[cisco-voip] Deploy IP communicator in a software package

[Aaron Banks]

I'm curious to know if anyone has deployed IP communicator en masse?  If so how 
did you deploy it to users who *likely* did not have admin rights on their 
machines?  How were the user settings configured? (user or administrator?)

Many thanks

Aaron
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[cisco-voip] UCCX 11 - Finesse Agent forced ready

[Aaron Banks]

I was wondering if there is a way for finesse to default to a ready state as 
soon as an agent logs in?  This would eliminate the possibility of an agent 
forgetting to put themselves into a ready state.


Aaron
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Re: [cisco-voip] Unusual configuration

[Aaron Banks]

Hey Ryan


===Jabber===


No ccsip messages at the gatway - ccapi inout.  At least that was the debug 
file sent to me.  Your assumption is correct - a jabber user is dialing from 
inside of the network to outside.


Check on the bearer capabilities.  That is set.


===Gateway===


The device is not configured in CUCM as an H323 gateway.  It is a SIP trunk 
pointing to the 2901 gateway if that makes sense.  This is where I said on the 
gateway itself it is configured as H323, not MGCP.  There are no SIP trunks 
involved at all.   The only transport is the 23 channel PRI.  This is where I 
was confused with how this configuration should be working.  If there are no 
carrier SIP trunks, why would there be a SIP trunk configured to an H323  
gateway (i.e. H323 statements in the router config, no MGCP)?


I think your list of questions confirmed to me that this device should have 
been configured as an H323 gateway in CUCM, not a SIP trunk pointing to a 
gateway.  You are correct there is no H323 integration.


Thanks for responding so quickly.  At least I know  my line of thinking isn't 
out of line.


Aaron



From: Ryan Huff <ryanh...@outlook.com>
Sent: November 1, 2016 2:18 AM
To: Aaron Banks; cisco-voip@puck.nether.net
Subject: Re: Unusual configuration


Aaron,


== Jabber ==


You mention, "intermittent jabber problems  high number of SIP register 
events ... call is not successful".


Where are you seeing this, I'm guessing here but, ccsip messages in the gateway 
debugs? When the Jabber client is attempting to make a call, I'm assuming your 
Jabber client is registered to CCM on the inside of the network and users are 
trying to dial out for an audio call? Based on your description, it sounds like 
the Jabber client is presenting a video codec to a PSTN carrier on the other 
end of your PRI and the carrier is dropping the call like its hot (as PSTN 
carriers will).


If the above is what you're facing here; under the T1 voice port (voice-port 
x/x/x:23) set the bearer capabilities to speech (bearer-cap speech). You also 
mention that the gateway is configured as H.323; on the H.323 gateway device 
configuration page, ensure that "Retry Video Call As Audio" is checked. Next, 
verify the regional relationship between the Jabber clients and the H.323 
gateway is not allowing a session bit rate for video calls (or immersive). 
Lastly, ensure all CCM egress paths for the Jabber client egress through the 
H.323 gateway and not any SIP trunks pointed at the gateway.


== Gateway ==


I'm a little confused here. In the second sentence you state the gateway is 
configured as H.323 however in the last sentence you state that you would have 
expected an MGCP / H.323 integration with CUCM Vs. a SIP integration; which 
leads me to believe there is no H.323 integration currently?


Does the gateway peer with SIP service at all, or is it simply just a PRI/T1? 
If in-fact the gateway only has a PRI/T1 I would integrate that as an H.323 
gateway into CUCM (with all the appropriate dial peers and bindings on the 
gateway) and verify that all your CCM egress goes to the H.323 gateway and not 
the CCM SIP trunk pointed at the gateway.


Thanks,


-Ryan


From: cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Aaron Banks 
<amichaelba...@hotmail.com>
Sent: Monday, October 31, 2016 7:30 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Unusual configuration


I am supporting a customer with 2901 GW (PRI) and CUCM 11.0(2).  The setup: SIP 
trunk from CUCM to the 2901 GW, GW is configured as H323 with a full 23 channel 
PRI.  I have never seen this kind of set up before.  Anyone on the list ever do 
(or see) this setup before?  Did you encounter any issues?  I see intermittent 
jabber problems where a high number of SIP register events occur and  the call 
is not successful.  I would have thought the gateway would have been configured 
as either MGCP or H323 in CUCM.


Aaron
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[cisco-voip] Unusual configuration

[Aaron Banks]

I am supporting a customer with 2901 GW (PRI) and CUCM 11.0(2).  The setup: SIP 
trunk from CUCM to the 2901 GW, GW is configured as H323 with a full 23 channel 
PRI.  I have never seen this kind of set up before.  Anyone on the list ever do 
(or see) this setup before?  Did you encounter any issues?  I see intermittent 
jabber problems where a high number of SIP register events occur and  the call 
is not successful.  I would have thought the gateway would have been configured 
as either MGCP or H323 in CUCM.


Aaron
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[cisco-voip] Nuance/UCCX 10.6

[Aaron Banks]

Has anyone ever used Nuance to deliver speech to text messages to an agent?  
The case is, a caller leaves a voicemail message, and then that message is 
delivered to an agent.  I have never heard of this before and not even sure how 
it would work given that the voicemail message would reside in unity.

Thanks

Aaron
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[cisco-voip] Unity/Google Email

[Aaron Banks]

Has anyone ever set up Unity with google mail for voicemail to email?  Is an 
internal server required to relay messages or can google apps be used?  I have 
done exchange but google mail, no.  Unity version is 10.5.2.  I didn't think 
google apps could be used because SMTP uses port 465, not 25.


Thanks


Aaron



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Re: [cisco-voip] About Device Pool (?zg?r ??ALAN)

[Aaron Banks]

You can assign the device pool that you want to the phone itself on the phone 
configuration page.



From: cisco-voip  on behalf of 
cisco-voip-requ...@puck.nether.net 
Sent: July 2, 2016 9:00 AM
To: cisco-voip@puck.nether.net
Subject: cisco-voip Digest, Vol 153, Issue 4

Send cisco-voip mailing list submissions to
cisco-voip@puck.nether.net

To subscribe or unsubscribe via the World Wide Web, visit
https://puck.nether.net/mailman/listinfo/cisco-voip
cisco-voip Info Page - 
puck.nether.net
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A list for people interested in using Cisco VoIP equipment in a variety of 
environments, including service provider, enterprise and over the public 
internet.

  10. About Device Pool (?zg?r ??ALAN)

--

Message: 10
Date: Sat, 2 Jul 2016 17:42:31 +0300
From: ?zg?r ??ALAN 
To: 
Subject: [cisco-voip] About Device Pool
Message-ID:



Content-Type: text/plain; charset="iso-8859-9"

Hi,



I have a CUCM cluster with 2 Device Pools (DP). I want that some ip phones
automatically  join the DP-A, and the others automatically join the DP-B.
How can i do that?



Best Regards

Ozgur




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[cisco-voip] UCCX Script redirect

[Aaron Banks]

I'm having a little trouble with a script redirect.  The redirect works fine, 
but the call is being sent to a PRI instead of a SIP trunk.  How it works is 
the call comes in over the SIP trunk to a RP and then is sent to UCCX script 
with a couple of options.  Selecting either option sends the call to a DID 
outside of the enterprise.  I have put in a route pattern that is an exact 
match to one of the options selected in the IVR.  My question is - do I have a 
pattern matching problem or is the CSS of the CTI port being used (which, if 
true, I know exactly what the problem is).  I had the redirect CSS on the 
trigger set to be the route point, but that clearly is not working.


Any comments, suggestions, mild criticism appreciated.


Aaron
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[cisco-voip] VG320 sample configuration

[Aaron Banks]

Does anyone have a VG320 configuration they wouldn't mind sharing?  I have the 
basic configuration down but I was curious about a couple of new commands that 
configures dial peers and ports in a dial peer group.  Has anyone tried that?

How I have it configured (it's for a group of analog phones):
dial-peer group 1 pots
service stcapp
port all

That's how it is shown in the documentation but sometimes it doesn't quite work 
[as the documentation indicates].

Aaron
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Re: [cisco-voip] Looking for beta testers for UCCX Call Flow Designer

[Aaron Banks]

I would love to try out that software!


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Re: [cisco-voip] DTMF digits not hitting Voice Gateway

[Aaron Banks]

Yes, G.711 and both software and hardware resources.

From: ryanh...@outlook.com
To: amichaelba...@hotmail.com
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] DTMF digits not hitting Voice Gateway
Date: Wed, 2 Mar 2016 23:25:02 +






Do these phones have access to MTP resources and if so, software resources or 
hardware resources?



Also, I assume this is G.711, correct?




Thanks,



Ryan


On Mar 2, 2016, at 4:40 PM, Aaron Banks <amichaelba...@hotmail.com> wrote:










This problem has been resolved but I thought I'd put it out to the group for 
feedback, suggestions or general discussion.



One the odd occasion, this customer (who has 8861,8841,7821 SIP phones) calls a 
business with an IVR and the DTMF digits don't work.  It's odd (and 
infrequent).  The caller waits after pressing a digit and eventually the call 
drops.



My solution is to reset the SIP trunk between CUCM (v10.5.2) and CUBE.  In all 
instances, the issue is resolved.  On the other side of the CUBE is service 
provider SIP trunks.



My question to the group is do any of you have the same issue with SIP phones?  
If so, what do you do?



Aaron





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[cisco-voip] DTMF digits not hitting Voice Gateway

[Aaron Banks]

This problem has been resolved but I thought I'd put it out to the group for 
feedback, suggestions or general discussion.

One the odd occasion, this customer (who has 8861,8841,7821 SIP phones) calls a 
business with an IVR and the DTMF digits don't work.  It's odd (and 
infrequent).  The caller waits after pressing a digit and eventually the call 
drops.

My solution is to reset the SIP trunk between CUCM (v10.5.2) and CUBE.  In all 
instances, the issue is resolved.  On the other side of the CUBE is service 
provider SIP trunks.

My question to the group is do any of you have the same issue with SIP phones?  
If so, what do you do?

Aaron
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Re: [cisco-voip] UCCX + CUIC 10.6: Understanding Permissions

[Aaron Banks]

Hey Anthony,

Same here!  What worked in previous versions no longer works.  The only way I 
have been able to overcome the permissions issue is to log in as the 
administrator who creates the object (whether it be a folder, dashboard or 
report) and assign the permission.   If I log in as me, save a report as 
something else, I can't even assign permissions to my group to execute or 
write, it has to be the admin. :(

Totally bizarre and yes more complicated than it needs to be.

Hope that helps.  Ping me if I can assist further.

Aaron

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[cisco-voip] Informacast

[Aaron Banks]

I've never configured Informacast before.  I have most of the setup complete, 
the only piece I'm hung up on is the 3850 switch (which is stacked).  I want to 
enable multicast but it is only for the voice vlan.  There is no routing being 
done between vlans or multiple switches.  Do I use ip multicast routing to 
enable multicasting globally or ip multicast auto-enable?  Also, is PIM or IGMP 
normally used?  IGMP is already running, so I was thinking I only had to 
configure the multicast address for it.

Aaron



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[cisco-voip] Finesse time-out period

[Aaron Banks]

Does anyone know where the timeout setting is for finesse?  I have agents who 
sit in the not ready state for 10+ minutes and when they go to change back to 
ready, they see a communication error (something like "cannot communicate with 
server").  If I hit the refresh button on the browser, they have to log back 
in.  Maybe this is normal behavior.

Thanks - Aaron
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[cisco-voip] External calls from Jabber

[Aaron Banks]

inShare



 

   


  



The jabber questions never end.  I am struggling 
with Jabber.  Users can call each other internally, check voicemail and 
IM each other, no problem.  When they go to make an external call to any 
number, they hear "your call cannot be completed as dialed" which would 
tell me maybe a calling search space issue.  Not that.  I've changed the
 jabber client to start calls with audio instead of video. I've checked 
regions and there are only defaults.  Would the jabber-config.xml file 
create issues?  I would think it might bugger up directory services but 
not actually calling.
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Re: [cisco-voip] External calls from Jabber

[Aaron Banks]

Per Charles' suggestion, I used the DNA tool to analyze the dialed call from 
jabber.  Nothing is being blocked.  Digits are not being dialed by clicking on 
the contact, the dial pad is being used in jabber or entered in the calling 
window.
Date: Thu, 5 Nov 2015 12:01:13 -0600
Subject: Re: [cisco-voip] External calls from Jabber
From: avholloway+cisco-v...@gmail.com
To: amichaelba...@hotmail.com
CC: cisco-voip@puck.nether.net

It's pretty straight forward, if you already have a grasp on CUCM core 
functionality.
Your reported error message sounds like the Annunciator telling you there were 
no matches in the dial plan for what you dialed.  You didn't mention what was 
dialed, so we cannot help you there.  If however, the error message was more 
like fast busy or ringback then call drop, I would say it was a failure to 
establish bi-directional media.  That's not that case though, so I would focus 
on your dialing habits and your matched patterns.
Also, Cisco Jabber does not use en bloc dialing, dispite the fact there is no 
dial tone.  Seems a bit backwards, but that's how it is.
On Thu, Nov 5, 2015 at 11:10 AM, Aaron Banks <amichaelba...@hotmail.com> wrote:



























inShare



 

   


  



The jabber questions never end.  I am struggling 
with Jabber.  Users can call each other internally, check voicemail and 
IM each other, no problem.  When they go to make an external call to any 
number, they hear "your call cannot be completed as dialed" which would 
tell me maybe a calling search space issue.  Not that.  I've changed the
 jabber client to start calls with audio instead of video. I've checked 
regions and there are only defaults.  Would the jabber-config.xml file 
create issues?  I would think it might bugger up directory services but 
not actually calling.
  

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Re: [cisco-voip] External calls from Jabber

[Aaron Banks]

I will give that a shot and let you know.

From: ryanh...@outlook.com
To: amichaelba...@hotmail.com; avholloway+cisco-v...@gmail.com
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] External calls from Jabber
Date: Thu, 5 Nov 2015 20:03:51 +











Hello Aaron,









We would love to help you troubleshoot this issue; to do that we're at the 
point of needing to see 'what is going on under the hood'.










Please set up a test case for us by selecting a jabber desktop client impacted 
by this issue and clearing the local cache content of the client.




Cache location for PC Clients: 
http://www.cisco.com/c/en/us/support/docs/unified-communications/jabber-windows/116433-probsol-jabber-00.html

Cache location for Mac Clients: 
http://www.cisco.com/c/en/us/support/docs/unified-communications/jabber-mac/116682-technote-jabber-00.html

After you've clear the cache files; use the Jabber client to attempt an 
external call (replicate the same method the business unit is attempting to use 
for external calls that are failing). Immediately after the call fails 
(assuming it still does fail) and
 you get that annunciation message, I need you to end the call with the Jabber 
client and collect the "Calls" PRT (Problem Reporting Tool) logs from that 
Jabber client. You can access the PRT menu from the
Help->Report A Problem menu of the Jabber Client. Follow the on-screen 
instructions and it will eventually allow you to download an archive file.







The next thing I'd like for you to do is to open the Real-Time Reporting Tool 
(RTMT) application and point it to the publisher node of your CCM cluster (if 
you do not have the RTMT application, you can download it from the
Applications->Plugins section of CUCM). If needed, you can research RTMT at

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/8_6_1/rtmt/rtmt/rtinst.html.
 Once you are logged into the tool, please replicate the following workflow and 
follow the on-screen prompts to download the traces (make sure you get the 
traces
 from ALL CUCM nodes for a time period covering the time of the example call). 
You can research more about getting trace files at

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/8_6_1/rtmt/rtmt/rttlc.html.
 


















Once you have all the files gathered, upload them to dropbox/box .. etc and 
send a linkcinto this list along with the called and calling party number of 
the example call and the device name of the CSF profile used by the Jabber 
Desktop client for the example
 call.





Hope this help,





-Ryan










From: cisco-voip <cisco-voip-boun...@puck.nether.net> on behalf of Aaron Banks 
<amichaelba...@hotmail.com>

Sent: Thursday, November 5, 2015 1:51 PM

To: Anthony Holloway

Cc: cisco-voip@puck.nether.net

Subject: Re: [cisco-voip] External calls from Jabber
 





Per Charles' suggestion, I used the DNA tool to analyze the dialed call from 
jabber.  Nothing is being blocked.  Digits are not being dialed by clicking on 
the contact, the dial pad is being used in jabber or entered in the calling 
window.



Date: Thu, 5 Nov 2015 12:01:13 -0600

Subject: Re: [cisco-voip] External calls from Jabber

From: avholloway+cisco-v...@gmail.com

To: amichaelba...@hotmail.com

CC: cisco-voip@puck.nether.net



It's pretty straight forward, if you already have a grasp on CUCM core 
functionality.



Your reported error message sounds like the Annunciator telling you there were 
no matches in the dial plan for what you dialed.  You didn't mention what was 
dialed, so we cannot help you there.  If however, the error message was more 
like fast busy or
 ringback then call drop, I would say it was a failure tCacheo establish 
bi-directional media.  That's not that case though, so I would focus on your 
dialing habits and your matched patterns.



Also, Cisco Jabber does not use en bloc dialing, dispite the fact there is no 
dial tone.  Seems a bit backwards, but that's how it is.



On Thu, Nov 5, 2015 at 11:10 AM, Aaron Banks 
<amichaelba...@hotmail.com> wrote:







inShare



The jabber questions never end.  I am struggling with Jabber.  Users can call 
each other internally, check voicemail and IM each other, no problem.  When 
they go to make an external call to any number, they hear "your call cannot be 
completed as dialed"
 which would tell me maybe a calling search space issue.  Not that.  I've 
changed the jabber client to start calls with audio instead of video. I've 
checked regions and there are only defaults.  Would the jabber-config.xml file 
create issues?  I would think
 it might bugger up directory services but not actually calling.






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[cisco-voip] Jabber - auto detect server

[Aaron Banks]

I've had some trouble getting jabber to auto detect the IM server.  Every 
time a new user signs in, they have to put in the IP of the server.  Anyone 
have tricks to offer to avoid using the IP or FQDN?
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[cisco-voip] Unity Connection 10.5.2 Split Brain Recovery

[Aaron Banks]

Has anyone seen/resolved a split brain recovery in Unity Connection 10.5.2?  
The primary and secondary keep swapping back and forth every few minutes.  I 
can ping and trace to each server.  I restarted the primary but that did not 
resolve the issue.  In the RTMT system logs, the secondary sends an NTP query 
to the primary the response is the primary is inaccessible or down.  I'm 
stumped.
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Re: [cisco-voip] Unity Connection 10.5.2 Split Brain Recovery

[Aaron Banks]

Thank you for all of that.  You know what it was - NTP.  I shut down the HA.  
NTP was doing weird things on the primary node and I asked the customer if the 
NTP server address he gave me was a windows server.  Bingo.  I changed the NTP 
source, rebooted the primary, called voicemail and then powered on the HA.

Lesson learned.

From: ryanh...@outlook.com
To: amichaelba...@hotmail.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Unity Connection 10.5.2 Split Brain Recovery
Date: Tue, 27 Oct 2015 17:25:47 +






1.) Shut down the HA node.



2.) Reboot the primary node



3.) Once the primary node is up, place a call into voicemail 



4.) Power the HA node back on



5.) Once HA is up, verify HA status.










Sent from my T-Mobile 4G LTE Device





 Original message 

From: Aaron Banks 

Date:10/27/2015 12:35 PM (GMT-05:00) 

To: cisco-voip@puck.nether.net 

Subject: [cisco-voip] Unity Connection 10.5.2 Split Brain Recovery 







Has anyone seen/resolved a split brain recovery in Unity Connection 10.5.2?  
The primary and secondary keep swapping back and forth every few minutes.  I 
can ping and trace to each server.  I restarted the primary but that did not 
resolve the issue.  In the
 RTMT system logs, the secondary sends an NTP query to the primary the response 
is the primary is inaccessible or down.  I'm stumped.

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Re: [cisco-voip] CUCM 10.5 and Exchange 2013 voicemail setup

[Aaron Banks]

Terry - I did a SIP trunk set up for Brentwood College on Vancouver Island.  If 
you don't resolve it, you could send me screenshots and I'll have a look.

Aaron

> From: cisco-voip-requ...@puck.nether.net
> Subject: cisco-voip Digest, Vol 144, Issue 12
> To: cisco-voip@puck.nether.net
> Date: Wed, 14 Oct 2015 12:00:04 -0400
> 
> Send cisco-voip mailing list submissions to
>   cisco-voip@puck.nether.net
> 
> To subscribe or unsubscribe via the World Wide Web, visit
>   https://puck.nether.net/mailman/listinfo/cisco-voip
> or, via email, send a message with subject or body 'help' to
>   cisco-voip-requ...@puck.nether.net
> 
> You can reach the person managing the list at
>   cisco-voip-ow...@puck.nether.net
> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of cisco-voip digest..."
> 
> 
> Today's Topics:
> 
>1. CUCM 10.5 and Exchange 2013 voicemail setup. (Terry Oakley)
>2. Re: CUCM 10.5 and Exchange 2013 voicemail setup. (Ryan Huff)
>3. Re: CUCM 10.5 and Exchange 2013 voicemail setup. (Daniel Pagan)
>4. Cisco CAR DB not running (Erick Bergquist)
>5. Re: Cisco CAR DB not running (Ryan Huff)
>6. Re: Cisco CAR DB not running (Erick Bergquist)
>7. Re: Cisco CAR DB not running (Brian Meade)
>8. Jabber questions (Louis Koekemoer (ZA))
> 
> 
> --
> 
> Message: 1
> Date: Tue, 13 Oct 2015 16:24:02 -0600
> From: Terry Oakley 
> To: "cisco-voip@puck.nether.net" 
> Subject: [cisco-voip] CUCM 10.5 and Exchange 2013 voicemail setup.
> Message-ID:
>   <15f47b5df14db045a241b0b13672e6f013f19e6...@rdcexmail1.rdcsrvcs.ads>
> Content-Type: text/plain; charset="us-ascii"
> 
> We currently are moving from Exchange 2007 to Exchange 2013.   We have three 
> CAS servers and 3 Mailbox servers, all virtualized.   In our test 
> environment, before we moved from CUCM 6.1 to 10.5 we are able to at least 
> get Exchange 2013 to answer a SIP trunk request from CUCM 6.1.   Now in CUCM 
> 10.5 we just get a fast busy when we dial the VM pilot number.   Does anyone 
> have experience with this and have a guide that we could follow?We have 
> followed a number of guides from Microsoft and they have not proven to be the 
> magic answer.
> 
> We have a SIP trunk set to the CAS servers with all three individual servers 
> listed in the Destination section (all FQDN) port 5060
> We have three separate SIP trunks to the three mailbox servers with all three 
> having the ports 5062 through 5068 listed and again FQDN for the destination 
> address.
> The VM pilot (route pattern) is associated with the CAS trunk.
> Do we need a route list and hence a route group?
> 
> Thank you for your knowledge and wiliness to share.  And especially thanks to 
> this forum for providing us the access.
> 
> Cheers
> 
> Terry
> 
> Terry Oakley
> Telecommunications Coordinator | Information Technology Services
> Red Deer College |100 College Blvd. | Box 5005 | Red Deer | Alberta | T4N 5H5
> work (403) 342-3521   |  FAX (403) 343-4034
> 
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> 
> --
> 
> Message: 2
> Date: Tue, 13 Oct 2015 19:29:04 -0400
> From: Ryan Huff 
> To: Terry Oakley ,
>   "cisco-voip@puck.nether.net" 
> Subject: Re: [cisco-voip] CUCM 10.5 and Exchange 2013 voicemail setup.
> Message-ID: 
> Content-Type: text/plain; charset="utf-8"
> 
> Terry,
> 
> Sounds like you have a lot going on there!
> 
> How did you move into the 10.5 environment? ?Did you do a bridge migration or 
> a 'stare and compare'?
> 
> A fast busy could be a few different things (css, partition ... etc) or dns 
> based since you mentioned fqdn or resource based.
> 
> What codec are you trying using?
> 
> Have you pulled traces?
> 
> What is the disconnect cause code for one of the failed calls into the hunt 
> pilot?
> 
> If you can reproduce a failed call and then send me the traces or the sip 
> messages I can give you a much better answer.
> 
> Thanks,
> 
> Ryan
> 
> 
> Sent from my T-Mobile 4G LTE Device
> 
>  Original message 
> From: Terry Oakley  
> Date:10/13/2015  6:24 PM  (GMT-05:00) 
> To: cisco-voip@puck.nether.net 
> Subject: [cisco-voip] CUCM 10.5 and Exchange 2013 voicemail setup. 
> 
> We currently are moving from Exchange 2007 to Exchange 2013.?? We have three 
> CAS servers and 3 Mailbox servers, all virtualized.?? In our test 
> environment, before we moved from CUCM 6.1 to 10.5 we are able to at least 
> get Exchange 2013 to answer a SIP trunk request from CUCM 6.1.?? Now in CUCM 
> 10.5 we just get a fast busy when we 

Re: [cisco-voip] cisco-voip Digest, Vol 142, Issue 25

[Aaron Banks]

Thank you to everyone who responded to me.  It was a service provider issue and 
they didn't admit it until the 3rd day when the issue was escalated to someone 
who was willing to do something about it.

I will take these tips away for future use because I'm sure it won't be the 
last time with this certain provider!

 From: cisco-voip-requ...@puck.nether.net
 Subject: cisco-voip Digest, Vol 142, Issue 25
 To: cisco-voip@puck.nether.net
 Date: Sat, 29 Aug 2015 12:00:03 -0400
 
 Send cisco-voip mailing list submissions to
   cisco-voip@puck.nether.net
 
 To subscribe or unsubscribe via the World Wide Web, visit
   https://puck.nether.net/mailman/listinfo/cisco-voip
 or, via email, send a message with subject or body 'help' to
   cisco-voip-requ...@puck.nether.net
 
 You can reach the person managing the list at
   cisco-voip-ow...@puck.nether.net
 
 When replying, please edit your Subject line so it is more specific
 than Re: Contents of cisco-voip digest...
 
 
 Today's Topics:
 
1. Service Provider SIP Trunks (Aaron Banks)
2. Re: Service Provider SIP Trunks (Ryan Huff)
3. Re: Service Provider SIP Trunks (Daniel Pagan)
4. Re: Service Provider SIP Trunks (Ryan Huff)
5. Re: Service Provider SIP Trunks (Daniel Pagan)
 
 
 --
 
 Message: 1
 Date: Fri, 28 Aug 2015 12:34:54 -0700
 From: Aaron Banks amichaelba...@hotmail.com
 To: cisco-voip@puck.nether.net cisco-voip@puck.nether.net
 Subject: [cisco-voip] Service Provider SIP Trunks
 Message-ID: blu182-w2574baa20ca25f0c2c91e3bc...@phx.gbl
 Content-Type: text/plain; charset=iso-8859-1
 
 
 
 I have a strange problem with SIP trunks.  Let me preface this with the 
 service provider moved the SIP trunks to a different device and that's what 
 certain calls stopped working.  Before this move, everything was tested and 
 working for 6 weeks.  Read on, someone might have had the same experience.
 
 Post SIP trunk move, callers inside of the organization cannot call 911 or a 
 mobile phone (ANY mobile phone).  When they dial the number, let's use 911 
 for example, the call rings once, the calling line ID is delivered to 911 and 
 then the call goes to busy.  So 911 knows that organization called.  The same 
 thing happens with mobile phones.  All other call types (local, LD, 
 international) work.  If I call forward a phone from inside of the 
 organization to a mobile phone and call that forwarded phone (from outside or 
 inside), the call is redirected to the mobile, call is answered and then the 
 call drops.  If I forward that same phone inside of the organization to an 
 outside land line ((either local or LD), the call is successful.
 
 For 911 or the mobile calls that fail, the SIP trace reveals a 500 (internal 
 server error), a BYE message with reason Q.850; cause=16.  The SIP call 
 messages show the state of the call is DEAD.
 
 My question is why would the number make any difference at all?  Has anyone 
 ever seen this type of issue before?  The provider says it is CUCM 10.5/Voice 
 GW 2901 that is rejecting the call.  I disagree.
 
 Many thanks
 
 Aaron
 
 
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 --
 
 Message: 2
 Date: Fri, 28 Aug 2015 15:38:36 -0400
 From: Ryan Huff ryanh...@outlook.com
 To: amichaelba...@hotmail.com, cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] Service Provider SIP Trunks
 Message-ID: col401-eas1249a294d219f1c393d1feec5...@phx.gbl
 Content-Type: text/plain; charset=utf-8
 
 Sounds like a codec/media issue. Are you supporting early offer?
 
 Thanks,
 
 Ryan
 
  Original Message 
 From: Aaron Banks amichaelba...@hotmail.com
 Sent: Friday, August 28, 2015 03:35 PM
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] Service Provider SIP Trunks
 
 
 
 I have a strange problem with SIP trunks.  Let me preface this with the 
 service provider moved the SIP trunks to a different device and that's what 
 certain calls stopped working.  Before this move, everything was tested and 
 working for 6 weeks.  Read on, someone might have had the same experience.
 
 Post SIP trunk move, callers inside of the organization cannot call 911 or a 
 mobile phone (ANY mobile phone).  When they dial the number, let's use 911 
 for example, the call rings once, the calling line ID is delivered to 911 
 and then the call goes to busy.  So 911 knows that organization called.  The 
 same thing happens with mobile phones.  All other call types (local, LD, 
 international) work.  If I call forward a phone from inside of the 
 organization to a mobile phone and call that forwarded phone (from outside 
 or inside), the call is redirected to the mobile, call is answered and then 
 the call drops.  If I forward that same phone inside

[cisco-voip] Service Provider SIP Trunks

[Aaron Banks]

I have a strange problem with SIP trunks.  Let me preface this with the service 
provider moved the SIP trunks to a different device and that's what certain 
calls stopped working.  Before this move, everything was tested and working for 
6 weeks.  Read on, someone might have had the same experience.

Post SIP trunk move, callers inside of the organization cannot call 911 or a 
mobile phone (ANY mobile phone).  When they dial the number, let's use 911 for 
example, the call rings once, the calling line ID is delivered to 911 and then 
the call goes to busy.  So 911 knows that organization called.  The same thing 
happens with mobile phones.  All other call types (local, LD, international) 
work.  If I call forward a phone from inside of the organization to a mobile 
phone and call that forwarded phone (from outside or inside), the call is 
redirected to the mobile, call is answered and then the call drops.  If I 
forward that same phone inside of the organization to an outside land line 
((either local or LD), the call is successful.

For 911 or the mobile calls that fail, the SIP trace reveals a 500 (internal 
server error), a BYE message with reason Q.850; cause=16.  The SIP call 
messages show the state of the call is DEAD.

My question is why would the number make any difference at all?  Has anyone 
ever seen this type of issue before?  The provider says it is CUCM 10.5/Voice 
GW 2901 that is rejecting the call.  I disagree.

Many thanks

Aaron

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[cisco-voip] Unity Auto Attendant question

[Aaron Banks]

I've built a simple auto attendant which the staff forwards to after hours: 
1-staff directory; 0 for general mailbox.  0 works.  When I press 1 (which is a 
voice enabled directory), say the person's name, I am redirected to their 
extension, but I never get to their mailbox.  The call is redirected back to 
the auto attendant.  This is a call initiated from outside of the organization. 
 If I call the auto attendant inside, same result.  If I have someone 
physically perform a transfer to the auto attendant, it works.  I get the 
extension and eventually hit the mailbox.  

I have gone as far as call forwarding someone's phone directly to voicemail, in 
additional to call forwarding to a cell phone, and if no physical transfer is 
performed, I hear the auto attendant again.  I've changed the CSS to be wide 
open with no success.

If no call forwarding is done on the main phone (ie it rings until the AA 
answers) - same.  Without a person physically transferring to the AA, the 
caller will hear the AA over and over again.

I've rebuilt both the call handler and the directory handler.  I have turned of 
the voice enabled option in favor of dial by name.  Same result.

The version CUCM/CUC 10.5.2.12900-14.  I should mention that the connection 
between CUCM and CUC is a SIP trunk.  Could this be a bug (in my head)?

TIA if you have an additional ideas.


Aaron
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Re: [cisco-voip] Attendant Console User 10.5.1 can't change layout?

[Aaron Banks]

Jason

I just deployed this a couple of weeks ago.  You can't change much of anything. 
 The customer's major complaint was the wasted real estate and their inability 
to move panes around.  They don't use call park either.  Sorry I don't have 
anything to offer.  The product's aesthetics kind of suck.  The only reason 
they bought it was for the reporting.

Aaron

 From: cisco-voip-requ...@puck.nether.net
 Subject: cisco-voip Digest, Vol 134, Issue 8
 To: cisco-voip@puck.nether.net
 Date: Mon, 8 Dec 2014 12:00:01 -0500
 
 Send cisco-voip mailing list submissions to
   cisco-voip@puck.nether.net
 
 To subscribe or unsubscribe via the World Wide Web, visit
   https://puck.nether.net/mailman/listinfo/cisco-voip
 or, via email, send a message with subject or body 'help' to
   cisco-voip-requ...@puck.nether.net
 
 You can reach the person managing the list at
   cisco-voip-ow...@puck.nether.net
 
 When replying, please edit your Subject line so it is more specific
 than Re: Contents of cisco-voip digest...
 
 
 Today's Topics:
 
1. Anyone looking for a CCE/CVP Engineer? (Ryan Burtch)
2. Attendant Console User 10.5.1 can't change layout?
   (Jason Aarons (AM))
 
 
 --
 
 Message: 1
 Date: Mon, 8 Dec 2014 00:10:54 -0500
 From: Ryan Burtch rburt...@gmail.com
 To: cisco-voip@puck.nether.net cisco-voip@puck.nether.net
 Subject: [cisco-voip] Anyone looking for a CCE/CVP Engineer?
 Message-ID:
   CAK+Shf5=25pis_85Kwx0m9i=ob69dhnokqd+w2e_zxpdorc...@mail.gmail.com
 Content-Type: text/plain; charset=utf-8
 
 *All:*
 
 I don't usually do this, but just wanted to ask if anyone is in need of a
 CCE/CVP Engineer?
 
 My company just got acquired, so there is some uncertainty here.
 
 If you know of anything, just email me off-net and I'll send you my details.
 
 Appreciate the help.
 
 
 
 
 Sincerely,
 
 Ryan Burtch
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 --
 
 Message: 2
 Date: Mon, 8 Dec 2014 15:32:25 +
 From: Jason Aarons (AM) jason.aar...@dimensiondata.com
 To: cisco-voip (cisco-voip@puck.nether.net)
   cisco-voip@puck.nether.net
 Subject: [cisco-voip] Attendant Console User 10.5.1 can't change
   layout?
 Message-ID:
   2eb6888cfb98614ea7384beb9af8b382179...@usispsvexdb03.na.didata.local
 Content-Type: text/plain; charset=windows-1252
 
 Attendant Console Advanced 10.5.1
 
 Customer doesn't use Call Park and would to move it around on screen or get 
 rid of it all together.
 
 Is there a way to customize the screen layout? For example move Speed Dials 
 F6 to the left and Call Park to the right?  See attached png for screenshot 
 from manual.
 
 http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/cucmac/cuaca/10_5_1/user_guide/eng/CUACA1051CUG_eng.pdf
 
 
 
 
 
 
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 End of cisco-voip Digest, Vol 134, Issue 8
 **
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