sent: Friday, March 2, 2018 9:07 AM
To: Hamu Ebiso; Cisco VoIP Group
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

Unfortunately, again, the number you're calling 4131, is not matching the 
dial-peers you posted; however, the debugs are still saying you're hitting DP 
5, which is an H323 DP not SIP.

On Thu, Mar 1, 2018 at 1:44 PM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:


Here is the debug output for debug voip ccapi inout, and debug ccsip messages.


Thank you again for your help.




From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, March 1, 2018 1:16 PM
To: Hamu Ebiso

Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
debug voip ccapi inout, and if you're using SIP, then debug ccsip messages as 
well.  the debug vpm signal is not very helpful, though it does show the port 
selected.

On Thu, Mar 1, 2018 at 1:05 PM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Thank you Anthony very much for your help.

The 19529473283<tel:(952)%20947-3283> for  connection plar 
19529473283<tel:(952)%20947-3283> I was testing with. After changing of few 
things on the gateway, I'm now seeing different messages when I run debug. The 
calls used to ring only one time and then fast busy, now it rings twice and 
then fax busy for incoming. for outgoing, it's says calls connot be completed 
as dialed.


Attached the new logs again. I really appreciate your help.


thanks again.

Hamu



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, March 1, 2018 11:05 AM

To: Hamu Ebiso
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
According to the debug, you're calling 19529473283<tel:(952)%20947-3283>.  Is 
that correct?


Called Number=19529473283<tel:(952)%20947-3283>(TON=Unknown, NPI=Unknown),

If so, then this debug is not for this gateway, because you're hitting Outgoing 
Dial-peer=5


Outgoing Dial-peer=5, Params=0x688FF3FC, Progress Indication=ORIGINATING SIDE 
IS NON ISDN(3)

And dial-peer 5, while it exists in your config you pasted, it doesn't match 
the called number.

dial-peer voice 5 voip
 description FXO test
 destination-pattern 12083437020<tel:(208)%20343-7020>
 session target ipv4:10.0.2.5
!

Even if it did match the destination pattern, you're missing some config on 
this dial-peer to make a SIP capable dial-peer, which means it's sending H323 
setup to CUCM.  If you built the gateway as H323, then I can see how it rang 
your CIPC, otherwise, if you built it as SIP, which you said you did, then it 
would never ring your CIPC.

So, something is not adding up here.

Also, you're hitting Incoming Dial-peer=0, which on a POTS leg is not too 
terrible, but it's also not good practice to be using dial-peer 0.  It's really 
VoIP DP 0 that is messy though.


Incoming Dial-peer=0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), 
Calling IE Present=FALSE,


On Thu, Mar 1, 2018 at 10:06 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Hi Anthony,


yes the call was hitting the gateway because I was seeing FXO port going 
Off-Hook. The problem was that Logging was not setup in the gateway. After I 
setup the Logging, I'm now seeing logs.


I have attached the log.

thanks

Hamu


From: avhollo...@gmail.com<mailto:avhollo...@gmail.com> 
<avhollo...@gmail.com<mailto:avhollo...@gmail.com>> on behalf of Anthony 
Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, March 1, 2018 8:45 AM
To: Hamu Ebiso
Cc: Cisco VoIP Group

Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
Since you didn't see the call in your debugs, your call likely didn't hit your 
gateway.

Can you confirm how you made your test call, when you had your debug running?

On Feb 28, 2018 2:46 PM, "Hamu Ebiso" 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Here is the dial-peer output

1#show run | sec dial-peer
dial-peer voice 1 pots
 trunkgroup FXO_EM
 translation-profile incoming Inbound
 call-block translation-profile incoming block_profile
 call-block disconnect-cause incoming call-reject
 incoming called-number .
 dir

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

Hi Anthony,


do you think I should change the setting between CUCM and Gateway to H.323 
instead of SIP? do you think that makes it easy?


Thanks

Hamu



From: Anthony Holloway <avholloway+cisco-v...@gmail.com>
Sent: Thursday, March 1, 2018 1:16 PM
To: Hamu Ebiso
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

debug voip ccapi inout, and if you're using SIP, then debug ccsip messages as 
well.  the debug vpm signal is not very helpful, though it does show the port 
selected.

On Thu, Mar 1, 2018 at 1:05 PM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Thank you Anthony very much for your help.

The 19529473283<tel:(952)%20947-3283> for  connection plar 
19529473283<tel:(952)%20947-3283> I was testing with. After changing of few 
things on the gateway, I'm now seeing different messages when I run debug. The 
calls used to ring only one time and then fast busy, now it rings twice and 
then fax busy for incoming. for outgoing, it's says calls connot be completed 
as dialed.


Attached the new logs again. I really appreciate your help.


thanks again.

Hamu



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, March 1, 2018 11:05 AM

To: Hamu Ebiso
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
According to the debug, you're calling 19529473283<tel:(952)%20947-3283>.  Is 
that correct?


Called Number=19529473283<tel:(952)%20947-3283>(TON=Unknown, NPI=Unknown),

If so, then this debug is not for this gateway, because you're hitting Outgoing 
Dial-peer=5


Outgoing Dial-peer=5, Params=0x688FF3FC, Progress Indication=ORIGINATING SIDE 
IS NON ISDN(3)

And dial-peer 5, while it exists in your config you pasted, it doesn't match 
the called number.

dial-peer voice 5 voip
 description FXO test
 destination-pattern 12083437020<tel:(208)%20343-7020>
 session target ipv4:10.0.2.5
!

Even if it did match the destination pattern, you're missing some config on 
this dial-peer to make a SIP capable dial-peer, which means it's sending H323 
setup to CUCM.  If you built the gateway as H323, then I can see how it rang 
your CIPC, otherwise, if you built it as SIP, which you said you did, then it 
would never ring your CIPC.

So, something is not adding up here.

Also, you're hitting Incoming Dial-peer=0, which on a POTS leg is not too 
terrible, but it's also not good practice to be using dial-peer 0.  It's really 
VoIP DP 0 that is messy though.


Incoming Dial-peer=0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), 
Calling IE Present=FALSE,


On Thu, Mar 1, 2018 at 10:06 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Hi Anthony,


yes the call was hitting the gateway because I was seeing FXO port going 
Off-Hook. The problem was that Logging was not setup in the gateway. After I 
setup the Logging, I'm now seeing logs.


I have attached the log.

thanks

Hamu


From: avhollo...@gmail.com<mailto:avhollo...@gmail.com> 
<avhollo...@gmail.com<mailto:avhollo...@gmail.com>> on behalf of Anthony 
Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, March 1, 2018 8:45 AM
To: Hamu Ebiso
Cc: Cisco VoIP Group

Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
Since you didn't see the call in your debugs, your call likely didn't hit your 
gateway.

Can you confirm how you made your test call, when you had your debug running?

On Feb 28, 2018 2:46 PM, "Hamu Ebiso" 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Here is the dial-peer output

1#show run | sec dial-peer
dial-peer voice 1 pots
 trunkgroup FXO_EM
 translation-profile incoming Inbound
 call-block translation-profile incoming block_profile
 call-block disconnect-cause incoming call-reject
 incoming called-number .
 direct-inward-dial
dial-peer voice 911 pots
 trunkgroup FXO_EM
 description Services
 translation-profile outgoing Strip9
 destination-pattern 911
dial-peer voice 9911 pots
 trunkgroup FXO_EM
 description Services
 translation-profile outgoing Strip9
 destination-pattern 9911
dial-peer voice 2 voip
 description CUCM Dial-Peer - Inbound
 preference 1
 destination-pattern 12086858038<tel:(208)%20685-8038>
 session protocol sipv2
 session target ipv4:10.0.2.6
 incoming uri via 10
 voice-class codec 1
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 fax nsf 00
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 3 voip
 preference 2
 destination-pattern 12086858038<tel:(208)%20685-8038>
 

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

Thank you Anthony very much for your help.

The 19529473283 for  connection plar 19529473283 I was testing with. After 
changing of few things on the gateway, I'm now seeing different messages when I 
run debug. The calls used to ring only one time and then fast busy, now it 
rings twice and then fax busy for incoming. for outgoing, it's says calls 
connot be completed as dialed.


Attached the new logs again. I really appreciate your help.


thanks again.

Hamu



From: Anthony Holloway <avholloway+cisco-v...@gmail.com>
Sent: Thursday, March 1, 2018 11:05 AM
To: Hamu Ebiso
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

According to the debug, you're calling 19529473283.  Is that correct?


Called Number=19529473283(TON=Unknown, NPI=Unknown),

If so, then this debug is not for this gateway, because you're hitting Outgoing 
Dial-peer=5


Outgoing Dial-peer=5, Params=0x688FF3FC, Progress Indication=ORIGINATING SIDE 
IS NON ISDN(3)

And dial-peer 5, while it exists in your config you pasted, it doesn't match 
the called number.

dial-peer voice 5 voip
 description FXO test
 destination-pattern 12083437020
 session target ipv4:10.0.2.5
!

Even if it did match the destination pattern, you're missing some config on 
this dial-peer to make a SIP capable dial-peer, which means it's sending H323 
setup to CUCM.  If you built the gateway as H323, then I can see how it rang 
your CIPC, otherwise, if you built it as SIP, which you said you did, then it 
would never ring your CIPC.

So, something is not adding up here.

Also, you're hitting Incoming Dial-peer=0, which on a POTS leg is not too 
terrible, but it's also not good practice to be using dial-peer 0.  It's really 
VoIP DP 0 that is messy though.


Incoming Dial-peer=0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), 
Calling IE Present=FALSE,


On Thu, Mar 1, 2018 at 10:06 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Hi Anthony,


yes the call was hitting the gateway because I was seeing FXO port going 
Off-Hook. The problem was that Logging was not setup in the gateway. After I 
setup the Logging, I'm now seeing logs.


I have attached the log.

thanks

Hamu


From: avhollo...@gmail.com<mailto:avhollo...@gmail.com> 
<avhollo...@gmail.com<mailto:avhollo...@gmail.com>> on behalf of Anthony 
Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Thursday, March 1, 2018 8:45 AM
To: Hamu Ebiso
Cc: Cisco VoIP Group

Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
Since you didn't see the call in your debugs, your call likely didn't hit your 
gateway.

Can you confirm how you made your test call, when you had your debug running?

On Feb 28, 2018 2:46 PM, "Hamu Ebiso" 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Here is the dial-peer output

1#show run | sec dial-peer
dial-peer voice 1 pots
 trunkgroup FXO_EM
 translation-profile incoming Inbound
 call-block translation-profile incoming block_profile
 call-block disconnect-cause incoming call-reject
 incoming called-number .
 direct-inward-dial
dial-peer voice 911 pots
 trunkgroup FXO_EM
 description Services
 translation-profile outgoing Strip9
 destination-pattern 911
dial-peer voice 9911 pots
 trunkgroup FXO_EM
 description Services
 translation-profile outgoing Strip9
 destination-pattern 9911
dial-peer voice 2 voip
 description CUCM Dial-Peer - Inbound
 preference 1
 destination-pattern 12086858038<tel:(208)%20685-8038>
 session protocol sipv2
 session target ipv4:10.0.2.6
 incoming uri via 10
 voice-class codec 1
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 fax nsf 00
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 3 voip
 preference 2
 destination-pattern 12086858038<tel:(208)%20685-8038>
 session target ipv4:10.0.2.3
dial-peer voice 650 pots
 trunkgroup Local
 description Local outbound
 translation-profile outgoing Strip9
 destination-pattern 9208...
dial-peer voice 4 pots
 trunkgroup FXO_EM
 description ** 10-digit Local Call **
 translation-profile outgoing Strip9
 destination-pattern ^9[2-9]..[2-9]..$
 forward-digits 10
dial-peer voice 5 voip
 description FXO test
 destination-pattern 12083437020<tel:(208)%20343-7020>
 session target ipv4:10.0.2.5


thank you very much!!


From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 1:29 PM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

Maybe you're actually using MGCP then?  You still haven't shown the di

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

Hi Anthony,


yes the call was hitting the gateway because I was seeing FXO port going 
Off-Hook. The problem was that Logging was not setup in the gateway. After I 
setup the Logging, I'm now seeing logs.


I have attached the log.

thanks

Hamu


From: avhollo...@gmail.com <avhollo...@gmail.com> on behalf of Anthony Holloway 
<avholloway+cisco-v...@gmail.com>
Sent: Thursday, March 1, 2018 8:45 AM
To: Hamu Ebiso
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

Since you didn't see the call in your debugs, your call likely didn't hit your 
gateway.

Can you confirm how you made your test call, when you had your debug running?

On Feb 28, 2018 2:46 PM, "Hamu Ebiso" 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Here is the dial-peer output

1#show run | sec dial-peer
dial-peer voice 1 pots
 trunkgroup FXO_EM
 translation-profile incoming Inbound
 call-block translation-profile incoming block_profile
 call-block disconnect-cause incoming call-reject
 incoming called-number .
 direct-inward-dial
dial-peer voice 911 pots
 trunkgroup FXO_EM
 description Services
 translation-profile outgoing Strip9
 destination-pattern 911
dial-peer voice 9911 pots
 trunkgroup FXO_EM
 description Services
 translation-profile outgoing Strip9
 destination-pattern 9911
dial-peer voice 2 voip
 description CUCM Dial-Peer - Inbound
 preference 1
 destination-pattern 12086858038<tel:(208)%20685-8038>
 session protocol sipv2
 session target ipv4:10.0.2.6
 incoming uri via 10
 voice-class codec 1
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 fax nsf 00
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 3 voip
 preference 2
 destination-pattern 12086858038<tel:(208)%20685-8038>
 session target ipv4:10.0.2.3
dial-peer voice 650 pots
 trunkgroup Local
 description Local outbound
 translation-profile outgoing Strip9
 destination-pattern 9208...
dial-peer voice 4 pots
 trunkgroup FXO_EM
 description ** 10-digit Local Call **
 translation-profile outgoing Strip9
 destination-pattern ^9[2-9]..[2-9]..$
 forward-digits 10
dial-peer voice 5 voip
 description FXO test
 destination-pattern 12083437020<tel:(208)%20343-7020>
 session target ipv4:10.0.2.5


thank you very much!!


From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 1:29 PM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

Maybe you're actually using MGCP then?  You still haven't shown the dial-peers 
on the gateway.  Is that because you don't have any?

show run | section dial-peer

On Wed, Feb 28, 2018 at 12:19 PM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

debug was turned on as shown below.


1#debug voice ccapi inout
voip ccapi inout debugging is on

made test calls and then show log, I see below output.

1#show log
Syslog logging: enabled (0 messages dropped, 37 messages rate-limited, 0 
flushes, 0 overruns, xml disabled, filtering disabled)

No Active Message Discriminator.


No Inactive Message Discriminator.

Console logging: level debugging, 517 messages logged, xml disabled,
 filtering disabled
Monitor logging: level debugging, 0 messages logged, xml disabled,
 filtering disabled
Buffer logging:  level debugging, 550 messages logged, xml disabled,
filtering disabled
Exception Logging: size (4096 bytes)
Count and timestamp logging messages: disabled
Persistent logging: disabled

No active filter modules.

Trap logging: level debugging, 620 message lines logged
Logging to 10.0.7.139  (udp port 514, audit disabled,
  link up),
  435 message lines logged,
  0 message lines rate-limited,
  0 message lines dropped-by-MD,
  xml disabled, sequence number disabled
  filtering disabled
Logging to 10.0.8.12  (udp port 514, audit disabled,
  link up),
  435 message lines logged,
  0 message lines rate-limited,
  0 message lines dropped-by-MD,
  xml disabled, sequence number disabled
  filtering disabled
Logging Source-Interface:   VRF Name:
Loopback0


I don't see any calls activity.


thanks



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 11:19 AM

To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
But you

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

thank you very much for that info!! I checked under Preferences and optimize is 
not checked.



From: bmead...@gmail.com <bmead...@gmail.com> on behalf of Brian Meade 
<bmead...@vt.edu>
Sent: Wednesday, February 28, 2018 2:55 PM
To: Hamu Ebiso
Cc: Anthony Holloway; cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

Didn't read the whole thread but make sure CIPC doesn't have "Optimize for low 
bandwidth" checked in the preferences.  This tries to force G.729.

On Wed, Feb 28, 2018 at 3:52 PM, Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Here is the region information.


[cid:8f436bad-8c75-481a-999e-8abf58c113af]


thank you very much for your help.


From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 1:31 PM

To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

Regions are important for a working call flow, yes.  However, regions only 
exist on CUCM, not on gateways.

The way in which regions/codec selection could produce the issue your seeing is 
as follows:

Dial Peer has g711ulaw on it
CUCM SIP Trunk has Gateway-Region on it
CIPC has Phone-Region on it
Gateway-Region to Phone-Region is using 8kbps

On Wed, Feb 28, 2018 at 12:30 PM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Region is not define in the config but CUCM. Is that what you need? That is the 
only config I have. and don't see any region defined in the gateway.


thanks

hamu



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 11:19 AM

To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
But you didn't send all the information, because your output doesn't show the 
dial-peers.

One method is to capture the output from "debug voip ccapi inout" and look for 
the following line "Outgoing Dial-peer="

You also didn't address the regions either.  Can you send that too?

On Wed, Feb 28, 2018 at 11:13 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

I was just trying to give you all the information so that you will be able to 
see the whole picture.

Regarding your questions, I don't know how to check that.



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 10:45 AM

To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
That's interesting that I asked for the dial-peer config, and you sent 
everything but.

Do you know how to confirm which dial-peer the gateway is using?

On Wed, Feb 28, 2018 at 10:38 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Below is gateway configuration




isdn switch-type primary-ni
!
!
trunk group FXO_EM
 hunt-scheme sequential
 translation-profile incoming Incoming
!
!
trunk group Local
 hunt-scheme sequential
 translation-profile outgoing outbound
!
!
trunk group LD
 translation-profile outgoing outbound
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 10.170.99.12
  ipv4 10.44.50.14
  ipv4 10.44.50.39
 no ip address trusted authenticate
 clid substitute name
 clid network-provided
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 redirect ip2ip
 signaling forward unconditional
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback 
pass-through g711ulaw
 no fax-relay sg3-to-g3
 modem passthrough nse codec g711ulaw
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  session transport tcp
  min-se 360 session-expires 360
  ds0-num
  header-passing
  error-passthru
  registrar server expires max 600 min 60
  early-offer forced
  midcall-signaling passthru
  no call service stop
!
!
voice class uri 10 sip
 host ipv4:10.x.x.x
 host ipv4:10..x.x.x
 host ipv4:10..x.x.x
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
voice class codec 2
 codec preference 1 g711ulaw
!
!
!
!
voice translation-rule 1
 rule 1 /^208343\(\)$/ /1208343\1/
!
voice translation-rule 9
 rule 1 /5.../ /1208343\0/
!
voice translation-rule 10
 rule 1 /^911/ /911/
 rule 2 /^9

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

Here is the region information.


[cid:8f436bad-8c75-481a-999e-8abf58c113af]


thank you very much for your help.


From: Anthony Holloway <avholloway+cisco-v...@gmail.com>
Sent: Wednesday, February 28, 2018 1:31 PM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

Regions are important for a working call flow, yes.  However, regions only 
exist on CUCM, not on gateways.

The way in which regions/codec selection could produce the issue your seeing is 
as follows:

Dial Peer has g711ulaw on it
CUCM SIP Trunk has Gateway-Region on it
CIPC has Phone-Region on it
Gateway-Region to Phone-Region is using 8kbps

On Wed, Feb 28, 2018 at 12:30 PM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Region is not define in the config but CUCM. Is that what you need? That is the 
only config I have. and don't see any region defined in the gateway.


thanks

hamu



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 11:19 AM

To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
But you didn't send all the information, because your output doesn't show the 
dial-peers.

One method is to capture the output from "debug voip ccapi inout" and look for 
the following line "Outgoing Dial-peer="

You also didn't address the regions either.  Can you send that too?

On Wed, Feb 28, 2018 at 11:13 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

I was just trying to give you all the information so that you will be able to 
see the whole picture.

Regarding your questions, I don't know how to check that.



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 10:45 AM

To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
That's interesting that I asked for the dial-peer config, and you sent 
everything but.

Do you know how to confirm which dial-peer the gateway is using?

On Wed, Feb 28, 2018 at 10:38 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Below is gateway configuration




isdn switch-type primary-ni
!
!
trunk group FXO_EM
 hunt-scheme sequential
 translation-profile incoming Incoming
!
!
trunk group Local
 hunt-scheme sequential
 translation-profile outgoing outbound
!
!
trunk group LD
 translation-profile outgoing outbound
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 10.170.99.12
  ipv4 10.44.50.14
  ipv4 10.44.50.39
 no ip address trusted authenticate
 clid substitute name
 clid network-provided
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 redirect ip2ip
 signaling forward unconditional
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback 
pass-through g711ulaw
 no fax-relay sg3-to-g3
 modem passthrough nse codec g711ulaw
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  session transport tcp
  min-se 360 session-expires 360
  ds0-num
  header-passing
  error-passthru
  registrar server expires max 600 min 60
  early-offer forced
  midcall-signaling passthru
  no call service stop
!
!
voice class uri 10 sip
 host ipv4:10.x.x.x
 host ipv4:10..x.x.x
 host ipv4:10..x.x.x
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
voice class codec 2
 codec preference 1 g711ulaw
!
!
!
!
voice translation-rule 1
 rule 1 /^208343\(\)$/ /1208343\1/
!
voice translation-rule 9
 rule 1 /5.../ /1208343\0/
!
voice translation-rule 10
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
!
voice translation-rule 15
!
voice translation-rule 30
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
 rule 4 /^9\(011.*\)/ /\1/
 rule 9 /^9\(.*\)/ /\1/
!
!
voice translation-profile Add1
 translate calling 91
 translate called 1
!
voice translation-profile Inbound
 translate called 1
!
voice translation-profile Strip9
 translate called 10
!
voice translation-profile block_profile
 translate calling 15
!
voice translation-profile outbound
 translate calling 10
!
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO3845-MB sn FOC123128X5
license accept end user agreement
archive
 log config
  hidekeys
username admin privilege 15 password 7 105D1F0C
username svuntbd privilege 15 secret 5 $1$vRxN$nu4JPZttcuPskPHoSdQZx.
!
!
controller T1 0/0/0

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

Here is the dial-peer output

1#show run | sec dial-peer
dial-peer voice 1 pots
 trunkgroup FXO_EM
 translation-profile incoming Inbound
 call-block translation-profile incoming block_profile
 call-block disconnect-cause incoming call-reject
 incoming called-number .
 direct-inward-dial
dial-peer voice 911 pots
 trunkgroup FXO_EM
 description Services
 translation-profile outgoing Strip9
 destination-pattern 911
dial-peer voice 9911 pots
 trunkgroup FXO_EM
 description Services
 translation-profile outgoing Strip9
 destination-pattern 9911
dial-peer voice 2 voip
 description CUCM Dial-Peer - Inbound
 preference 1
 destination-pattern 12086858038
 session protocol sipv2
 session target ipv4:10.0.2.6
 incoming uri via 10
 voice-class codec 1
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 fax nsf 00
 ip qos dscp cs3 signaling
 no vad
dial-peer voice 3 voip
 preference 2
 destination-pattern 12086858038
 session target ipv4:10.0.2.3
dial-peer voice 650 pots
 trunkgroup Local
 description Local outbound
 translation-profile outgoing Strip9
 destination-pattern 9208...
dial-peer voice 4 pots
 trunkgroup FXO_EM
 description ** 10-digit Local Call **
 translation-profile outgoing Strip9
 destination-pattern ^9[2-9]..[2-9]..$
 forward-digits 10
dial-peer voice 5 voip
 description FXO test
 destination-pattern 12083437020
 session target ipv4:10.0.2.5


thank you very much!!


From: Anthony Holloway <avholloway+cisco-v...@gmail.com>
Sent: Wednesday, February 28, 2018 1:29 PM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

Maybe you're actually using MGCP then?  You still haven't shown the dial-peers 
on the gateway.  Is that because you don't have any?

show run | section dial-peer

On Wed, Feb 28, 2018 at 12:19 PM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

debug was turned on as shown below.


1#debug voice ccapi inout
voip ccapi inout debugging is on

made test calls and then show log, I see below output.

1#show log
Syslog logging: enabled (0 messages dropped, 37 messages rate-limited, 0 
flushes, 0 overruns, xml disabled, filtering disabled)

No Active Message Discriminator.


No Inactive Message Discriminator.

Console logging: level debugging, 517 messages logged, xml disabled,
 filtering disabled
Monitor logging: level debugging, 0 messages logged, xml disabled,
 filtering disabled
Buffer logging:  level debugging, 550 messages logged, xml disabled,
filtering disabled
Exception Logging: size (4096 bytes)
Count and timestamp logging messages: disabled
Persistent logging: disabled

No active filter modules.

Trap logging: level debugging, 620 message lines logged
Logging to 10.0.7.139  (udp port 514, audit disabled,
  link up),
  435 message lines logged,
  0 message lines rate-limited,
  0 message lines dropped-by-MD,
  xml disabled, sequence number disabled
  filtering disabled
Logging to 10.0.8.12  (udp port 514, audit disabled,
  link up),
  435 message lines logged,
  0 message lines rate-limited,
  0 message lines dropped-by-MD,
  xml disabled, sequence number disabled
  filtering disabled
Logging Source-Interface:   VRF Name:
Loopback0


I don't see any calls activity.


thanks



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 11:19 AM

To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
But you didn't send all the information, because your output doesn't show the 
dial-peers.

One method is to capture the output from "debug voip ccapi inout" and look for 
the following line "Outgoing Dial-peer="

You also didn't address the regions either.  Can you send that too?

On Wed, Feb 28, 2018 at 11:13 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

I was just trying to give you all the information so that you will be able to 
see the whole picture.

Regarding your questions, I don't know how to check that.



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 10:45 AM

To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
That's interesting that I asked for the dial-peer config, and you sent 
everything but.

Do you know 

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

Region is not define in the config but CUCM. Is that what you need? That is the 
only config I have. and don't see any region defined in the gateway.


thanks

hamu



From: Anthony Holloway <avholloway+cisco-v...@gmail.com>
Sent: Wednesday, February 28, 2018 11:19 AM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

But you didn't send all the information, because your output doesn't show the 
dial-peers.

One method is to capture the output from "debug voip ccapi inout" and look for 
the following line "Outgoing Dial-peer="

You also didn't address the regions either.  Can you send that too?

On Wed, Feb 28, 2018 at 11:13 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

I was just trying to give you all the information so that you will be able to 
see the whole picture.

Regarding your questions, I don't know how to check that.



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 10:45 AM

To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
That's interesting that I asked for the dial-peer config, and you sent 
everything but.

Do you know how to confirm which dial-peer the gateway is using?

On Wed, Feb 28, 2018 at 10:38 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Below is gateway configuration




isdn switch-type primary-ni
!
!
trunk group FXO_EM
 hunt-scheme sequential
 translation-profile incoming Incoming
!
!
trunk group Local
 hunt-scheme sequential
 translation-profile outgoing outbound
!
!
trunk group LD
 translation-profile outgoing outbound
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 10.170.99.12
  ipv4 10.44.50.14
  ipv4 10.44.50.39
 no ip address trusted authenticate
 clid substitute name
 clid network-provided
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 redirect ip2ip
 signaling forward unconditional
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback 
pass-through g711ulaw
 no fax-relay sg3-to-g3
 modem passthrough nse codec g711ulaw
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  session transport tcp
  min-se 360 session-expires 360
  ds0-num
  header-passing
  error-passthru
  registrar server expires max 600 min 60
  early-offer forced
  midcall-signaling passthru
  no call service stop
!
!
voice class uri 10 sip
 host ipv4:10.x.x.x
 host ipv4:10..x.x.x
 host ipv4:10..x.x.x
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
voice class codec 2
 codec preference 1 g711ulaw
!
!
!
!
voice translation-rule 1
 rule 1 /^208343\(\)$/ /1208343\1/
!
voice translation-rule 9
 rule 1 /5.../ /1208343\0/
!
voice translation-rule 10
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
!
voice translation-rule 15
!
voice translation-rule 30
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
 rule 4 /^9\(011.*\)/ /\1/
 rule 9 /^9\(.*\)/ /\1/
!
!
voice translation-profile Add1
 translate calling 91
 translate called 1
!
voice translation-profile Inbound
 translate called 1
!
voice translation-profile Strip9
 translate called 10
!
voice translation-profile block_profile
 translate calling 15
!
voice translation-profile outbound
 translate calling 10
!
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO3845-MB sn FOC123128X5
license accept end user agreement
archive
 log config
  hidekeys
username admin privilege 15 password 7 105D1F0C
username svuntbd privilege 15 secret 5 $1$vRxN$nu4JPZttcuPskPHoSdQZx.
!
!
controller T1 0/0/0
 shutdown
 cablelength long 0db
!
controller T1 0/1/0
 shutdown
 cablelength long 0db
!
!
!
!
!
!
interface Loopback0
 ip address 10.x.x.x 255.255.255.128
!
interface GigabitEthernet0/0
 description uplink to Lan1
 ip address 10.x.x.x 255.255.255.252
 duplex auto
 speed auto
 media-type rj45
!
interface GigabitEthernet0/1
 description uplink to Lan2
 ip address 10.x.x.x 255.255.255.252
 duplex auto
 speed auto
 media-type rj45
!
router ospf 1
 auto-cost reference-bandwidth 1
 network 10.0.0.0 0.0.255.255 area 10.0.0.0
!
ip forward-protocol nd
!
!
no ip http server
no ip http secure-server
!


___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

debug was turned on as shown below.


1#debug voice ccapi inout
voip ccapi inout debugging is on

made test calls and then show log, I see below output.

1#show log
Syslog logging: enabled (0 messages dropped, 37 messages rate-limited, 0 
flushes, 0 overruns, xml disabled, filtering disabled)

No Active Message Discriminator.


No Inactive Message Discriminator.

Console logging: level debugging, 517 messages logged, xml disabled,
 filtering disabled
Monitor logging: level debugging, 0 messages logged, xml disabled,
 filtering disabled
Buffer logging:  level debugging, 550 messages logged, xml disabled,
filtering disabled
Exception Logging: size (4096 bytes)
Count and timestamp logging messages: disabled
Persistent logging: disabled

No active filter modules.

Trap logging: level debugging, 620 message lines logged
Logging to 10.0.7.139  (udp port 514, audit disabled,
  link up),
  435 message lines logged,
  0 message lines rate-limited,
  0 message lines dropped-by-MD,
  xml disabled, sequence number disabled
  filtering disabled
Logging to 10.0.8.12  (udp port 514, audit disabled,
  link up),
  435 message lines logged,
  0 message lines rate-limited,
  0 message lines dropped-by-MD,
  xml disabled, sequence number disabled
  filtering disabled
Logging Source-Interface:   VRF Name:
Loopback0


I don't see any calls activity.


thanks



From: Anthony Holloway <avholloway+cisco-v...@gmail.com>
Sent: Wednesday, February 28, 2018 11:19 AM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

But you didn't send all the information, because your output doesn't show the 
dial-peers.

One method is to capture the output from "debug voip ccapi inout" and look for 
the following line "Outgoing Dial-peer="

You also didn't address the regions either.  Can you send that too?

On Wed, Feb 28, 2018 at 11:13 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

I was just trying to give you all the information so that you will be able to 
see the whole picture.

Regarding your questions, I don't know how to check that.



From: Anthony Holloway 
<avholloway+cisco-v...@gmail.com<mailto:avholloway%2bcisco-v...@gmail.com>>
Sent: Wednesday, February 28, 2018 10:45 AM

To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router
That's interesting that I asked for the dial-peer config, and you sent 
everything but.

Do you know how to confirm which dial-peer the gateway is using?

On Wed, Feb 28, 2018 at 10:38 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Below is gateway configuration




isdn switch-type primary-ni
!
!
trunk group FXO_EM
 hunt-scheme sequential
 translation-profile incoming Incoming
!
!
trunk group Local
 hunt-scheme sequential
 translation-profile outgoing outbound
!
!
trunk group LD
 translation-profile outgoing outbound
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 10.170.99.12
  ipv4 10.44.50.14
  ipv4 10.44.50.39
 no ip address trusted authenticate
 clid substitute name
 clid network-provided
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 redirect ip2ip
 signaling forward unconditional
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback 
pass-through g711ulaw
 no fax-relay sg3-to-g3
 modem passthrough nse codec g711ulaw
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  session transport tcp
  min-se 360 session-expires 360
  ds0-num
  header-passing
  error-passthru
  registrar server expires max 600 min 60
  early-offer forced
  midcall-signaling passthru
  no call service stop
!
!
voice class uri 10 sip
 host ipv4:10.x.x.x
 host ipv4:10..x.x.x
 host ipv4:10..x.x.x
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
voice class codec 2
 codec preference 1 g711ulaw
!
!
!
!
voice translation-rule 1
 rule 1 /^208343\(\)$/ /1208343\1/
!
voice translation-rule 9
 rule 1 /5.../ /1208343\0/
!
voice translation-rule 10
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
!
voice translation-rule 15
!
voice translation-rule 30
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
 rule 4 /^9\(011.*\)/ /\1/
 rule 9 /^9\(.*\)/ /\1/
!
!
voice translation-profile Add1
 translate calling 91
 translate called 1
!
voice trans

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

I was just trying to give you all the information so that you will be able to 
see the whole picture.

Regarding your questions, I don't know how to check that.



From: Anthony Holloway <avholloway+cisco-v...@gmail.com>
Sent: Wednesday, February 28, 2018 10:45 AM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help configuring router for 911 with POTS Lines 
connected to FXO port on the router

That's interesting that I asked for the dial-peer config, and you sent 
everything but.

Do you know how to confirm which dial-peer the gateway is using?

On Wed, Feb 28, 2018 at 10:38 AM Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Below is gateway configuration




isdn switch-type primary-ni
!
!
trunk group FXO_EM
 hunt-scheme sequential
 translation-profile incoming Incoming
!
!
trunk group Local
 hunt-scheme sequential
 translation-profile outgoing outbound
!
!
trunk group LD
 translation-profile outgoing outbound
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 10.170.99.12
  ipv4 10.44.50.14
  ipv4 10.44.50.39
 no ip address trusted authenticate
 clid substitute name
 clid network-provided
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 redirect ip2ip
 signaling forward unconditional
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback 
pass-through g711ulaw
 no fax-relay sg3-to-g3
 modem passthrough nse codec g711ulaw
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  session transport tcp
  min-se 360 session-expires 360
  ds0-num
  header-passing
  error-passthru
  registrar server expires max 600 min 60
  early-offer forced
  midcall-signaling passthru
  no call service stop
!
!
voice class uri 10 sip
 host ipv4:10.x.x.x
 host ipv4:10..x.x.x
 host ipv4:10..x.x.x
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
voice class codec 2
 codec preference 1 g711ulaw
!
!
!
!
voice translation-rule 1
 rule 1 /^208343\(\)$/ /1208343\1/
!
voice translation-rule 9
 rule 1 /5.../ /1208343\0/
!
voice translation-rule 10
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
!
voice translation-rule 15
!
voice translation-rule 30
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
 rule 4 /^9\(011.*\)/ /\1/
 rule 9 /^9\(.*\)/ /\1/
!
!
voice translation-profile Add1
 translate calling 91
 translate called 1
!
voice translation-profile Inbound
 translate called 1
!
voice translation-profile Strip9
 translate called 10
!
voice translation-profile block_profile
 translate calling 15
!
voice translation-profile outbound
 translate calling 10
!
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO3845-MB sn FOC123128X5
license accept end user agreement
archive
 log config
  hidekeys
username admin privilege 15 password 7 105D1F0C
username svuntbd privilege 15 secret 5 $1$vRxN$nu4JPZttcuPskPHoSdQZx.
!
!
controller T1 0/0/0
 shutdown
 cablelength long 0db
!
controller T1 0/1/0
 shutdown
 cablelength long 0db
!
!
!
!
!
!
interface Loopback0
 ip address 10.x.x.x 255.255.255.128
!
interface GigabitEthernet0/0
 description uplink to Lan1
 ip address 10.x.x.x 255.255.255.252
 duplex auto
 speed auto
 media-type rj45
!
interface GigabitEthernet0/1
 description uplink to Lan2
 ip address 10.x.x.x 255.255.255.252
 duplex auto
 speed auto
 media-type rj45
!
router ospf 1
 auto-cost reference-bandwidth 1
 network 10.0.0.0 0.0.255.255 area 10.0.0.0
!
ip forward-protocol nd
!
!
no ip http server
no ip http secure-server
!


___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

Below is gateway configuration




isdn switch-type primary-ni
!
!
trunk group FXO_EM
 hunt-scheme sequential
 translation-profile incoming Incoming
!
!
trunk group Local
 hunt-scheme sequential
 translation-profile outgoing outbound
!
!
trunk group LD
 translation-profile outgoing outbound
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 10.170.99.12
  ipv4 10.44.50.14
  ipv4 10.44.50.39
 no ip address trusted authenticate
 clid substitute name
 clid network-provided
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 redirect ip2ip
 signaling forward unconditional
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback 
pass-through g711ulaw
 no fax-relay sg3-to-g3
 modem passthrough nse codec g711ulaw
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  session transport tcp
  min-se 360 session-expires 360
  ds0-num
  header-passing
  error-passthru
  registrar server expires max 600 min 60
  early-offer forced
  midcall-signaling passthru
  no call service stop
!
!
voice class uri 10 sip
 host ipv4:10.x.x.x
 host ipv4:10..x.x.x
 host ipv4:10..x.x.x
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
voice class codec 2
 codec preference 1 g711ulaw
!
!
!
!
voice translation-rule 1
 rule 1 /^208343\(\)$/ /1208343\1/
!
voice translation-rule 9
 rule 1 /5.../ /1208343\0/
!
voice translation-rule 10
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
!
voice translation-rule 15
!
voice translation-rule 30
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
 rule 4 /^9\(011.*\)/ /\1/
 rule 9 /^9\(.*\)/ /\1/
!
!
voice translation-profile Add1
 translate calling 91
 translate called 1
!
voice translation-profile Inbound
 translate called 1
!
voice translation-profile Strip9
 translate called 10
!
voice translation-profile block_profile
 translate calling 15
!
voice translation-profile outbound
 translate calling 10
!
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO3845-MB sn FOC123128X5
license accept end user agreement
archive
 log config
  hidekeys
username admin privilege 15 password 7 105D1F0C
username svuntbd privilege 15 secret 5 $1$vRxN$nu4JPZttcuPskPHoSdQZx.
!
!
controller T1 0/0/0
 shutdown
 cablelength long 0db
!
controller T1 0/1/0
 shutdown
 cablelength long 0db
!
!
!
!
!
!
interface Loopback0
 ip address 10.x.x.x 255.255.255.128
!
interface GigabitEthernet0/0
 description uplink to Lan1
 ip address 10.x.x.x 255.255.255.252
 duplex auto
 speed auto
 media-type rj45
!
interface GigabitEthernet0/1
 description uplink to Lan2
 ip address 10.x.x.x 255.255.255.252
 duplex auto
 speed auto
 media-type rj45
!
router ospf 1
 auto-cost reference-bandwidth 1
 network 10.0.0.0 0.0.255.255 area 10.0.0.0
!
ip forward-protocol nd
!
!
no ip http server
no ip http secure-server
!


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[cisco-voip] Need help configuring router for 911 with POTS Lines connected to FXO port on the router

[Hamu Ebiso]

Hi Team,


I really appreciated if someone can help me with this configuration. Below is 
what I'm trying to achieve.


I have CUCM Version: 11.0

Voice router 3845 with 4FXO port module

2 POTS Line

Added 1 Pot line to Cisco IP communicator in CUCM

Router is connected to 110 block

connection between Gateway and CUCM SIP Trunk.



I have made configuration and put the router on the network, I can login fine. 
This configuration is for 911 call for remote location. I have added one of the 
pot line to CIPC. When I make inbound call to the pot line I have added to 
CIPC, I see FXO port going off-Hook but the call ring once and then fast busy. 
I have been trying to figure out for a while but I'm not able to figure out why 
that is happening. So I'm hope someone has done this before and knows how to 
make it work.


your help is greatly appreciated in advance.


thanks

Hamu

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[cisco-voip] High Availability for 3rd Party Spectra Link phones

[Hamu Ebiso]

Hi Team


We have clustered CUCM for geographically dispersed Distribution Center and 
retails. We have deployed Spectra link Phones to those distribution centers. We 
run into some issues and our servers which Spectra link registered to went down 
for days and customers at DC was not able to use Spectra link because of this 
outage and it was not good for Voice team including me.  Originally most of the 
Spectralink phones XML files were setup with only one SIP server. After this 
outage, I'm trying to add secondary server. I have added the secondary server 
to test but phones are not registering. I'm just wondering just if someone can 
share tested XML file for High Availability and let me know if I'm missing 
anything on my below XML file.

voIpProt.server.1.address="10.2.5.6"
voIpProt.server.2.address="10.2.5.7"
voIpProt.server.1.port="5060"
voIpProt.server.2.port="5060"
voIpProt.server.1.transport="TCPpreferred"
voIpProt.server.2.transport="TCPpreferred"
reg.1.server.1.address="10.2.5.6"
reg.2.server.2.address="10.2.5.7"
reg.1.server.1.port="5060"
reg.2.server.2.port="5060"
reg.1.server.1.transport="TCPpreferred"
reg.2.server.2.transport="TCPpreferred"


Thanks in Advance!!
Hamu
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Re: [cisco-voip] Need help with SIP FAX Configuration.

[Hamu Ebiso]

sorry  I was out on vocation!!


Yes the voice gateway is connected to service provider via ISDN.

yes I am expecting calls to come in from the service provider, match a 10 digit 
number and go to the FAX dial-peer? (ie not going to CUCM at all.

I'm using T.38
yes the fax is expecting TCP.
onsite contact is not available this week and will attach when the person comes 
back from vacation.

thanks


From: Dana Tong <dana.t...@yellit.com.au>
Sent: Wednesday, November 1, 2017 4:44 PM
To: Hamu Ebiso; Sreekanth
Cc: cisco-voip voyp list
Subject: RE: [cisco-voip] Need help with SIP FAX Configuration.


Can you explain the setup a bit further?



Is the Voice Gateway connecting to a service provider? SIP /ISDN?

Are you expecting calls to come in from the service provider, match a 10 digit 
number and go to the FAX dial-peer? (ie not going to CUCM at all?)

Are you using T.38/ G711?

TCP / UDP?  (Is the FAX expecting TCP which I think is the default dial-peer 
configuration).

Can you share the debug ccsip messages?





Cheers

Dana









From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Hamu 
Ebiso
Sent: Thursday, 2 November 2017 6:04 AM
To: Sreekanth <sknt...@gmail.com>
Cc: cisco-voip voyp list <cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Need help with SIP FAX Configuration.



I did setup Ricoh SIP-FAX as a Third-Party SIP Device per instruction but not 
registering to CUCM.



Thanks

hamu





From: Sreekanth <sknt...@gmail.com<mailto:sknt...@gmail.com>>
Sent: Wednesday, November 1, 2017 8:08 AM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help with SIP FAX Configuration.



Where is the gateway sending the INVITE? Is that going to the service provider 
or CUCM?

Here's a discussion on support forums regarding Ricoh SIP fax device.
https://supportforums.cisco.com/t5/ip-telephony/cucm-7-1-help-configuring-ricoh-ip-fax/td-p/1582086

CUCM 7.1 - Help Configuring Ricoh IP Fa... - Cisco Support 
...<https://supportforums.cisco.com/t5/ip-telephony/cucm-7-1-help-configuring-ricoh-ip-fax/td-p/1582086>

supportforums.cisco.com

Greetings! I have a number of Ricoh Multifunction Printers that support 
H.323/SIP IP-Fax capabilities. I would like to be able to use them with my CUCM 
7.1 and 2821 ...




As long as it sends a re-invite for voice call -> fax escalation, and responds 
to invites for fax escalation, the calls will work fine.



On 1 November 2017 at 17:51, Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

I did turn on debug and made inbound call and see that gateway is sending 3 
invite but not getting any response.



The other question is, has anyone configured any SIP FAX as 3rd party sip 
device in CUCM? Is that even capable.



thanks

Hamu





From: Sreekanth <sknt...@gmail.com<mailto:sknt...@gmail.com>>
Sent: Wednesday, November 1, 2017 3:01 AM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help with SIP FAX Configuration.



Check if the call is being answered first. The first step of a fax call is the 
voice call getting established, and then getting escalated to a fax call.

So usual troubleshooting for a voice call to see why the call is not answered.

The next step is to check what protocol of fax you are using and check if that 
is configured correctly end-to-end.



On 1 November 2017 at 02:11, Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Hi Everyone,



Has any body experiencing configuring RICOH MP C5503  for SIP FAX on Either 
CUCM as 3th party SIP or on voice gateway?



11.0 CUCM

SIP FAX: RICOH MP C5503

Voice router 4431.



I have configured Route Pattern in CUCM and pointed the router group and then 
to the voice gateway.

I configured Dial peer in voice router but when I dial the FAX is not picking 
up. When I try to dial out, nothing is happening as well.



Dial Peer as shown below:

dial-peer voice 500 voip
 description Ricoh 072
 destination-pattern 10 digit #
 session protocol sipv2
 session target ipv4:x.x.x.x
 voice-class codec 1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 501 voip
 description Ricoh 073
 destination-pattern 10 digit #
 session protocol sipv2
 session target ipv4:x.x.x.x
 voice-class codec 1
 dtmf-relay rtp-nte



I'm just wondering if I'm missing any configuration.



if you can point to the any document that I can follow to make sure everything 
is setup correctly, that would be appreciated.





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Re: [cisco-voip] Need help with SIP FAX Configuration.

[Hamu Ebiso]

I did setup Ricoh SIP-FAX as a Third-Party SIP Device per instruction but not 
registering to CUCM.


Thanks

hamu



From: Sreekanth <sknt...@gmail.com>
Sent: Wednesday, November 1, 2017 8:08 AM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help with SIP FAX Configuration.

Where is the gateway sending the INVITE? Is that going to the service provider 
or CUCM?

Here's a discussion on support forums regarding Ricoh SIP fax device.
https://supportforums.cisco.com/t5/ip-telephony/cucm-7-1-help-configuring-ricoh-ip-fax/td-p/1582086

CUCM 7.1 - Help Configuring Ricoh IP Fa... - Cisco Support 
...<https://supportforums.cisco.com/t5/ip-telephony/cucm-7-1-help-configuring-ricoh-ip-fax/td-p/1582086>
supportforums.cisco.com
Greetings! I have a number of Ricoh Multifunction Printers that support 
H.323/SIP IP-Fax capabilities. I would like to be able to use them with my CUCM 
7.1 and 2821 ...




As long as it sends a re-invite for voice call -> fax escalation, and responds 
to invites for fax escalation, the calls will work fine.

On 1 November 2017 at 17:51, Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

I did turn on debug and made inbound call and see that gateway is sending 3 
invite but not getting any response.


The other question is, has anyone configured any SIP FAX as 3rd party sip 
device in CUCM? Is that even capable.


thanks

Hamu



From: Sreekanth <sknt...@gmail.com<mailto:sknt...@gmail.com>>
Sent: Wednesday, November 1, 2017 3:01 AM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help with SIP FAX Configuration.

Check if the call is being answered first. The first step of a fax call is the 
voice call getting established, and then getting escalated to a fax call.
So usual troubleshooting for a voice call to see why the call is not answered.
The next step is to check what protocol of fax you are using and check if that 
is configured correctly end-to-end.

On 1 November 2017 at 02:11, Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Hi Everyone,


Has any body experiencing configuring RICOH MP C5503  for SIP FAX on Either 
CUCM as 3th party SIP or on voice gateway?


11.0 CUCM

SIP FAX: RICOH MP C5503

Voice router 4431.


I have configured Route Pattern in CUCM and pointed the router group and then 
to the voice gateway.

I configured Dial peer in voice router but when I dial the FAX is not picking 
up. When I try to dial out, nothing is happening as well.


Dial Peer as shown below:

dial-peer voice 500 voip
 description Ricoh 072
 destination-pattern 10 digit #
 session protocol sipv2
 session target ipv4:x.x.x.x
 voice-class codec 1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 501 voip
 description Ricoh 073
 destination-pattern 10 digit #
 session protocol sipv2
 session target ipv4:x.x.x.x
 voice-class codec 1
 dtmf-relay rtp-nte

I'm just wondering if I'm missing any configuration.


if you can point to the any document that I can follow to make sure everything 
is setup correctly, that would be appreciated.



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Re: [cisco-voip] Need help with SIP FAX Configuration.

[Hamu Ebiso]

yes he gateway is sending INVITE 3 times and then failing. It's going to 
Service provider. The only configurations is route pattern which is pointing to 
route group and the route group is pointing the voicegateway.


thanks

Hamu



From: Sreekanth <sknt...@gmail.com>
Sent: Wednesday, November 1, 2017 8:08 AM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help with SIP FAX Configuration.

Where is the gateway sending the INVITE? Is that going to the service provider 
or CUCM?

Here's a discussion on support forums regarding Ricoh SIP fax device.
https://supportforums.cisco.com/t5/ip-telephony/cucm-7-1-help-configuring-ricoh-ip-fax/td-p/1582086

CUCM 7.1 - Help Configuring Ricoh IP Fa... - Cisco Support 
...<https://supportforums.cisco.com/t5/ip-telephony/cucm-7-1-help-configuring-ricoh-ip-fax/td-p/1582086>
supportforums.cisco.com
Greetings! I have a number of Ricoh Multifunction Printers that support 
H.323/SIP IP-Fax capabilities. I would like to be able to use them with my CUCM 
7.1 and 2821 ...




As long as it sends a re-invite for voice call -> fax escalation, and responds 
to invites for fax escalation, the calls will work fine.

On 1 November 2017 at 17:51, Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

I did turn on debug and made inbound call and see that gateway is sending 3 
invite but not getting any response.


The other question is, has anyone configured any SIP FAX as 3rd party sip 
device in CUCM? Is that even capable.


thanks

Hamu



From: Sreekanth <sknt...@gmail.com<mailto:sknt...@gmail.com>>
Sent: Wednesday, November 1, 2017 3:01 AM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help with SIP FAX Configuration.

Check if the call is being answered first. The first step of a fax call is the 
voice call getting established, and then getting escalated to a fax call.
So usual troubleshooting for a voice call to see why the call is not answered.
The next step is to check what protocol of fax you are using and check if that 
is configured correctly end-to-end.

On 1 November 2017 at 02:11, Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Hi Everyone,


Has any body experiencing configuring RICOH MP C5503  for SIP FAX on Either 
CUCM as 3th party SIP or on voice gateway?


11.0 CUCM

SIP FAX: RICOH MP C5503

Voice router 4431.


I have configured Route Pattern in CUCM and pointed the router group and then 
to the voice gateway.

I configured Dial peer in voice router but when I dial the FAX is not picking 
up. When I try to dial out, nothing is happening as well.


Dial Peer as shown below:

dial-peer voice 500 voip
 description Ricoh 072
 destination-pattern 10 digit #
 session protocol sipv2
 session target ipv4:x.x.x.x
 voice-class codec 1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 501 voip
 description Ricoh 073
 destination-pattern 10 digit #
 session protocol sipv2
 session target ipv4:x.x.x.x
 voice-class codec 1
 dtmf-relay rtp-nte

I'm just wondering if I'm missing any configuration.


if you can point to the any document that I can follow to make sure everything 
is setup correctly, that would be appreciated.



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Re: [cisco-voip] Need help with SIP FAX Configuration.

[Hamu Ebiso]

I did turn on debug and made inbound call and see that gateway is sending 3 
invite but not getting any response.


The other question is, has anyone configured any SIP FAX as 3rd party sip 
device in CUCM? Is that even capable.


thanks

Hamu



From: Sreekanth <sknt...@gmail.com>
Sent: Wednesday, November 1, 2017 3:01 AM
To: Hamu Ebiso
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Need help with SIP FAX Configuration.

Check if the call is being answered first. The first step of a fax call is the 
voice call getting established, and then getting escalated to a fax call.
So usual troubleshooting for a voice call to see why the call is not answered.
The next step is to check what protocol of fax you are using and check if that 
is configured correctly end-to-end.

On 1 November 2017 at 02:11, Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Hi Everyone,


Has any body experiencing configuring RICOH MP C5503  for SIP FAX on Either 
CUCM as 3th party SIP or on voice gateway?


11.0 CUCM

SIP FAX: RICOH MP C5503

Voice router 4431.


I have configured Route Pattern in CUCM and pointed the router group and then 
to the voice gateway.

I configured Dial peer in voice router but when I dial the FAX is not picking 
up. When I try to dial out, nothing is happening as well.


Dial Peer as shown below:

dial-peer voice 500 voip
 description Ricoh 072
 destination-pattern 10 digit #
 session protocol sipv2
 session target ipv4:x.x.x.x
 voice-class codec 1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 501 voip
 description Ricoh 073
 destination-pattern 10 digit #
 session protocol sipv2
 session target ipv4:x.x.x.x
 voice-class codec 1
 dtmf-relay rtp-nte

I'm just wondering if I'm missing any configuration.


if you can point to the any document that I can follow to make sure everything 
is setup correctly, that would be appreciated.



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[cisco-voip] Need help with SIP FAX Configuration.

[Hamu Ebiso]

Hi Everyone,


Has any body experiencing configuring RICOH MP C5503  for SIP FAX on Either 
CUCM as 3th party SIP or on voice gateway?


11.0 CUCM

SIP FAX: RICOH MP C5503

Voice router 4431.


I have configured Route Pattern in CUCM and pointed the router group and then 
to the voice gateway.

I configured Dial peer in voice router but when I dial the FAX is not picking 
up. When I try to dial out, nothing is happening as well.


Dial Peer as shown below:

dial-peer voice 500 voip
 description Ricoh 072
 destination-pattern 10 digit #
 session protocol sipv2
 session target ipv4:x.x.x.x
 voice-class codec 1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 501 voip
 description Ricoh 073
 destination-pattern 10 digit #
 session protocol sipv2
 session target ipv4:x.x.x.x
 voice-class codec 1
 dtmf-relay rtp-nte

I'm just wondering if I'm missing any configuration.


if you can point to the any document that I can follow to make sure everything 
is setup correctly, that would be appreciated.


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Re: [cisco-voip] Need guidence to move Avaya users to cisco CUCM

[Hamu Ebiso]

Thank you very much Ryan!



Sent from my Sprint Samsung Galaxy S® 6.


 Original message 
From: "Ryan Ratliff (rratliff)" <rratl...@cisco.com>
Date: 8/6/17 8:47 AM (GMT-06:00)
To: Hamu Ebiso <hebiso2...@hotmail.com>
Cc: "Wykoff, Robert" <rwyk...@sentinel.com>, Matthew Loraditch 
<mloradi...@heliontechnologies.com>, cisco-voip voyp list 
<cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Need guidence to move Avaya users to cisco CUCM

Use a hunt pilot that points to the Unity ports or SIP trunk just like you do 
for voicemail, but use a routing rule in Unity to send the calls to the 
appropriate call handler.
Start with a doc similar to 
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/administration/guide/10xcucsagx/10xcucsag080.html
 (for your version) and go from there.

-Ryan

On Aug 4, 2017, at 1:51 PM, Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:


Hi Robert,

This kind of configuration is new to me and need guidance. I am just wondering 
how do I sent the call to unity for Schedule and Holiday?

thanks
Hamu


From: Wykoff, Robert <rwyk...@sentinel.com<mailto:rwyk...@sentinel.com>>
Sent: Friday, August 4, 2017 12:40 PM
To: Matthew Loraditch; Hamu Ebiso; cisco-voip voyp list
Subject: RE: Need guidence to move Avaya users to cisco CUCM

Correct send the calls to unity for holiday, and then us caller input to send 
the call to separate hunt groups in Call Manager based on skill sets should 
work.




Robert Wykoff
Western Region Team Lead, Collaboration, Security, R/S,  Support Services
CCIE Voice/Collaboration # 18774
Sentinel Technologies, Inc.
Single Number Reach:  1-480-897-5938
Email/Webex: rwyk...@sentinel.com<mailto:rwyk...@sentinel.com>

Customer Service:  1-800-860-8102<tel:+18008608102> (24x7x365)



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Loraditch
Sent: Friday, August 4, 2017 10:34 AM
To: Hamu Ebiso <hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>>; 
cisco-voip voyp list 
<cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Subject: Re: [cisco-voip] Need guidence to move Avaya users to cisco CUCM



Unity Connection has holiday scheduling.
Here is the 9.x guide, albeit nothing has really changed:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/9x/administration/guide/9xcucsagx/9xcucsag100.pdf
You can run your Hunt group through there via an auto attendant and probably 
get something close to what you want.





There are no skill levels in CUCM, that is what contact center is for.  The 
closest you will get is longest idle in your line group settings.



Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
Network Engineer
Direct Voice: 443.541.1518
Facebook<https://www.facebook.com/heliontech?ref=hl> | 
Twitter<https://twitter.com/HelionTech> | 
LinkedIn<https://www.linkedin.com/company/helion-technologies?trk=top_nav_home> 
| G+<https://plus.google.com/+Heliontechnologies/posts>



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Hamu 
Ebiso
Sent: Friday, August 4, 2017 1:23 PM
To: cisco-voip voyp list 
<cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Need guidence to move Avaya users to cisco CUCM



I will be moving payroll group from Avaya phone system to Cisco CUCM. These 
users are accustomed to features that is not available in CUCM. I am just 
wondering if you can help how to replicate what those user have in Avaya to 
Cisco CUCM.



They have Holiday recording in Avaya and wondering how to replicate this 
without contact Center.




If calls dialed to group main number, it will go to the most idle Agent based 
on defined skill Level





Any help is greatly appreciated.



Thank you very much in advance.



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Re: [cisco-voip] Need guidence to move Avaya users to cisco CUCM

[Hamu Ebiso]

Hi Robert,


This kind of configuration is new to me and need guidance. I am just wondering 
how do I sent the call to unity for Schedule and Holiday?


thanks

Hamu



From: Wykoff, Robert <rwyk...@sentinel.com>
Sent: Friday, August 4, 2017 12:40 PM
To: Matthew Loraditch; Hamu Ebiso; cisco-voip voyp list
Subject: RE: Need guidence to move Avaya users to cisco CUCM


Correct send the calls to unity for holiday, and then us caller input to send 
the call to separate hunt groups in Call Manager based on skill sets should 
work.



[CCIE_Collaboration_75px]

Robert Wykoff

Western Region Team Lead, Collaboration, Security, R/S,  Support Services

CCIE Voice/Collaboration # 18774

Sentinel Technologies, Inc.

Single Number Reach:  1-480-897-5938

Email/Webex: rwyk...@sentinel.com<mailto:rwyk...@sentinel.com>



Customer Service:  1-800-860-8102<tel:+18008608102> (24x7x365)



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Matthew Loraditch
Sent: Friday, August 4, 2017 10:34 AM
To: Hamu Ebiso <hebiso2...@hotmail.com>; cisco-voip voyp list 
<cisco-voip@puck.nether.net>
Subject: Re: [cisco-voip] Need guidence to move Avaya users to cisco CUCM



Unity Connection has holiday scheduling.

Here is the 9.x guide, albeit nothing has really changed:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/9x/administration/guide/9xcucsagx/9xcucsag100.pdf

You can run your Hunt group through there via an auto attendant and probably 
get something close to what you want.





There are no skill levels in CUCM, that is what contact center is for.  The 
closest you will get is longest idle in your line group settings.



Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
Network Engineer
Direct Voice: 443.541.1518

Facebook<https://www.facebook.com/heliontech?ref=hl> | 
Twitter<https://twitter.com/HelionTech> | 
LinkedIn<https://www.linkedin.com/company/helion-technologies?trk=top_nav_home> 
| G+<https://plus.google.com/+Heliontechnologies/posts>



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Hamu 
Ebiso
Sent: Friday, August 4, 2017 1:23 PM
To: cisco-voip voyp list 
<cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>>
Subject: [cisco-voip] Need guidence to move Avaya users to cisco CUCM



I will be moving payroll group from Avaya phone system to Cisco CUCM. These 
users are accustomed to features that is not available in CUCM. I am just 
wondering if you can help how to replicate what those user have in Avaya to 
Cisco CUCM.



They have Holiday recording in Avaya and wondering how to replicate this 
without contact Center.

[cid:image002.png@01D30D0E.066C4B20]



If calls dialed to group main number, it will go to the most idle Agent based 
on defined skill Level

[cid:image003.png@01D30D0E.066C4B20]

[cid:image004.png@01D30D0E.066C4B20]



Any help is greatly appreciated.



Thank you very much in advance.


___
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Re: [cisco-voip] Need guidence to move Avaya users to cisco CUCM

[Hamu Ebiso]

Thank you Very much Matt for your imput. I know there will not be any Skill 
group CUCM but wondering guidance how configure other functionality in CUCM and 
unity server.


thanks

Hamu



From: Matthew Loraditch <mloradi...@heliontechnologies.com>
Sent: Friday, August 4, 2017 12:33 PM
To: Hamu Ebiso; cisco-voip voyp list
Subject: RE: Need guidence to move Avaya users to cisco CUCM


Unity Connection has holiday scheduling.

Here is the 9.x guide, albeit nothing has really changed:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/9x/administration/guide/9xcucsagx/9xcucsag100.pdf

Managing Schedules and Holidays in Cisco Unity Connection 
9<http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/9x/administration/guide/9xcucsagx/9xcucsag100.pdf>
www.cisco.com
11-2 System Administration Guide for Cisco Unity Connection Release 9.x Chapter 
11 Managing Schedules and Holidays in Cisco Unity Connection 9.x



You can run your Hunt group through there via an auto attendant and probably 
get something close to what you want.





There are no skill levels in CUCM, that is what contact center is for.  The 
closest you will get is longest idle in your line group settings.



Matthew G. Loraditch – CCNP-Voice, CCNA-R, CCDA
Network Engineer
Direct Voice: 443.541.1518


Facebook<https://www.facebook.com/heliontech?ref=hl> | 
Twitter<https://twitter.com/HelionTech> | 
LinkedIn<https://www.linkedin.com/company/helion-technologies?trk=top_nav_home> 
| G+<https://plus.google.com/+Heliontechnologies/posts>



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Hamu 
Ebiso
Sent: Friday, August 4, 2017 1:23 PM
To: cisco-voip voyp list <cisco-voip@puck.nether.net>
Subject: [cisco-voip] Need guidence to move Avaya users to cisco CUCM



I will be moving payroll group from Avaya phone system to Cisco CUCM. These 
users are accustomed to features that is not available in CUCM. I am just 
wondering if you can help how to replicate what those user have in Avaya to 
Cisco CUCM.



They have Holiday recording in Avaya and wondering how to replicate this 
without contact Center.

[cid:image002.png@01D30D25.CD016C90]



If calls dialed to group main number, it will go to the most idle Agent based 
on defined skill Level

[cid:image003.png@01D30D25.CD016C90]

[cid:image004.png@01D30D25.CD016C90]



Any help is greatly appreciated.



Thank you very much in advance.


___
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[cisco-voip] Need guidence to move Avaya users to cisco CUCM

[Hamu Ebiso]

I will be moving payroll group from Avaya phone system to Cisco CUCM. These 
users are accustomed to features that is not available in CUCM. I am just 
wondering if you can help how to replicate what those user have in Avaya to 
Cisco CUCM.


They have Holiday recording in Avaya and wondering how to replicate this 
without contact Center.

[cid:d8bd4874-f66c-4482-87e3-c0eb0afe0bc3]


If calls dialed to group main number, it will go to the most idle Agent based 
on defined skill Level

[cid:a3ab799b-a91c-41bf-b36b-d5acdc89ae0b]

[cid:8ba9c414-c300-45d3-8514-580e8530bc4f]


Any help is greatly appreciated.


Thank you very much in advance.

___
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

Re: [cisco-voip] Configure 3845 router for 911 calls.

[Hamu Ebiso]

Hi Dave,


The document you have provided is great for SIP Gateway config. I am just 
wondering if there is any document for MGCP Gateway as well?

Thanks



From: cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Dave Goodwin 
<dave.good...@december.net>
Sent: Thursday, July 27, 2017 9:24 PM
To: Scott Voll
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Configure 3845 router for 911 calls.

There are several ways to accomplish this on the UCM side. I would consider 
creating a RG for Site A GW and a RG for Site B GW, have one 911 route pattern 
accessible via the main CSS that is routed to a RL with Standard Local Route 
Group. Then use the Device Pool configurations for Site A and Site B to assign 
the Local Route Group. In any case, all of the UCM configurations are valid and 
meet the stated goals.

What I think the OP is asking for is help configuring the actual IOS GW with 
FXO ports to route the calls. Hamu, configuration options for an IOS voice 
gateway are quite extensive. Everyone has their opinion on which of the 
available options are "best," and I'm not going to try to convince you which 
approach (H.323, MGCP, SIP protocols) is best. I would say you might consider 
calling Cisco TAC and asking for help with this very basic initial 
configuration of a voice gateway... but I doubt you'll get help on your 3845 
since that model's end of support date has passed. Instead, I'll just give you 
a link I saw from a simple Google search that will get you started down one of 
those roads (SIP). Hopefully this will give you what you need to modify it for 
your specific scenario and get it working. Good luck! 
http://ucpros.net/cisco-sip-gateway-configuration/

Cisco SIP Gateway configuration: The Ultimate Guide - 
UCPros<http://ucpros.net/cisco-sip-gateway-configuration/>
ucpros.net
SIP is Cisco's recommended protocol for Voice Gateway & CUCM interconnection. 
This is the most comprehensive guide for Cisco SIP Gateway configuration.



On Thu, Jul 27, 2017 at 6:29 PM, Scott Voll 
<svoll.v...@gmail.com<mailto:svoll.v...@gmail.com>> wrote:
I agree with Neal,

but I would use one partition for each site for 911, then a third partition for 
everything else, then, put the correct 911 partition in the site CSS and use 
the third partition in both CSS.  Save a lot of time recreating RP for the 
other stuff Guess it depends on how many RP you have. we have hundreds.

Scott


On Thu, Jul 27, 2017 at 3:13 PM, Haas, Neal 
<nh...@co.fresno.ca.us<mailto:nh...@co.fresno.ca.us>> wrote:
Everything is done in the CUCM.

Pots are configured with MGCP.

Create New Partition. (site Name)
Assign 911 Route for site A to Site Partition 1
Assign 911 Route for Site B to Site Partition 1
Assign all other Routes to both Partitions.

Only 911 calls are treated different per site – if you do not take phones 
between sites, you won’t have a problem. Two POTS lines per site is correct.

I don’t think I missed anything, it’s not hard to accomplish the separate 911 
calls between PSAPS.

Neal

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>]
 On Behalf Of Hamu Ebiso
Sent: Thursday, July 27, 2017 2:40 PM
To: Norton, Mike <mikenor...@pwsd76.ab.ca<mailto:mikenor...@pwsd76.ab.ca>>;
Subject: Re: [cisco-voip] Configure 3845 router for 911 calls.


Let me rephrase the questions with more understanding update after I talk to 
Mike and few other people.



We have 2 locations. one is main building were everything reside and the second 
 building is remote building shared with someone else.

About 50 people reside at  the secondary remote location. This people use main 
locations CUCM over the WAN for calling. Remote location is about 25 minutes 
from main building.



So if these people at remote location call 911, emergency responder will see 
main building location not remote location where the call is residing.

Therefore we are tasked to put together plan to give users option to be able to 
call 911 and show their true locations.



We are forced to find spare 3845 router to configure for this users for 911 
call. below is what we have.

ordered 2 POT Lines,

have 3845 router

We have both FOX and FXS modules for POT Lines.

Amphenol Cable to connect to FOX and 110 block.

CUCM version: 11.0



The reason we ordered 2 POT lines instead of circuit is because they are cheap. 
Circuit cost about $500 a month while POT line cost about $40 a line. Since 
this router is only used for 911 call we don't want to spend that much money.



The question, has any one configured router for 911 and have basic config that 
I can use for my config. Since I haven't done voice gateway configuration 
before, I just need help with this.



Any help is greatly appreciated.



Thanks

Hamu


From: Norton, Mike <mikenor...@pwsd76.ab.ca<mai

Re: [cisco-voip] Configure 3845 router for 911 calls.

[Hamu Ebiso]

Thank you very much all for your great suggestions. I really appreciated.


Thanks



From: cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Dave Goodwin 
<dave.good...@december.net>
Sent: Thursday, July 27, 2017 9:24 PM
To: Scott Voll
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] Configure 3845 router for 911 calls.

There are several ways to accomplish this on the UCM side. I would consider 
creating a RG for Site A GW and a RG for Site B GW, have one 911 route pattern 
accessible via the main CSS that is routed to a RL with Standard Local Route 
Group. Then use the Device Pool configurations for Site A and Site B to assign 
the Local Route Group. In any case, all of the UCM configurations are valid and 
meet the stated goals.

What I think the OP is asking for is help configuring the actual IOS GW with 
FXO ports to route the calls. Hamu, configuration options for an IOS voice 
gateway are quite extensive. Everyone has their opinion on which of the 
available options are "best," and I'm not going to try to convince you which 
approach (H.323, MGCP, SIP protocols) is best. I would say you might consider 
calling Cisco TAC and asking for help with this very basic initial 
configuration of a voice gateway... but I doubt you'll get help on your 3845 
since that model's end of support date has passed. Instead, I'll just give you 
a link I saw from a simple Google search that will get you started down one of 
those roads (SIP). Hopefully this will give you what you need to modify it for 
your specific scenario and get it working. Good luck! 
http://ucpros.net/cisco-sip-gateway-configuration/

Cisco SIP Gateway configuration: The Ultimate Guide - 
UCPros<http://ucpros.net/cisco-sip-gateway-configuration/>
ucpros.net
SIP is Cisco's recommended protocol for Voice Gateway & CUCM interconnection. 
This is the most comprehensive guide for Cisco SIP Gateway configuration.



On Thu, Jul 27, 2017 at 6:29 PM, Scott Voll 
<svoll.v...@gmail.com<mailto:svoll.v...@gmail.com>> wrote:
I agree with Neal,

but I would use one partition for each site for 911, then a third partition for 
everything else, then, put the correct 911 partition in the site CSS and use 
the third partition in both CSS.  Save a lot of time recreating RP for the 
other stuff Guess it depends on how many RP you have. we have hundreds.

Scott


On Thu, Jul 27, 2017 at 3:13 PM, Haas, Neal 
<nh...@co.fresno.ca.us<mailto:nh...@co.fresno.ca.us>> wrote:
Everything is done in the CUCM.

Pots are configured with MGCP.

Create New Partition. (site Name)
Assign 911 Route for site A to Site Partition 1
Assign 911 Route for Site B to Site Partition 1
Assign all other Routes to both Partitions.

Only 911 calls are treated different per site – if you do not take phones 
between sites, you won’t have a problem. Two POTS lines per site is correct.

I don’t think I missed anything, it’s not hard to accomplish the separate 911 
calls between PSAPS.

Neal

From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net<mailto:cisco-voip-boun...@puck.nether.net>]
 On Behalf Of Hamu Ebiso
Sent: Thursday, July 27, 2017 2:40 PM
To: Norton, Mike <mikenor...@pwsd76.ab.ca<mailto:mikenor...@pwsd76.ab.ca>>;
Subject: Re: [cisco-voip] Configure 3845 router for 911 calls.


Let me rephrase the questions with more understanding update after I talk to 
Mike and few other people.



We have 2 locations. one is main building were everything reside and the second 
 building is remote building shared with someone else.

About 50 people reside at  the secondary remote location. This people use main 
locations CUCM over the WAN for calling. Remote location is about 25 minutes 
from main building.



So if these people at remote location call 911, emergency responder will see 
main building location not remote location where the call is residing.

Therefore we are tasked to put together plan to give users option to be able to 
call 911 and show their true locations.



We are forced to find spare 3845 router to configure for this users for 911 
call. below is what we have.

ordered 2 POT Lines,

have 3845 router

We have both FOX and FXS modules for POT Lines.

Amphenol Cable to connect to FOX and 110 block.

CUCM version: 11.0



The reason we ordered 2 POT lines instead of circuit is because they are cheap. 
Circuit cost about $500 a month while POT line cost about $40 a line. Since 
this router is only used for 911 call we don't want to spend that much money.



The question, has any one configured router for 911 and have basic config that 
I can use for my config. Since I haven't done voice gateway configuration 
before, I just need help with this.



Any help is greatly appreciated.



Thanks

Hamu


From: Norton, Mike <mikenor...@pwsd76.ab.ca<mailto:mikenor...@pwsd76.ab.ca>>
Sent: Monday, July 24, 2017 3:58 

Re: [cisco-voip] Configure 3845 router for 911 calls.

[Hamu Ebiso]

Let me rephrase the questions with more understanding update after I talk to 
Mike and few other people.


We have 2 locations. one is main building were everything reside and the second 
 building is remote building shared with someone else.

About 50 people reside at  the secondary remote location. This people use main 
locations CUCM over the WAN for calling. Remote location is about 25 minutes 
from main building.


So if these people at remote location call 911, emergency responder will see 
main building location not remote location where the call is residing.

Therefore we are tasked to put together plan to give users option to be able to 
call 911 and show their true locations.


We are forced to find spare 3845 router to configure for this users for 911 
call. below is what we have.

ordered 2 POT Lines,

have 3845 router

We have both FOX and FXS modules for POT Lines.

Amphenol Cable to connect to FOX and 110 block.

CUCM version: 11.0


The reason we ordered 2 POT lines instead of circuit is because they are cheap. 
Circuit cost about $500 a month while POT line cost about $40 a line. Since 
this router is only used for 911 call we don't want to spend that much money.


The question, has any one configured router for 911 and have basic config that 
I can use for my config. Since I haven't done voice gateway configuration 
before, I just need help with this.


Any help is greatly appreciated.


Thanks

Hamu



From: Norton, Mike <mikenor...@pwsd76.ab.ca>
Sent: Monday, July 24, 2017 3:58 PM
To: Hamu Ebiso
Subject: RE: Configure 3845 router for 911 call.


For the IP phones to be usable during WAN outage, you need something they can 
register to. E.g. SRST or CME on the router, or else a CUCM server on-site.



The config you posted shows that this router already has T1 lines. You can send 
911 calls to T1 lines, I don’t understand why you want to add analog POTS lines 
if you already have T1s.



If you want more suggestions then perhaps take this conversation back on to the 
mailing list, others besides me might be able to give you ideas.



-mn



From: Hamu Ebiso [mailto:hebiso2...@hotmail.com]
Sent: July 24, 2017 2:45 PM
To: Norton, Mike <mikenor...@pwsd76.ab.ca>
Subject: Re: Configure 3845 router for 911 call.



Thank you Mike for following up!!



I Just clarified with person who gave me that project to put it together. That 
person was not really familiar with voice staff just mentioned analog phones. 
There is no analog phones here.  These users have 7942 IP phones and the idea 
was to user POT lines to call out to 911 if WAN connect down at the site. The 
idea was, since these users are in some else's building, we need to find out 
how to make them call out to 911 if WAN connection is down. Since I am not 100% 
sure of how we can achieve this. I am ok with your suggestion how to achieve 
this.



ordered 2 POT Lines,

have 3845 router

We have both FOX and FXS modules

Amphenol Cable.



thanks

Hamu





From: Norton, Mike <mikenor...@pwsd76.ab.ca<mailto:mikenor...@pwsd76.ab.ca>>
Sent: Monday, July 24, 2017 3:26 PM
To: Hamu Ebiso
Subject: RE: Configure 3845 router for 911 call.



Hi Hamu, I don’t fully understand your call flow. POTS lines would normally 
connect to an FXO card. In that case, the easiest way to handle incoming calls 
is to do “connection plar ” on the voice-port, where  is the extension 
number where you want to send the calls. But since you mention analog phones 
then it is not clear to me exactly what you are trying to do.



Maybe take the conversation back to the mail list and give more explanation of 
the call flow, might be able to get more help that way.



-mn



From: Hamu Ebiso [mailto:hebiso2...@hotmail.com]
Sent: July 21, 2017 5:49 PM
To: Norton, Mike <mikenor...@pwsd76.ab.ca<mailto:mikenor...@pwsd76.ab.ca>>
Subject: Re: Configure 3845 router for 911 call.





Thank you very much Mike. I really appreciate your help.

As I mentioned it before, These employees are residing in someone else's 
building and using CUCM across WAN. The reason behind trying to use POT Lines 
is, If WAN is done these employee will be able to use POT Lines from Analog 
phones to call out.



One question is, if you can share command to setting how to send 911 calls to 
reception or security desk.



I really appreciate your input and advise.



Thanks

Hamu



From: Norton, Mike <mikenor...@pwsd76.ab.ca<mailto:mikenor...@pwsd76.ab.ca>>
Sent: Friday, July 21, 2017 5:43 PM
To: Hamu Ebiso; cisco-voip voyp list
Subject: RE: Configure 3845 router for 911 call.



Currently your config will send the 911 calls out the “Local” T1. Once you have 
the POTS lines added, you will need to put your POTS ports into their own trunk 
group and direct your outgoing dial-peers to the new trunk group instead of 
“Local.”



You should do something w

Re: [cisco-voip] IM License.

[Hamu Ebiso]

I knew the answer because when every I add IM users my license was not 
increasing. I have suspected that was the case but I have asked my cisco Rep 
but he told me I need license. That is the reseason I sent this email out for 
clarification.


thanks

hamu




From: Ben Amick <bam...@humanarc.com>
Sent: Thursday, July 20, 2017 9:48 AM
To: Hamu Ebiso; cisco-voip@puck.nether.net
Subject: RE: IM License.


If you roll them out as IM-only, they don’t consume licenses as you don’t 
configure CSFs. Just enable the user for presence and have them log into 
jabber. They will use the jabber-config of your global config, and you won’t be 
able to set group setups.



If they need phone capabilities, then you need CSFs, which if they are 
configured by themselves, consume an Enhanced license at minimum. If you 
configure them with any other device, they will need an enhanced plus, any more 
than that needs a UWL



From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Hamu 
Ebiso
Sent: Thursday, July 20, 2017 10:45 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] IM License.



Hello Everyone,



I need help understanding how IM license work. We have about 3700 license all 
together and most of the people use CSF and some of them use IP communicator 
and hard phone. We havr about 700 available right now.



Our company just made major acquisition with over 1000 employee. And we are 
tasked to role out IM to these users.



My question is has any one done this before and do we need node licence for IM 
availability and functionality? Since we will be roling out IM to more than 
1000 employee, do we need additional 1000 lisence for IM?



We are using 10.5 CUCM and will be moving to 11.0 very soon.



ANY HELP IS GREATLY APPRECIATED.



THANKS

HAMU







Sent from my Sprint Samsung Galaxy S® 6.

Confidentiality Note: This message is intended for use only by 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 
the reader of this message is not the intended recipient or the employee or 
agent responsible for delivering the message to the intended recipient, you are 
hereby notified that any dissemination, distribution or copying of this 
communication is strictly prohibited. If you have received this communication 
in error, please contact the sender immediately and destroy the material in its 
entirety, whether electronic or hard copy. Thank you
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[cisco-voip] Configure 3845 router for 911 call.

[Hamu Ebiso]

Hi Everyone,


I need help with this configuration for 911 only.


back ground history, our company bought another company few years back. After 
about 6 months, the deal fail apart. Within that 6 months employees moved 
around to different building mixed. When separation started, leaders negotiated 
so that we still provide professional service to the company that is separating 
from us for 3 years. In that 2 years, all the assets and employees need to be 
separated. There are about 50 employee of ours located in the building of the 
company that is in the process of separating from us.  We are in the process of 
separated them from this company system but in the same building. We had done 
all separation form phones to email server and etc. while in the process of 
separating this employee we have run into issue with 911 if the site loose WAN 
connection. I was tasked to figure out this. I am just wondering if anyone has 
done this and help me.


below is the router I am try to configure.

3845 with EVM-HD-8FXS/DID module

ordered Amphenol cable to connect to 110 block

ordered 2 POT Lines


I need help configuring this router for 911 call. I have attached the config I 
have and I am just wondering if I am missing anything. I haven't don't this 
kind of configuration before and I need guidance.


Thank you very much!!


q7005voicegw01#sh inv
NAME: "3845 chassis", DESCR: "3845 chassis"
PID: CISCO3845 , VID: V01 , SN: FTX1234A0J6

NAME: "c3845 Motherboard with Gigabit Ethernet on Slot 0", DESCR: "c3845 
Motherboard with Gigabit Ethernet"
PID: CISCO3845-MB  , VID: V06 , SN: FOC123128V4

NAME: "VWIC2-1MFT-T1/E1 - 1-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 
SubSlot 0", DESCR: "VWIC2-1MFT-T1/E1 - 1-Port RJ-48 Multiflex Trunk - T1/E1"
PID: VWIC2-1MFT-T1/E1  , VID: V01 , SN: FOC142038K1

NAME: "VWIC2-1MFT-T1/E1 - 1-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 
SubSlot 1", DESCR: "VWIC2-1MFT-T1/E1 - 1-Port RJ-48 Multiflex Trunk - T1/E1"
PID: VWIC2-1MFT-T1/E1  , VID: V01 , SN: FOC142039DQ

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 4", DESCR: "PVDMII DSP 
SIMM with four DSPs"
PID: PVDM2-64  , VID: V01 , SN: FOC122929P9

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 5", DESCR: "PVDMII DSP 
SIMM with four DSPs"
PID: PVDM2-64  , VID: V01 , SN: FOC122929GS

 EVM-HD-8FXS/DID will be installed on this router.
  
q7005voicegw01#sh run
Building configuration...


Current configuration : 4421 bytes
!
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname q7007voipgw1
!
boot-start-marker
boot-end-marker
!
!
card type t1 0 0
card type t1 0 1
!
no aaa new-model
clock timezone MNT -7 0
clock summer-time MDT recurring
network-clock-participate wic 0 
network-clock-participate wic 1 
network-clock-select 1 T1 0/0/0
! 
dot11 syslog
ip source-route
!
ip cef
!
!
!
!
ip domain name 
ip name-server x.x.x.x
ip name-server x.x.x.x
no ipv6 cef
multilink bundle-name authenticated
!
!
!
isdn switch-type primary-ni
!
!
trunk group Local
 hunt-scheme sequential
 translation-profile outgoing outbound
! 
!
trunk group LD
 translation-profile outgoing outbound
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice rtp send-recv
!
voice service voip
 no ip address trusted authenticate
 allow-connections sip to sip
 sip
  midcall-signaling passthru
!
voice class codec 1
 codec preference 1 g711ulaw
!
!
!
! 
voice translation-rule 1
 rule 1 /\(.*\)/ /1\1/
!
voice translation-rule 10
 rule 1 /^1\(.*\)/ /\1/
!
voice translation-rule 15
!
voice translation-rule 30
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
 rule 4 /^9\(011.*\)/ /\1/
 rule 9 /^9\(.*\)/ /\1/
!
!
voice translation-profile Inbound
 translate called 1
!
voice translation-profile Strip9
 translate called 30
!
voice translation-profile block_profile
 translate calling 15
!
voice translation-profile outbound
 translate calling 10
!
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO3845-MB sn FOC123128V4
license accept end user agreement
archive
 log config
  hidekeys
!
!
controller T1 0/0/0
 cablelength long 0db
 pri-group timeslots 1-24
!
controller T1 0/1/0
 cablelength long 0db
 pri-group timeslots 1-24
!
ip ssh version 2
!
!
!
!
!
interface Loopback0
 ip address 1.x.x.x 255.255.255.252
 no shutdown
!
interface GigabitEthernet0/0
 ip address x.x.x.x 255.255.255.252
 no shutdown
 duplex auto
 speed auto
 media-type rj45
!
interface GigabitEthernet0/1
 ip address x.x.x.x 255.255.255.252
 no shutdown
 duplex auto
 speed auto
 media-type rj45
!
interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 trunk-group Local
 no cdp enable
!
interface Serial0/1/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 trunk-group LD
 no cdp enable
!
ip forward-protocol nd
!
!
no ip http server
no ip http 

[cisco-voip] IM License.

[Hamu Ebiso]

Hello Everyone,

I need help understanding how IM license work. We have about 3700 license all 
together and most of the people use CSF and some of them use IP communicator 
and hard phone. We havr about 700 available right now.

Our company just made major acquisition with over 1000 employee. And we are 
tasked to role out IM to these users.

My question is has any one done this before and do we need node licence for IM 
availability and functionality? Since we will be roling out IM to more than 
1000 employee, do we need additional 1000 lisence for IM?

We are using 10.5 CUCM and will be moving to 11.0 very soon.

ANY HELP IS GREATLY APPRECIATED.

THANKS
HAMU



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Re: [cisco-voip] Bogen paging system with TAMB2 and 310 VG.

[Hamu Ebiso]

Yes we are using Bogens and in order to make VG work with Bogens, we used 
TAMB2. The questions were there are already 3 Zones and the customers are 
looking for the ways to Tie all 3 Zones to all Zones PAGING ZONE. I was just 
wondering if someone has ever done that. Using Bogens, TAMB2 and VG 310.

Thanks
Hamu



Sent from my Sprint Samsung Galaxy S® 6.


 Original message 
From: "Norton, Mike" <mikenor...@pwsd76.ab.ca>
Date: 6/12/17 12:01 PM (GMT-06:00)
To: Hamu Ebiso <hebiso2...@hotmail.com>, cisco-voip@puck.nether.net
Subject: RE: Bogen paging system with TAMB2 and 310 VG.

The TAMB doesn’t do zoning – one zone only. If you need zones, then take a look 
at the Bogen PCM2000 system instead of the TAMB.

-mn


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Hamu 
Ebiso
Sent: June 8, 2017 1:22 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Bogen paging system with TAMB2 and 310 VG.

Hi Everyone,

One of our customer was using Avaya phone system with Bogens paging system. We 
have converted them to cisco systems. Qe have ordered TAMB2 in order to make 
the their paging system work qith VG 310. They have about 4 Zones and all zones 
are working fine. Now the customer came back and asking how to create new zone 
and connect to all other existing zones. I am just wondering if someone has 
this kind of setup and can share them with me how to set them up.

Thanks
Hamu




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[cisco-voip] Bogen paging system with TAMB2 and 310 VG.

[Hamu Ebiso]

Hi Everyone,

One of our customer was using Avaya phone system with Bogens paging system. We 
have converted them to cisco systems. Qe have ordered TAMB2 in order to make 
the their paging system work qith VG 310. They have about 4 Zones and all zones 
are working fine. Now the customer came back and asking how to create new zone 
and connect to all other existing zones. I am just wondering if someone has 
this kind of setup and can share them with me how to set them up.

Thanks
Hamu




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[cisco-voip] Paging port on VG-310 doesn't go off-hook

[Hamu Ebiso]

Hi Everyone,

Has anyone seen this issue before and know how to fix it. We are Using Bogen's 
paging system with TAMB2 and some of the port on VG doesn't go off-hook when 
dialed. When you run show voce port summ, it shows On-hook ringing. Any help 
will be appreciated.

Thanks
Hamu



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[cisco-voip] CISCO SOFTKEY TEMPLATE CONFIGURATION.

[Hamu Ebiso]

Hi Everyone,

We have converted Avaya phone system to cisco phone system. TI Manager at the 
site asking me if this functionality are possible in call manager. If he is 
already connected to call can he br able to use pickup softkey to pickup group 
callm

Example: on a call and call comes into call pickup group.
1) A phone ring in his pickup group
2) He Pick up the handset
3) The pickup softkey vanishes
4) H can't pick up the call
5) He must reset yhe handset
6) Press pickup softkey
7) Wait for the speaker phone
8) pick up handset.

Thanks
Hamu


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Re: [cisco-voip] BOGEN PAGING SYSTEM.

[Hamu Ebiso]

I have a router with 4 FXO port and i am thinking about bypassing VG and 
directly connecting to Bogan Paging system however, i haven't done this kind of 
configuration and that is the reason i am asking.

I am just wondering if someone has a working config.

Thanks
Hamu



Sent from my Sprint Samsung Galaxy S® 6.


 Original message 
From: James O'Neill <jone...@pasadenaisd.org>
Date: 5/23/17 1:11 PM (GMT-06:00)
To: Hamu Ebiso <hebiso2...@hotmail.com>, cisco-voip@puck.nether.net
Subject: RE: BOGEN PAGING SYSTEM.

We use our SRST routers with FXO cards as the gateway for our PAs.  We’ve used 
them successful with Bogan in the past using the module that had the POTS 
connection.  We’ve also used them successfully with Valcom PA equipment.

I’ve found that if you have a POTS/Analog connection and know the DTMF codes 
you can use an FXO port as the gateway on most PA vendors.  Some are a little 
tricky and might require a port created by punching down to a 66 block.

James O'Neill

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Hamu 
Ebiso
Sent: Tuesday, May 23, 2017 11:48 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] BOGEN PAGING SYSTEM.

Hi all,

Has anyone know how to connect bogen paging system yo VG for paging.

We are converting avaya phone system to Cisco and the customer have old paging 
system wich is bogen. We ordered VG for their analog and I don't know we havr 
to move from yhat system to VG. If we have to connect directly from bouce 
gateway or some other way to connect to VG.

Thanks
Hamu



Sent from my Sprint Samsung Galaxy S® 6.






This email and any files transmitted with it are confidential and intended 
solely for the use of the individual or entity to whom they are addressed. If 
you have received this email in error please notify the system manager. Please 
note that any views or opinions presented in this email are solely those of the 
author and do not necessarily represent those of Pasadena I.S.D. Finally, the 
recipient should check this email and any attachments for the presence of 
viruses. The company accepts no liability for any damage caused by any virus 
transmitted by this email.

 Pasadena ISD is an equal opportunity employer and provides equal access to all 
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Re: [cisco-voip] Cisco 7942 phone Echoing.

[Hamu Ebiso]

Thank you very much for your imput Ryan.



Sent from my Sprint Samsung Galaxy S® 6.


 Original message 
From: Ryan Huff <ryanh...@outlook.com>
Date: 5/15/17 8:21 PM (GMT-06:00)
To: Hamu Ebiso <hebiso2...@hotmail.com>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Cisco 7942 phone Echoing.

I'm assuming you're dealing with TDM/analog technologies here. More often than 
not you're up against something called, "Talker's Echo" in which the local 
talker is hearing their voice duplicated back on their receiver. Many times, 
this comes from too much power coming from the local gateway and the excess 
signal is reflected back at the gateway from the PSTN.

If the the far-end is hearing an echo, it is generally due to equipment at the 
far end (nothing you can usually control).

To try and reduce local end echo / signal duplication; go into the 
configuration of the voice port paired with the interface for the voice card 
(Ex. voice-port 0/0/1:23) and reduce the input gain (Ex. input gain -3).

Thanks,

Ryan

On May 15, 2017, at 9:05 PM, Ryan Huff 
<ryanh...@outlook.com<mailto:ryanh...@outlook.com>> wrote:

Let me ask you this; when a user talks, are they hearing their own voice echoed 
back, OR is the far end (called party) hearing the talker's (calling party) 
voice twice in a way that sounds like an echo?

Thanks,

Ryan

On May 15, 2017, at 9:00 PM, Hamu Ebiso 
<hebiso2...@hotmail.com<mailto:hebiso2...@hotmail.com>> wrote:

Hi everyone,

I am just wondering if someone could help me with this!!

We have converted Avaya phone system to cisco phone system 4 weeks ago. 
Everything was working fine as it should be for a while but now hearing issues 
with echoing a lot of times.
The site is using SIP Trank
CUCM 10.5.2




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[cisco-voip] Cisco 8821 wireless phone.

[Hamu Ebiso]

Hi Everyone,

I am having issue with 8821 cisco phone. I am testing one of the phone to send 
to warehouse. While testing, i am getting one-way audio, call disconnecting. If 
call disconnect,  the phone doesn't allow me to make another call for about a 
minute or so. I am just wondering id someone experienced this issue before what 
was the fix. I am at office environment were there is many Aps but haveing 
issue when roaming.
If anyone has best practice Wlan  configuration documents on from wireless 
stand point, that will be appreciated as well.

Thanks
Hamu



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[cisco-voip] Cisco 7942 phone Echoing.

[Hamu Ebiso]

Hi everyone,

I am just wondering if someone could help me with this!!

We have converted Avaya phone system to cisco phone system 4 weeks ago. 
Everything was working fine as it should be for a while but now hearing issues 
with echoing a lot of times.
The site is using SIP Trank
CUCM 10.5.2




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[cisco-voip] CISCO PHONES BAT Template.

[Hamu Ebiso]

Does anyone has a good phone BATTING Tamplate? I am converting Avaya phone 
system to cisco phone system and having issue getting good tamplate. CUCM 
VERSION: 10.5.2.
PHONES MODEL, 7942's, 881's, 8851's and some wireless phones.

Anything with good instructions will be appreciate.

Thanks
Hamu



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[cisco-voip] Spartralink wireless phone one way audio.

[Hamu Ebiso]

Hi Everyone,

I work for the retails base company that utilizes spatralink wireless phones 
and Vertical phone system for the stores. I started with this company 3 weeks 
ago as a Voice System Engineer and I don't know anything about spartralink 
phones and Vertical phone system. I am all about cisco system. Some stores are 
moving to cisco systems but all lot of stores use vertical phone system because 
it's easy to trouble shoot and can be installed locally at the stores. 
Everything else is cisco, from switches to routers to ASA, and Unified 
communications.
Enough with the background and let me back to my points.

For over a year or so, most stores were having one way audio with spartralink 
phones. It doesn't matter if you are roaming between AP's or standing one 
place, you will have one-way audio and gargling voice at some point. Many 
people from vertical, cisco and spartralink were involved with troubleshooting 
but never able to resolve the issue since one-way audio is static and people at 
the stores were lazy and not following up or responding to asked questions with 
argent and they only react to when issue get worst.

Now i am tasked to spearhead the team and bring everyone together to resolve 
this issue for once and all.

I am just wondering if anybody had any interaction with these systems and can 
suggest anything that might help in resolving this issue.

Thanks
Hamu Ebiso



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Re: [cisco-voip] Significant delay with Jabber 11.8 voicemail widget

[Hamu Ebiso]

Sent from my Sprint Samsung Galaxy S® 6.

 Original message 
From: Lelio Fulgenzi 
Date: 12/21/16 4:13 PM (GMT-06:00)
To: cisco-voip voyp list 
Subject: [cisco-voip] Significant delay with Jabber 11.8 voicemail widget


I've noticed a significant delay in the voicemail widget in jabber 11.8 on 
iPhone and iPad. Haven't compared with android yet.

I'm referring to how the selected voicemail will expand to present a triangle 
play button. And shrink when another is selected, with the newly selected 
message expanding with the same triangle play button.

Anyone see the same thing?

Sent from my iPhone
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Re: [cisco-voip] Documenting a Cisco ICM Script

[Hamu Ebiso]

What is really amazing offer. I really  need that.
Thank you


Sent from my Verizon Wireless 4G LTE smartphone

 Original message 
From: Tanner Ezell tanner.ez...@gmail.com
Date: 03/18/2015  12:09 PM  (GMT-06:00)
To: Matthew Loraditch mloradi...@heliontechnologies.com
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Documenting a Cisco ICM Script

For anyone interested shoot me an email off-list for access to a private
beta (NDA required) coming up!

On Wed, Mar 18, 2015 at 10:05 AM, Matthew Loraditch 
mloradi...@heliontechnologies.com wrote:

  That would be amazing…



 Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA
 Network Engineer
 Direct Voice: 443.541.1518

  Facebook https://www.facebook.com/heliontech?ref=hl | Twitter
 https://twitter.com/HelionTech | LinkedIn
 https://www.linkedin.com/company/helion-technologies?trk=top_nav_home |
 G+ https://plus.google.com/+Heliontechnologies/posts



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Anthony Holloway
 *Sent:* Wednesday, March 18, 2015 1:02 PM
 *To:* Tanner Ezell; Walenta, Philip
 *Cc:* cisco-voip@puck.nether.net

 *Subject:* Re: [cisco-voip] Documenting a Cisco ICM Script



 Uh, hell yes Tanner!  Got a sample to share?



 On Wed, Mar 18, 2015 at 11:53 AM Tanner Ezell tanner.ez...@gmail.com
 wrote:

  Just curious to poke the hornets nest here.



 Would anyone be interested in such a solution for UCCX scripts? (I'm
 talking about Visio file generation [including full step configuration
 information], not screen shots of Steps..)



 On Wed, Mar 18, 2015 at 8:35 AM, Walenta, Philip 
 philip.wale...@polycom.com wrote:

  Years ago when I was doing a bunch of ICM I begged Cisco for a JSON or
 XML export or something for the scripts.



 Screenshots is still the only method of which I am aware.



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Terry Oakley
 *Sent:* Wednesday, March 18, 2015 10:33 AM
 *To:* Ryan Burtch; cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] Documenting a Cisco ICM Script



 I echo that.. trying to document those scripts has been a challenge to say
 the least.



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net
 cisco-voip-boun...@puck.nether.net] *On Behalf Of *Ryan Burtch
 *Sent:* March 18, 2015 9:19 AM
 *To:* cisco-voip@puck.nether.net
 *Subject:* [cisco-voip] Documenting a Cisco ICM Script



 *All*:



 Does anyone know of a good way to document a Cisco ICM script?



 All I do today is take a boat load of screen shots and add comments to
 them.



 Just wanted to know if anyone has a better system.










 Sincerely,



 Ryan Burtch


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