Re: [cisco-voip] S2 Netbox and CUCM

2014-02-12 Thread Ryan Huff
You should be able to setup a standard SIP trunk and point a route pattern at 
Netbox through the trunk. Then just put a SD to the RP on a phone.

Sent from my iPhone

 On Feb 12, 2014, at 12:13 PM, Ben Story ben.st...@gmail.com wrote:
 
 Has anyone ever done any integration between S2 Netbox security and Cisco IP 
 phones?  Specifically we would like to be able to have an extension or button 
 on a phone open a door.
 --
 Ben Story 
 CCSP, CCNA, CCNA Wireless, CCDA
 ben.st...@gmail.com
 @ntwrk80
 http://showbrain.blogspot.com
 http://rand0mw0rds.blogspot.com
 
 
 From sour-faced saints and silly devotions, good Lord, preserve us!. -- St. 
 Teresa of Avila
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Re: [cisco-voip] 2 different jabber TCT devices for one CUWL Standard User

2014-02-18 Thread Ryan Huff
It wouldn't do what I think you're trying to do. Is it possible to put both 
dn's on the same dual mode profile?

Sent from my iPad

 On Feb 17, 2014, at 1:53 PM, Anthony Kouloglou ak...@dataways.gr wrote:
 
 Hi all,
 what would be the effect of creating 2 different TCT devices, assigning the 
 same owner and associating this end user with both devices?
 Assume that ELM has a CUWL standard license available and the platforms are 
 Cisco IM  P 9.1 and CUCM 9.1.
 The idea is to have 2 different jabber numbers:
 one for an iphone for US
 one for an iphone for Europe.
 
 Cheers,
 Anthony
 
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Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat

2014-03-15 Thread Ryan Huff
Is it an option to use LDAP integration? There are certain cups features that 
won't work with out it. In a lab environment, you can always use OpenLDAP or 
something like that.

Sent from my iPad

 On Mar 15, 2014, at 9:11 PM, Jeffrey Girard jeffrey.gir...@girardinc.com 
 wrote:
 
 Jason –
   I appreciate your response.
  
   As to your first question, yes, I have setup CCMCIP.
  
   From my original posting
  
 Application - Legacy Clients - CCMCIP Profile.  I 
 created a new profile, set the Primary and Secondary CCMCIP Host to be the IP 
 Address of the CUCM Publisher.  For Server Certification Verfication, I 
 selected Any Certificate from the drop down.  I assigned this profile to all 
 4 of my users.
  
   When I mouse over a contact, or right click and select View 
 Profile, there is no indication of an IM address at all.  If I select Edit 
 Contact, I do not have the ability to enter an address into the Instant 
 Messaging Address window.  For all of my users, I have had to manually create 
 the contacts as I am not using LDAP integration. 
  
   This is why Im not able to initiate a chat – but the question is – 
 why am I not able to manually enter the IM address like I can the DN?
  
   Im also not using an AD domain, so under CUPS - System - Cluster 
 Topology, I have entered DOMAIN.NET.SET (which I retrieved from CUCM 
 enterprise parameters).
  
   Im beginning to wonder if CUPS is so dependent upon DNS and LDAP 
 that not having them available is causing my failures.
  
   As for the log files, yes, I have looked into the files in CUPC8 - 
 Local Settings and CUPC8 - App Data.  Under Client Services Framework - 
 Local Settings - Logs, I have searched for “error” and it only matches 
 against entries for video.
  
   I have not attempted any Wireshark captures yet.
  
 
 Dr. Jeffrey T. Girard (Jeff), PhD
 Colonel, United States Army (Retired)
 Senior Network Engineer / VoIP Engineer - WireMeHappy.com
 reply to: jeffrey.gir...@wiremehappy.com
 (607)835-0406 (home office)
 (845)764-1661 (mobile)
 (607)835-0458 (fax)
  
 From: Jason Aarons (AM) [mailto:jason.aar...@dimensiondata.com] 
 Sent: Saturday, March 15, 2014 7:40 PM
 To: Jeffrey Girard; cisco-voip@puck.nether.net
 Subject: RE: [cisco-voip] CUPC - Unable to go into deskphone control mode and 
 unable to chat
  
 In IMP do you have a CCMCIP setup?
  
 When you put a mouse over a contact or select the Contact Details what is the 
 IM address?  It should match your Presence Domain.
  
 Check the log files/Wireshark.
  
 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
 Jeffrey Girard
 Sent: Friday, March 14, 2014 6:12 PM
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] CUPC - Unable to go into deskphone control mode and 
 unable to chat
  
  
 
 Home Lab – learning Presence
  
 CUCM 9
 CUPS 9 (not 8.6 as virtually all of the help and discussion forums, 
 documents, etc are for)
 CUPC 8.6 (3 copies running on 3 different laptops)
 4 users configured in CUCM for Presence capabilities
 No LDAP integration
  
  
 Current status:
 Presence indicators for phones works fine, no issues
 3 users can log into their CUPC clients and Presence information from their 
 own associated deskphone displays correctly (ie if User 1 takes his deskphone 
 offhook, the CUPC for User 1 shows as “On the Phone”
 Users can place phone calls with each other by typing the DN in the “Search 
 for name or number” field
  
 I have two different issues that I have not been able to solve all day – they 
 may be related to each other.
  
 The first issue:  I am unable to go into deskphone mode.  Although the 
 checkbox is visible – it appears to be greyed out and clicking in it does 
 nothing.
 The second issue:  I am unable to start a chat session.  I highlite a contact 
 (that I manually entered), right click, and select chat.  I get an error 
 message that says “Failed to start conversation.  Invalid parameter”
  
 I have scoured the Cisco site for docs and most of them pertain to CUPS 8.6 
 and not 9.  There is a difference in that v9 does not have the Application - 
 IP Phone Messenger selection.
  
 My current configs:
 CUCM
   4 application users – CUPS-AXL (with Stand CCM SuperUsers 
 permissions), CUPS-Deskphone, CUPS-CTIGW, and CUPS-PhoneMSG (all with 
 Standard CTI Allow control of all devices).
   3 end users – From top to bottom, all users have passwords and 
 digest credentials.  Under Service Settings, all users have the “Enable user 
 for Unified CM IM and Presence” checked.  This is the replacement for the 
 assigning license capabilities from version 8.6.  Also under Service 
 Settings, all users have a UC Service Profile assigned.  The UC Service 
 Profile has two UC service settings – a Presence and IM Profile and a CTI 
 Profile.  

Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat

2014-03-19 Thread Ryan Huff
H ... CUPC is old tech . Let me fire up the not-so-way-back machine.  
Device and client control almost always comes down to an ownership issue 

1. End User has Standard CTI Enabled privilege 
2. User owns the Client Services Framework profile
3. User owns the actual phone (unless just a CSF user)
4. User is associated to the DN, of the prime line on the Phone/CSF
5. The Primary Phone field in the phone/CSF config is set to the Mac of the 
phone the user owns (only if there is an actual phone for the user)
6. End User has primary extension field set to the prime line of the CSF / 
Phone config
8. For kicks, play with the end user permissions too if none of the above 
works, give the user all permissions and see what that does.

Let me know how it works out for you.

Sent from my iPad

 On Mar 19, 2014, at 9:51 PM, Jeffrey Girard jeffrey.gir...@girardinc.com 
 wrote:
 
 Update –
   Well, I finally bit the bullet and stood up a MS AD domain in my 
 lab.
  
   I integrated CUCM into the LDAP and imported the users from LDAP
  
   I also stood up a DNS server.
  
   I reconfigured CUPS to the new domain, joined all of the endpoint 
 laptops to the domain, and retested.
  
   Same place as I was before with the deskphone mode – however, my 
 ability to IM is fixed (as was expected).
  
   Using CUPS, I am not able to go into deskphone control mode.  The 
 option box is still greyed out.
  
   However, I installed Jabber for Windows on the same laptops.
  
   In Jabber, I am able to select the option at the bottom of the 
 window to use my deskphone for calls, and then I can move it back to the 
 Jabber client.
  
   So, anyone have any ideas why CUPC refuses to let me go into 
 deskphone control mode?
  
   All features and functions of the JFW work great.  No problems.
  
 Jeff
  
 From: Ryan Huff [mailto:rthconsulta...@gmail.com] 
 Sent: Saturday, March 15, 2014 10:43 PM
 To: Jeffrey Girard
 Cc: Jason Aarons (AM); cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and 
 unable to chat
  
 Is it an option to use LDAP integration? There are certain cups features that 
 won't work with out it. In a lab environment, you can always use OpenLDAP or 
 something like that.
 
 Sent from my iPad
 
 On Mar 15, 2014, at 9:11 PM, Jeffrey Girard jeffrey.gir...@girardinc.com 
 wrote:
 
 Jason –
   I appreciate your response.
  
   As to your first question, yes, I have setup CCMCIP.
  
   From my original posting
  
 Application - Legacy Clients - CCMCIP Profile.  I 
 created a new profile, set the Primary and Secondary CCMCIP Host to be the IP 
 Address of the CUCM Publisher.  For Server Certification Verfication, I 
 selected Any Certificate from the drop down.  I assigned this profile to all 
 4 of my users.
  
   When I mouse over a contact, or right click and select View 
 Profile, there is no indication of an IM address at all.  If I select Edit 
 Contact, I do not have the ability to enter an address into the Instant 
 Messaging Address window.  For all of my users, I have had to manually create 
 the contacts as I am not using LDAP integration. 
  
   This is why Im not able to initiate a chat – but the question is – 
 why am I not able to manually enter the IM address like I can the DN?
  
   Im also not using an AD domain, so under CUPS - System - Cluster 
 Topology, I have entered DOMAIN.NET.SET (which I retrieved from CUCM 
 enterprise parameters).
  
   Im beginning to wonder if CUPS is so dependent upon DNS and LDAP 
 that not having them available is causing my failures.
  
   As for the log files, yes, I have looked into the files in CUPC8 - 
 Local Settings and CUPC8 - App Data.  Under Client Services Framework - 
 Local Settings - Logs, I have searched for “error” and it only matches 
 against entries for video.
  
   I have not attempted any Wireshark captures yet.
  
 
 Dr. Jeffrey T. Girard (Jeff), PhD
 Colonel, United States Army (Retired)
 Senior Network Engineer / VoIP Engineer - WireMeHappy.com
 reply to: jeffrey.gir...@wiremehappy.com
 (607)835-0406 (home office)
 (845)764-1661 (mobile)
 (607)835-0458 (fax)
  
 From: Jason Aarons (AM) [mailto:jason.aar...@dimensiondata.com] 
 Sent: Saturday, March 15, 2014 7:40 PM
 To: Jeffrey Girard; cisco-voip@puck.nether.net
 Subject: RE: [cisco-voip] CUPC - Unable to go into deskphone control mode and 
 unable to chat
  
 In IMP do you have a CCMCIP setup?
  
 When you put a mouse over a contact or select the Contact Details what is the 
 IM address?  It should match your Presence Domain.
  
 Check the log files/Wireshark.
  
 From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
 Jeffrey Girard
 Sent: Friday, March 14, 2014

Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat

2014-03-20 Thread Ryan Huff
Well at this point is say get a TAC case opened if it's for production

Sent from my iPhone

 On Mar 20, 2014, at 9:14 AM, Jeffrey Girard jeffrey.gir...@girardinc.com 
 wrote:
 
 Replies inline….
 Jeff
  
 From: Ryan Huff [mailto:rthconsulta...@gmail.com] 
 Sent: Wednesday, March 19, 2014 10:21 PM
 To: Jeffrey Girard
 Cc: cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and 
 unable to chat
  
 H ... CUPC is old tech . Let me fire up the not-so-way-back machine.  
 Device and client control almost always comes down to an ownership issue 
  
 1.  End User has Standard CTI Enabled privilege
  
 Yes
  
 2.  User owns the Client Services Framework profile
 Yes
 3.  User owns the actual phone (unless just a CSF user)
 Yes
 4.  User is associated to the DN, of the prime line on the Phone/CSF
 Yes
 5.  The Primary Phone field in the phone/CSF config is set to the Mac of 
 the phone the user owns (only if there is an actual phone for the user)
 Yes
 6.  End User has primary extension field set to the prime line of the CSF 
 / Phone config
 Yes.  Configured in Active Directory, imported via LDAP, confirmed on End 
 User Page
 7.  For kicks, play with the end user permissions too if none of the 
 above works, give the user all permissions and see what that does.
 Assigned the user to all privilege groups.  Went to CUPS and restarted the 
 DirSync service.  Closed out of the CUPS application and restarted.  No 
 change.  In CUPC, I am unable to go into deskphone control mode – the option 
 is displayed at the bottom of the window, but its greyed out.  As with 
 yesterday, JFW work great.
  
 So, I am baffled.
  
 Let me know how it works out for you.
 
 Sent from my iPad
 
 On Mar 19, 2014, at 9:51 PM, Jeffrey Girard jeffrey.gir...@girardinc.com 
 wrote:
 
 Update –
   Well, I finally bit the bullet and stood up a MS AD domain in my 
 lab.
  
   I integrated CUCM into the LDAP and imported the users from LDAP
  
   I also stood up a DNS server.
  
   I reconfigured CUPS to the new domain, joined all of the endpoint 
 laptops to the domain, and retested.
  
   Same place as I was before with the deskphone mode – however, my 
 ability to IM is fixed (as was expected).
  
   Using CUPS, I am not able to go into deskphone control mode.  The 
 option box is still greyed out.
  
   However, I installed Jabber for Windows on the same laptops.
  
   In Jabber, I am able to select the option at the bottom of the 
 window to use my deskphone for calls, and then I can move it back to the 
 Jabber client.
  
   So, anyone have any ideas why CUPC refuses to let me go into 
 deskphone control mode?
  
   All features and functions of the JFW work great.  No problems.
  
 Jeff
  
 From: Ryan Huff [mailto:rthconsulta...@gmail.com] 
 Sent: Saturday, March 15, 2014 10:43 PM
 To: Jeffrey Girard
 Cc: Jason Aarons (AM); cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and 
 unable to chat
  
 Is it an option to use LDAP integration? There are certain cups features that 
 won't work with out it. In a lab environment, you can always use OpenLDAP or 
 something like that.
 
 Sent from my iPad
 
 On Mar 15, 2014, at 9:11 PM, Jeffrey Girard jeffrey.gir...@girardinc.com 
 wrote:
 
 Jason –
   I appreciate your response.
  
   As to your first question, yes, I have setup CCMCIP.
  
   From my original posting
  
 Application - Legacy Clients - CCMCIP Profile.  I 
 created a new profile, set the Primary and Secondary CCMCIP Host to be the IP 
 Address of the CUCM Publisher.  For Server Certification Verfication, I 
 selected Any Certificate from the drop down.  I assigned this profile to all 
 4 of my users.
  
   When I mouse over a contact, or right click and select View 
 Profile, there is no indication of an IM address at all.  If I select Edit 
 Contact, I do not have the ability to enter an address into the Instant 
 Messaging Address window.  For all of my users, I have had to manually create 
 the contacts as I am not using LDAP integration. 
  
   This is why Im not able to initiate a chat – but the question is – 
 why am I not able to manually enter the IM address like I can the DN?
  
   Im also not using an AD domain, so under CUPS - System - Cluster 
 Topology, I have entered DOMAIN.NET.SET (which I retrieved from CUCM 
 enterprise parameters).
  
   Im beginning to wonder if CUPS is so dependent upon DNS and LDAP 
 that not having them available is causing my failures.
  
   As for the log files, yes, I have looked into the files in CUPC8 - 
 Local Settings and CUPC8 - App Data.  Under Client Services Framework - 
 Local Settings - Logs, I have searched for “error” and it only matches 
 against entries for video

Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and unable to chat

2014-03-20 Thread Ryan Huff
At this point if say get a TAC case opened if it is production and you have a 
contract/support.

I retired my use of CUPC and moved to JFW . CUPC should be working as long 
as all the ownership stuff is setup correctly.

Sent from my iPhone

 On Mar 20, 2014, at 9:14 AM, Jeffrey Girard jeffrey.gir...@girardinc.com 
 wrote:
 
 Replies inline….
 Jeff
  
 From: Ryan Huff [mailto:rthconsulta...@gmail.com] 
 Sent: Wednesday, March 19, 2014 10:21 PM
 To: Jeffrey Girard
 Cc: cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and 
 unable to chat
  
 H ... CUPC is old tech . Let me fire up the not-so-way-back machine.  
 Device and client control almost always comes down to an ownership issue 
  
 1.  End User has Standard CTI Enabled privilege
  
 Yes
  
 2.  User owns the Client Services Framework profile
 Yes
 3.  User owns the actual phone (unless just a CSF user)
 Yes
 4.  User is associated to the DN, of the prime line on the Phone/CSF
 Yes
 5.  The Primary Phone field in the phone/CSF config is set to the Mac of 
 the phone the user owns (only if there is an actual phone for the user)
 Yes
 6.  End User has primary extension field set to the prime line of the CSF 
 / Phone config
 Yes.  Configured in Active Directory, imported via LDAP, confirmed on End 
 User Page
 7.  For kicks, play with the end user permissions too if none of the 
 above works, give the user all permissions and see what that does.
 Assigned the user to all privilege groups.  Went to CUPS and restarted the 
 DirSync service.  Closed out of the CUPS application and restarted.  No 
 change.  In CUPC, I am unable to go into deskphone control mode – the option 
 is displayed at the bottom of the window, but its greyed out.  As with 
 yesterday, JFW work great.
  
 So, I am baffled.
  
 Let me know how it works out for you.
 
 Sent from my iPad
 
 On Mar 19, 2014, at 9:51 PM, Jeffrey Girard jeffrey.gir...@girardinc.com 
 wrote:
 
 Update –
   Well, I finally bit the bullet and stood up a MS AD domain in my 
 lab.
  
   I integrated CUCM into the LDAP and imported the users from LDAP
  
   I also stood up a DNS server.
  
   I reconfigured CUPS to the new domain, joined all of the endpoint 
 laptops to the domain, and retested.
  
   Same place as I was before with the deskphone mode – however, my 
 ability to IM is fixed (as was expected).
  
   Using CUPS, I am not able to go into deskphone control mode.  The 
 option box is still greyed out.
  
   However, I installed Jabber for Windows on the same laptops.
  
   In Jabber, I am able to select the option at the bottom of the 
 window to use my deskphone for calls, and then I can move it back to the 
 Jabber client.
  
   So, anyone have any ideas why CUPC refuses to let me go into 
 deskphone control mode?
  
   All features and functions of the JFW work great.  No problems.
  
 Jeff
  
 From: Ryan Huff [mailto:rthconsulta...@gmail.com] 
 Sent: Saturday, March 15, 2014 10:43 PM
 To: Jeffrey Girard
 Cc: Jason Aarons (AM); cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] CUPC - Unable to go into deskphone control mode and 
 unable to chat
  
 Is it an option to use LDAP integration? There are certain cups features that 
 won't work with out it. In a lab environment, you can always use OpenLDAP or 
 something like that.
 
 Sent from my iPad
 
 On Mar 15, 2014, at 9:11 PM, Jeffrey Girard jeffrey.gir...@girardinc.com 
 wrote:
 
 Jason –
   I appreciate your response.
  
   As to your first question, yes, I have setup CCMCIP.
  
   From my original posting
  
 Application - Legacy Clients - CCMCIP Profile.  I 
 created a new profile, set the Primary and Secondary CCMCIP Host to be the IP 
 Address of the CUCM Publisher.  For Server Certification Verfication, I 
 selected Any Certificate from the drop down.  I assigned this profile to all 
 4 of my users.
  
   When I mouse over a contact, or right click and select View 
 Profile, there is no indication of an IM address at all.  If I select Edit 
 Contact, I do not have the ability to enter an address into the Instant 
 Messaging Address window.  For all of my users, I have had to manually create 
 the contacts as I am not using LDAP integration. 
  
   This is why Im not able to initiate a chat – but the question is – 
 why am I not able to manually enter the IM address like I can the DN?
  
   Im also not using an AD domain, so under CUPS - System - Cluster 
 Topology, I have entered DOMAIN.NET.SET (which I retrieved from CUCM 
 enterprise parameters).
  
   Im beginning to wonder if CUPS is so dependent upon DNS and LDAP 
 that not having them available is causing my failures.
  
   As for the log files, yes, I have looked into the files in CUPC8 - 
 Local Settings and CUPC8 - App

Re: [cisco-voip] CIACO VG224 issue

2014-03-31 Thread Ryan Huff
Have you tried downloading a new image from Cisco.com?

Checksum error may indicate a corrupted image (among other things).

What does the topology between your tftp server and gateway look like? layer 2? 
Layer 3? Try and get your tftp server on the the same segment as the gateway if 
it isn't already.

Sent from my iPhone

 On Mar 31, 2014, at 3:04 PM, Sivakumar Donthamchetty 
 s...@dateksystems.com wrote:
 
 Hi,
 
 I have a VG224 and not able to boot it.
 
 When I boot it comes to ROMMON prompt.
 
 I tried to install new image file vi TFTP, it downloads the file and at the
 end it says checksum error
 
 Please need help how I can rectify this unit?
 
 Regards
 Siva
 
 
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Re: [cisco-voip] CUC 8.6.2.20000 Import from CUCM takes 15 minutes...

2014-03-31 Thread Ryan Huff
Unified reporting... 
- Does the cluster and db replication show healthy?

Unified serviceability
- bounce the BAT service

Sent from my iPhone

 On Mar 31, 2014, at 1:29 AM, Jonathan Charles jonv...@gmail.com wrote:
 
 Repeatable; just started... can add users locally, but cannot import from 
 CUCM... searching takes forever, and once you find them, the import takes 
 forever as well (realistically, about 15 minutes for each...)
 
 Any ideas?
 
 
 
 Jonathan
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Re: [cisco-voip] Incoming CLI Unknown but shows up at bottom of display

2014-04-01 Thread Ryan Huff
So by CNAM dips, what I mean is that in telephony, it is generally the
responsibility of the terminating carrier (the carrier terminating the
inbound call to your peer) to provide the lookup CLID and pass that on to
your peer (http://www.voip-info.org/wiki/view/CNAM). If they don't do the
lookup, all you'll get is the calling party number. Generally, SIP
providers charge extra for this service (on your bill the line item might
read like 'CNAM lookup service  queries').

Your trunk looks good. I would verify that your provider is passing the
CLID name.

Thanks,

Ryan Huff


On Tue, Apr 1, 2014 at 4:45 AM, Damian Turburville
d_turburvi...@yahoo.comwrote:

 I'm afraid I don't know about the CNAM dips, what exactly do you mean?
 Here are the screenshots for the SIP trunks though. I wouldn't be
 surprised if these are not set up optimally as it was done in rather a
 hurry and since it works I haven't revisited it.
 Thanks
   On Tuesday, 1 April 2014, 0:07, Ryan Huff rthconsulta...@gmail.com
 wrote:
  So you're getting the calling party number, but not the calling party
 name?

 If that's the case, is your SIP provider doing CNAM dips for termination?

 Can you provide a screenshot of your trunk configuration between the cucm
 and SBC from the cucm's perspective?

 Thanks,

 Ryan

 Sent from my iPhone

 On Mar 31, 2014, at 10:30 AM, Damian Turburville d_turburvi...@yahoo.com
 wrote:

 Unfortunately we do not have a CUBE, the calls are delivered from our
 carrier via SIP to our SBC and then over a direct SIP trunk to CUCM6.
 I just dont understand why the call shows as Unknown Number and yet the
 caller number is shown at the bottom of the screen (see attached photo)
 Here are some of the SIP traces from RTMT (I have anonymised the data a
 bit)

 03/31/2014 11:46:55.114 CCM|//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP
 message size 868 from 10.133.201.81:[5060]:
 INVITE sip:44404900@10.168.8.76 SIP/2.0
 Via: SIP/2.0/UDP 10.133.201.81:5060;branch=z9hG4bKac1195424339
 Max-Forwards: 10
 From: sip:+44266...@bnsplc.com;user=phone;tag=1c1195243906
 To:  *404900 sip:44404900@*
 Call-ID: 11951844583132014104653@10.133.201.81
 CSeq: 1 INVITE
 Contact: sip:+44266000@10.133.201.81:5060
 Supported: 100rel,timer
 Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
 Session-Expires: 900;refresher=uac
 Min-SE: 60
 User-Agent: Mediant 4000/v.6.60A.265.010
 Accept: application/media_control+xml,application/sdp,multipart/mixed
 Content-Type: application/sdp
 Content-Length: 201

 v=0
 o=BroadWorks 1195087044 1195087027 IN IP4 10.133.201.81
 s=-
 c=IN IP4 10.133.201.81
 t=0 0
 m=audio 21860 RTP/AVP 8 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15

 03/31/2014 11:46:55.170 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP
 UDP message to 10.133.201.81:[5060]:
 SIP/2.0 180 Ringing
 Date: Mon, 31 Mar 2014 10:46:55 GMT
 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
 SUBSCRIBE, NOTIFY, PUBLISH
 From: sip:+44266...@bnsplc.com;user=phone;tag=1c1195243906
 Allow-Events: presence
 Remote-Party-ID: sip:49900@10.168.8.76
 ;party=called;screen=yes;privacy=off
 Content-Length: 0
 To:  *404900 sip:44404900@
 *;tag=a4b020a1-c203-404a-ab13-da4aee5de9d8-56767054
 Contact: sip:44404900@10.168.8.76:5060
 Call-ID: 11951844583132014104653@10.133.201.81
 Via: SIP/2.0/UDP 10.133.201.81:5060;branch=z9hG4bKac1228556740
 CSeq: 2 INVITE

 03/31/2014 11:46:55.623 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP
 UDP message to 10.133.201.81:[5060]:
 SIP/2.0 200 OK
 Date: Mon, 31 Mar 2014 10:46:55 GMT
 From: sip:+44266...@bnsplc.com;user=phone;tag=1c1195243906
 Content-Length: 0
 To:  *404900 sip:44404900@**
 Call-ID: 11951844583132014104653@10.133.201.81
 Via: SIP/2.0/UDP 10.133.201.81:5060;branch=z9hG4bKac1228556740
 CSeq: 2 CANCEL

   On Monday, 31 March 2014, 13:31, zoltan.kele...@emerson.com 
 zoltan.kele...@emerson.com wrote:
   Hi Damian,

 A SIP debug on your CUBE can clarify which fields you are getting and then
 you could use voice class sip-profile to modify some sip headers on your
 CUBE to fix it.
 This assuming you have  a CUBE controlled by you between the CUCM cluster
 and the provider.

 Lacking that you could pull the SIP traces from the CUCM cluster and have
 your provider send the appropriate fields.

 Regards,

  *Zoltan Kelemen*
 Global Communications and Information Security
 Implementation Engineering
 w: +40 374 132356 | m: +40 757 039093

  *From:* cisco-voip 
 [mailto:cisco-voip-boun...@puck.nether.netcisco-voip-boun...@puck.nether.net]
 *On Behalf Of *Damian Turburville
 *Sent:* Monday, March 31, 2014 2:15 PM
 *To:* cisco-voip@puck.nether.net
 *Subject:* [cisco-voip] Incoming CLI Unknown but shows up at bottom of
 display

  Hi,
 Bit of a weird one here. Recently transferred PSTN services to a SIP
 carrier, so incoming calls are now coming through a SIP trunk from an SBC.
 Incoming

[cisco-voip] a one-off challenge

2014-04-04 Thread Ryan Huff
I need to interop some old Allied Telesis gear with CUCM; here is the
challenge:

I have an Allied Telesis iMG646MOD gateway (1 SM Fiber WAN, 6 fasteternet
and 4 FXS). I need to figure out a way to support the FXS ports using CUCM
9.X.

The AT gateway supports SIP and MGCP. Obviously, Cisco doesn't support the
MGCP stack on this kind of gear (unless there is a way to fake it), so I am
left with SIP.

The Syntax of the CLI for the AT gear is so foreign to me; I have the
manual for the gateway but in terms of documentation, it's rather poor.

Has anyone done this before / have an example configuration I can look at?

Thanks,

Ryan
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Re: [cisco-voip] CUWL 10.x licenses

2014-09-22 Thread Ryan Huff
If you're talking about CUCM and CUC; in the 9.x branch Cisco introduced the 
ELM (Enterprise License Manager). When you move into the 10.x's you have to use 
the PLM (Prime License Manager).

If your licensing is on the ELM and you want to goto the PLM (for 10.x) then 
you have to get fancy with TAC. They can deduct the amount of licensing that 
you need from the ELM and re-host it for the PLM (but you have to provide them 
with the license request from the PLM).

So lets say you have 200 CUWL Pro licenses on an ELM for UCOS 9.1.x and you 
want 100 of them to goto a PLM for UCOS 10.x. TAC can take your license from 
the ELM and re-host it minus 100 (leaving you with 100 CUWL Pro in the ELM), 
then TAC can take the extra 100 and re-host them into the PLM as CUWL Pro for 
UCOS 10.X.

If you used the PLM with 9.1.X and have upgraded UCOS to 10.X from 9.1.X then 
the licenses need to be re-hosted. Prove your upgrade entitlement to TAC and 
ask them to re-host the licenses.

Ryan Huff
CCNA R/S, CCNA Wireless, CCNA Voice, 
CCNP Voice, UC on UCSOS, UCCX Specialist, 
CCIE Collaboration (written)
From: peders...@bennettjones.com
To: mloradi...@heliontechnologies.com; cisco-voip@puck.nether.net
Date: Mon, 22 Sep 2014 21:12:34 +
Subject: Re: [cisco-voip] CUWL 10.x licenses









Interesting that Cisco switched to a manual process. I opened an SR and the guy 
from Licensing said I should contact The Product Manager whatever that means. 
Good thing I get 60 days to sort this out
J
 


From: Matthew Loraditch [mailto:mloradi...@heliontechnologies.com]


Sent: 22 September 2014 1:56 PM

To: Eric Pedersen; cisco-voip (cisco-voip@puck.nether.net)

Subject: RE: CUWL 10.x licenses


 
You submit a manual case for all major upgrades now. Just like for 8 to 9.
 

 
 
Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA
1965 Greenspring Drive

Timonium, MD 21093



direct voice. 443.541.1518

fax.  410.252.9284



Twitter  |  
Facebook  | Website  |  
Email Support
Support Phone. 410.252.8830
 

 


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Eric Pedersen

Sent: Monday, September 22, 2014 3:48 PM

To: cisco-voip (cisco-voip@puck.nether.net)

Subject: [cisco-voip] CUWL 10.x licenses


 
We upgraded to CUCX 10.5 from 9.1 on the weekend and the 9.x licenses installed 
on PLM need to be upgraded. Does anyone know how to get a PAK for this? I 
didn't receive any licenses in the upgrade e-delivery for CUCM and CUCX.
 
Thanks,
Eric
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Re: [cisco-voip] Delete Log Files

2014-09-23 Thread Ryan Huff
Martin,

If you have a space concern (trying to do an in-place upgrade  etc), you 
can adjust the High/Low logging watermarks  but remember to set them back 
after you're finished!

Thanks,

Ryan Huff

CCNA R/S, CCNA Wireless, 
CCNA Voice, CCNP Voice, 
UC on UCS Specialist, 
UCCX Product Specialist, 
CCIE Collaboration (Written)

Date: Tue, 23 Sep 2014 18:19:38 +0530
From: sknt...@gmail.com
To: m...@bilobit.com
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Delete Log Files

Martin,
Is there a reason you'd like to do this? What is the use case?By default, the 
older traces will get overwritten when there is no space left for newer 
traces.You can manually delete the files from the CLI or the Remote Browse on 
RTMT.

ThanksSreekanth
On 23 September 2014 15:43, Martin Schmuker m...@bilobit.com wrote:





Guys, 



is there any way to delete CUCM log files (aka traces) after x days?



Thanks,  Martin



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Re: [cisco-voip] MSP VoIP Monitoring

2014-09-25 Thread Ryan Huff
Ryan,

Prime is pretty sweet but if you're looking for qos tools (jitter stats, mos 
scores, delays ... etc) then I would look at two products:

1.) LiveAction . very qos centric
2.) NetBrains .. does do a lot with QOS but also does general layer2/3/4 
mapping

We're eval'ing NetBrains now and it looks pretty sweet. For instance, I can 
open a mapped Visio drawing of my topo that NetBrains created and I can hover 
on an interface and it will show any jitter/delays outside the threshold I 
define

From: dennis.h...@wwt.com
To: rburt...@gmail.com
Date: Thu, 25 Sep 2014 12:04:22 -0500
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] MSP VoIP Monitoring

Prime Collab with Assurance and Analytics would come to mind. I don’t have 
first-hand experience with that piece to the Prime Pie. Dennis Heim | 
Collaboration Solutions ArchitectWorld Wide Technology, Inc. | +1 314-212-1814  
From: Ryan Burtch [mailto:rburt...@gmail.com] 
Sent: Thursday, September 25, 2014 1:00 PM
To: Heim, Dennis
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] MSP VoIP Monitoring Dennis: We have VOSS and we use 
that to admin our HCS system. The thing we need is voice traffic monitoring for 
our on-prem customers in one system. This needs to give us the ability to see 
what is going on with the VoIP specific traffic. e.g. who has excessive jitter 
on their line, dropped calls, MOS scores, etc. Any Ideas?   Sincerely, Ryan 
Burtch On Thu, Sep 25, 2014 at 12:53 PM, Heim, Dennis dennis.h...@wwt.com 
wrote:There is also Voss-4-UC, which has RBAC and multi-tenancy.  Dennis Heim | 
Collaboration Solutions ArchitectWorld Wide Technology, Inc. | +1 314-212-1814  
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Burtch
Sent: Thursday, September 25, 2014 11:09 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] MSP VoIP Monitoring My company is looking for a VoIP 
monitoring solution that is Multi-Tennant capable. We manage several Customer's 
Voice environment and we need to be able to provide them statistics, 
troubleshooting, etc through a single pane of glass. This would be something 
like multi-tennant SolarWinds. Does anyone have any ideas?   Sincerely, Ryan 
Burtch 
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Re: [cisco-voip] Phone Configuration

2014-09-30 Thread Ryan Huff
The only truly supported call manager way of barging a connected call is barge 
or cbarge, but they both require a 'button push'. Now if your connected call is 
offnet, you could potentially do a PLAR on the client phone, so as soon as the 
off-hook event triggers on the client phone it dials into the offnet 
conference. If the connected call in onnet, then barge or cbarge are really 
your options.
 
If you want to venture into layer 2 of the OSI model however, you could do a 
SPAN on the ports of the phones that are doing the connected calls and then 
dump the RTP/signaling stream to the port of the client phone. Not sure if that 
would work or not, you'd just have to play with it a bit.
 
Thanks,
 
Ryan
 
From: nh...@co.fresno.ca.us
To: cisco-voip@puck.nether.net
Date: Tue, 30 Sep 2014 22:41:57 +
Subject: [cisco-voip] Phone Configuration









We use several Translations services, is there a way to configure a phone to 
join into a phone call without pushing anything?
Basically, our Employees call translation service and gives the account 
information, then gestures to the client to pick up the phone, at this point 
the client phone just joins into the conference without the customer pushing 
any buttons.
 
Is this possible?
 

 

Neal Haas
IT Analyst, Communications
 




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Re: [cisco-voip] masking internal caller ID?

2014-09-30 Thread Ryan Huff
If you're using a recent version of CCM, you could use a newer routing 
technique called route next hop by calling party number (particularly useful 
for call blocking at the cucm level).
 
Admittedly, a fair bit more work than a simple translation (you have to get 
creative with class of control elements) but it is a great excuse to use the 
feature and try something new. 
 
From: dennis.h...@wwt.com
To: bmead...@vt.edu; edward.country...@presencehealth.org
Date: Tue, 30 Sep 2014 21:35:01 -0500
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] masking internal caller ID?

The pesky mid-call update for xform patterns strikes again. As Brian mentioned, 
xlation patterns, or if you want to write something that uses the CURRI API ( 
Cisco unified Routing Rules Interface). Dennis Heim | Collaboration Solutions 
ArchitectWorld Wide Technology, Inc. | +1 314-212-1814  From: cisco-voip 
[mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Brian Meade
Sent: Tuesday, September 30, 2014 3:42 PM
To: Countryman, Edward
Cc: cisco-voip voyp list
Subject: Re: [cisco-voip] masking internal caller ID? That's expected behavior. 
 I would send the calls through a translation pattern to updating the calling 
number instead. On Tue, Sep 30, 2014 at 5:08 PM, Countryman, Edward 
edward.country...@presencehealth.org wrote:We need to “hide” or change the 
extension number of a small group of cisco phones when they are dialing 
internally (4 digits) to another phone. Our thought was to use a calling party 
transformation pattern for this, which appears to work fine when alerting and 
connected to the called number. However, the call history directory on the 
receiving phone (the one that see’s the transformed calling number not the real 
calling number) reflects the real number.   This doesn’t seem to make sense and 
defeats the whole purpose. what are we missing?? Do you know of a better way to 
approach this request? 
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Re: [cisco-voip] E1 CAS card in Mexico not getting enough digits

2014-11-16 Thread Ryan Huff
So on router B, do the following and then place an inbound call from a known 
number:

hostname# conf t
hostname (config)# logging console 
hostname (config)# exit
hostname# term mon 
hostname# debug isdn q931

What do you see for Calling Party Number i =, Plan: and Type:? The plan 
and type fields should be located below the calling party  number field. 
Once you've determined that you really are receiving 4 digits from the telco on 
that circuit the next step is to figure out what in the router config is 
stripping the digit. If you find that you are not receiving 4 digits from the 
telco, you can either work with the telco to fix it on the PRI or if the 
missing digit is a constant, you can add it with an inbound voice translation 
rule.

Look at your inbound dial peers; do you have any voice translation rules that 
are doing any digit stripping? Are the inbound voice translations using 
different regex than on the router that is working and if so, what is different?

Thanks,

Ryan Huff
CCIE Collaboration (Written), CCNP Voice, CCNA Voice
CCNA Route/Switch, CCNA Wireless, UCCX Specialist

From: jason.aar...@dimensiondata.com
To: cisco-voip@puck.nether.net
Date: Sun, 16 Nov 2014 08:18:22 +
Subject: [cisco-voip] E1 CAS card in Mexico not getting enough digits









I have a problem with a E1 CAS site in Mexico.  I have two routers. Router B is 
problem.  
 
In router A I see 4 digits come in on DTMF.  
In router B I see 3 digits come in on DTMF. We are missing 1 digit.
 
I swap the circuit and works fine in router A.  Same IOS both routers.
 
Controller e1 0/0/0
   Framing NO-CRC4
   dso-group 0 timeslots 1-10 type r2-digital dtmf dnis
   cas-custom 0
   Country telmex
   Seizure-ackt-time 2
 
 
 
 






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Re: [cisco-voip] E1 CAS card in Mexico not getting enough digits

2014-11-16 Thread Ryan Huff
Ahh, missed the part about CAS, sorry. 

So is the provider doing something on their side with the MAC of the E1 on your 
side? If the provider is sending 3 digits (regardless if it works on the other 
router) then it should be their issue.

You have the plan and type set according to what the provider expects?

Are you getting 3 digits for the calling or called party?

From: jason.aar...@dimensiondata.com
To: ryanh...@outlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits
Date: Sun, 16 Nov 2014 19:56:50 +









CAS no q931
 
Debug shows on bad router we receive 3 digits from provider, move circuit to 
another router we receive 4 digits from provider.
 


From: Ryan Huff [mailto:ryanh...@outlook.com]


Sent: Sunday, November 16, 2014 8:52 AM

To: Jason Aarons (AM); cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits


 
 

So on router B, do the following and then place an inbound call from a known 
number:



hostname# conf t

hostname (config)# logging console 

hostname (config)# exit

hostname# term mon 

hostname# debug isdn q931



What do you see for Calling Party Number i =, Plan: and Type:? The plan 
and type fields should be located below the calling party  number field. 
Once you've determined that you really are receiving 4 digits from the telco on 
that circuit the next
 step is to figure out what in the router config is stripping the digit. If you 
find that you are not receiving 4 digits from the telco, you can either work 
with the telco to fix it on the PRI or if the missing digit is a constant, you 
can add it with an inbound
 voice translation rule.



Look at your inbound dial peers; do you have any voice translation rules that 
are doing any digit stripping? Are the inbound voice translations using 
different regex than on the router that is working and if so, what is different?



Thanks,



Ryan Huff

CCIE Collaboration (Written), CCNP Voice, CCNA Voice

CCNA Route/Switch, CCNA Wireless, UCCX Specialist



From: 
jason.aar...@dimensiondata.com

To: cisco-voip@puck.nether.net

Date: Sun, 16 Nov 2014 08:18:22 +

Subject: [cisco-voip] E1 CAS card in Mexico not getting enough digits

I have a problem with a E1 CAS site in Mexico.  I have two routers. Router B is 
problem.  
 
In router A I see 4 digits come in on DTMF.  
In router B I see 3 digits come in on DTMF. We are missing 1 digit.
 
I swap the circuit and works fine in router A.  Same IOS both routers.
 
Controller e1 0/0/0
   Framing NO-CRC4
   dso-group 0 timeslots 1-10 type r2-digital dtmf dnis
   cas-custom 0
   Country telmex
   Seizure-ackt-time 2
 
 
 
 



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Re: [cisco-voip] E1 CAS card in Mexico not getting enough digits

2014-11-17 Thread Ryan Huff
Well it had to either be the Telco or an inbound xlate. Glad you found it! Good 
job, now go have an Iced Tea and a vacation!

From: jason.aar...@dimensiondata.com
To: avholloway+cisco-v...@gmail.com; ryanh...@outlook.com; 
cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits
Date: Mon, 17 Nov 2014 06:03:23 +









So after about 5 CCIEs looked at it and 5 Cisco AS onsite engineers and 2 days 
of effort we found the translation-profile on the voice-port was somehow 
responsible. 
 Seems the incoming digits took longer on router-b.  Go figure.  We move the 
translation-profile from the voice port to the dial-peer.
 
I hate CAS.
 
From: Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com]


Sent: Sunday, November 16, 2014 9:38 PM

To: Jason Aarons (AM); Ryan Huff; cisco-voip@puck.nether.net

Subject: Re: [cisco-voip] E1 CAS card in Mexico not getting enough digits
 
What if...just what if, it's alternating between 3 and 4 digits every time you 
unplug it?  Try unplugging it and re-plugging it into the same router to 
validate this crazy idea.

On Sun Nov 16 2014 at 7:21:11 PM Jason Aarons (AM) 
jason.aar...@dimensiondata.com wrote:



If you move circuit to another router and it works, then its hard to blame the 
carrier! 


 


CAS stinks.


 


 



Sent from my Verizon Wireless 4G LTE Smartphone




 

 Original message 


From: Ryan Huff 


Date:11/16/2014 15:45 (GMT-05:00) 


To: Jason Aarons (AM) , 
cisco-voip@puck.nether.net 


Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits



 



Ahh, missed the part about CAS, sorry.




So is the provider doing something on their side with the MAC of the E1 on your 
side? If the provider is sending 3 digits (regardless if it works on the other 
router) then it should be their issue.



You have the plan and type set according to what the provider expects?



Are you getting 3 digits for the calling or called party?



From: 
jason.aar...@dimensiondata.com

To: ryanh...@outlook.com;
cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits

Date: Sun, 16 Nov 2014 19:56:50 +

CAS no q931

 

Debug shows on bad router we receive 3 digits from provider, move circuit to 
another router we receive 4 digits from provider.

 



From: Ryan Huff [mailto:ryanh...@outlook.com]


Sent: Sunday, November 16, 2014 8:52 AM

To: Jason Aarons (AM); 
cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] E1 CAS card in Mexico not getting enough digits



 

 


So on router B, do the following and then place an inbound call from a known 
number:



hostname# conf t

hostname (config)# logging console 

hostname (config)# exit

hostname# term mon 

hostname# debug isdn q931



What do you see for Calling Party Number i =, Plan: and Type:? The plan 
and type fields should be located below the calling party  number field. 
Once you've determined that you really are receiving 4 digits from the telco on 
that circuit the next
 step is to figure out what in the router config is stripping the digit. If you 
find that you are not receiving 4 digits from the telco, you can either work 
with the telco to fix it on the PRI or if the missing digit is a constant, you 
can add it with an inbound
 voice translation rule.



Look at your inbound dial peers; do you have any voice translation rules that 
are doing any digit stripping? Are the inbound voice translations using 
different regex than on the router that is working and if so, what is different?



Thanks,



Ryan Huff

CCIE Collaboration (Written), CCNP Voice, CCNA Voice

CCNA Route/Switch, CCNA Wireless, UCCX Specialist




From: jason.aar...@dimensiondata.com

To: cisco-voip@puck.nether.net

Date: Sun, 16 Nov 2014 08:18:22 +

Subject: [cisco-voip] E1 CAS card in Mexico not getting enough digits


I have a problem with a E1 CAS site in Mexico.  I have two routers. Router B is 
problem.  

 

In router A I see 4 digits come in on DTMF.  

In router B I see 3 digits come in on DTMF. We are missing 1 digit.

 

I swap the circuit and works fine in router A.  Same IOS both routers.

 

Controller e1 0/0/0

   Framing NO-CRC4

   dso-group 0 timeslots 1-10 type r2-digital dtmf dnis

   cas-custom 0

   Country telmex

   Seizure-ackt-time 2

 

 

 

 




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Re: [cisco-voip] HELP

2014-11-17 Thread Ryan Huff
Well, you can't make calls because there are no resources available to 
negotiate the call, that much is clear. 

I
 assume this issue came about after you upgraded the IOS on the router? 
What platform do you have and what code are you currently running?

The
 firmware for the DSP is in the IOS code itself. As long as you're on 
supported code for the platform, activate the DSP Farm and then reboot 
the router. This should resolve the issue.

Thanks,

Ryan Huff
CCIE Collaboration (Written), CCNP Voice, CCNA Voice
CCNA R/S, CCNA Wireless
UCCX Specialist

Date: Mon, 17 Nov 2014 13:42:03 +0100
From: eteng.o...@unicem.com.ng
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] HELP

HELP,I CAN'T CALL OUT ON E1 AND FXO. SEE MESSAGE ON THE ROUTER.
voice-card 0 ! Warning! DSPs 5 in slot 0 are using non-default firmware from 
device flash: ! This is not recommended, the IOS default version is 24.3.3







If you are not the intended recipient and have received this e-mail in error, 
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Re: [cisco-voip] VMware 5.5

2014-11-17 Thread Ryan Huff
Martin,





Have you disabled Large Receive Offset (LRO) on all the Elastic Sky hosts? 
http://docwiki.cisco.com/wiki/Disable_LRO





Thanks,





Ryan Huff


CCIE Collaboration (Written), CCNP Voice, CCNA Voice,


CCNA Route  Switch, CCNA Wireless


UCCX Specialist

From: m...@bilobit.com
To: james.buchan...@gmail.com
Date: Mon, 17 Nov 2014 18:06:51 +
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] VMware 5.5









So 5.5 is supported with CUCM 9.1.x

 
So the question is: Why do I have CUCM at 100% CPU? Only hard reset helps at 
this point.
 
Thanks, Martin
 



From: James Buchanan [mailto:james.buchan...@gmail.com]


Sent: Monday, November 17, 2014 1:10 PM

To: Martin Schmuker

Cc: Cisco VoIP Mailing List

Subject: Re: [cisco-voip] VMware 5.5


 

Hello,



Whenever you are looking for answers on UC virtualization, look here: 
http://docwiki.cisco.com/wiki/Virtualization_for_Cisco_Unified_Communications_Manager_%28CUCM%29



Thanks,

James


 

On Mon, Nov 17, 2014 at 5:36 AM, Martin Schmuker m...@bilobit.com wrote:


Guys,
 
since 3 weeks we are running our UC Environment (CUCM, Unity Cxn, IMP) on ESXi 
5.5. At Friday we setup new vCenter 5.5 and added the esx hosts.
 
Since saturday, all machines are stuck in 100% CPU after a few hours. Sometimes 
all machines at the *same* time! They don’t reply Ping (ICMP Echo), and
 CPU is at 100%.
 
CUCM and Cxn are on 9.1(2)SU2a (9.1.2.12901-3).
 
Someone has any idea? Is 9.1.x not supported on vSphere 5.5?
 
Thanks, Martin




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Re: [cisco-voip] VMware 5.5

2014-11-17 Thread Ryan Huff
While that particular DocWiki does suggest that it is no longer needed to 
disable Large Receive Offset on ESXi 4.1 with CUCUM 8.6 and above; you should 
also note that one of the specific issues having LRO on can cause is consistent 
CPU pegging for the UCOS guests (wouldn't be the first time something didn't 
work the way Cisco's docs say it should).

Please also review: 
https://supportforums.cisco.com/document/95886/disable-lro-ucs-uc-application-deployments
 Again, I can't say beyond a shadow of doubt that this is the issue anymore 
than anyone can say that it isn't.

I would also look into the RAID drivers as well as all the physical connections 
on the UCS boxes.

Thanks,

rh
From: wo...@justfamily.org
Date: Mon, 17 Nov 2014 14:30:21 -0700
Subject: Re: [cisco-voip] VMware 5.5
To: ryanh...@outlook.com
CC: m...@bilobit.com; james.buchan...@gmail.com; cisco-voip@puck.nether.net

That page specifically says you don't have to disable LRO if you are above 4.1 
esxi with8.6 CUCM.
Martin, which version of 5.5 are you on?  GA, update 1 or 2?  I recently ran 
into an issue with a customer on 5.5 u1 and too new of raid drivers on the card 
and had to downgrade the drivers.  Wasn't causing 100% cpu, but very slow drive 
access times.
So checked your vmware version and compatibility on the hardware you are 
running.
On Mon, Nov 17, 2014 at 11:59 AM, Ryan Huff ryanh...@outlook.com wrote:



Martin,





Have you disabled Large Receive Offset (LRO) on all the Elastic Sky hosts? 
http://docwiki.cisco.com/wiki/Disable_LRO





Thanks,





Ryan Huff


CCIE Collaboration (Written), CCNP Voice, CCNA Voice,


CCNA Route  Switch, CCNA Wireless


UCCX Specialist

From: m...@bilobit.com
To: james.buchan...@gmail.com
Date: Mon, 17 Nov 2014 18:06:51 +
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] VMware 5.5









So 5.5 is supported with CUCM 9.1.x


 

So the question is: Why do I have CUCM at 100% CPU? Only hard reset helps at 
this point.

 

Thanks, Martin

 




From: James Buchanan [mailto:james.buchan...@gmail.com]


Sent: Monday, November 17, 2014 1:10 PM

To: Martin Schmuker

Cc: Cisco VoIP Mailing List

Subject: Re: [cisco-voip] VMware 5.5



 


Hello,



Whenever you are looking for answers on UC virtualization, look here: 
http://docwiki.cisco.com/wiki/Virtualization_for_Cisco_Unified_Communications_Manager_%28CUCM%29



Thanks,

James



 


On Mon, Nov 17, 2014 at 5:36 AM, Martin Schmuker m...@bilobit.com wrote:



Guys,

 

since 3 weeks we are running our UC Environment (CUCM, Unity Cxn, IMP) on ESXi 
5.5. At Friday we setup new vCenter 5.5 and added the esx hosts.

 

Since saturday, all machines are stuck in 100% CPU after a few hours. Sometimes 
all machines at the *same* time! They don’t reply Ping (ICMP Echo), and
 CPU is at 100%.

 

CUCM and Cxn are on 9.1(2)SU2a (9.1.2.12901-3).

 

Someone has any idea? Is 9.1.x not supported on vSphere 5.5?

 

Thanks, Martin





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Re: [cisco-voip] VMware 5.5

2014-11-18 Thread Ryan Huff
I had a CPU pegging issue about 4.5 months ago on ESXi 5.0. I turned LRO off 
(only change I made), and I haven't had an issue with CPU pegging since. 

My issue was due to the TCP throughput being irregular on VMXNET2 and VMXNET3 
and it was causing HELLO/ACK issues with the database replication (it would get 
into a setup loop and never stop). Guest reboots didn't fix it, if a rebooted 
the UCS box it worked for a little bit and then went right back to the issue

VMware states that the issue is known in Linux kernel 2.6.24 and later with 
VMXNET3 
(http://kb.vmware.com/selfservice/microsites/search.do?language=en_UScmd=displayKCexternalId=1027511)

There are plenty of reports on the Internet were the LRO/CUC issue has been 
found beyond esxi 4.1 and cucm 8.6.

RHEL probably isn't going to get any performance gains from LRO anyhow so if it 
were me, I'd turn it off. It's broke now, right? Generally, you can't break 
broke so I would try it, if for no other reason than to say that isn't it.

Again, I am not saying this IS your issue, just that it COULD BE a contributing 
factor, it would be worth a shot in my book if you haven't found the issue yet.


From: wo...@justfamily.org
Date: Mon, 17 Nov 2014 23:50:52 -0700
Subject: Re: [cisco-voip] VMware 5.5
To: ryanh...@outlook.com
CC: m...@bilobit.com; james.buchan...@gmail.com; cisco-voip@puck.nether.net

This discussion comes up about every 6 months on this list, LRO is no longer 
affected and was only a 4.1 issue on esxi.
http://puck.nether.net/pipermail/cisco-voip/2012-October/029907.html 
http://docwiki.cisco.com/wiki/Unified_Communications_VMware_Requirements#Supported_Versions.2C_Patches_and_Updates_of_VMware_vSphere_ESXi
 

The latter being definitive for me, they keep that doc updated pretty well and 
note, it states 4.1, not any other version of esxi.
Martin has another issue.  :)
On Mon, Nov 17, 2014 at 4:46 PM, Ryan Huff ryanh...@outlook.com wrote:



While that particular DocWiki does suggest that it is no longer needed to 
disable Large Receive Offset on ESXi 4.1 with CUCUM 8.6 and above; you should 
also note that one of the specific issues having LRO on can cause is consistent 
CPU pegging for the UCOS guests (wouldn't be the first time something didn't 
work the way Cisco's docs say it should).

Please also review: 
https://supportforums.cisco.com/document/95886/disable-lro-ucs-uc-application-deployments
 Again, I can't say beyond a shadow of doubt that this is the issue anymore 
than anyone can say that it isn't.

I would also look into the RAID drivers as well as all the physical connections 
on the UCS boxes.

Thanks,

rh
From: wo...@justfamily.org
Date: Mon, 17 Nov 2014 14:30:21 -0700
Subject: Re: [cisco-voip] VMware 5.5
To: ryanh...@outlook.com
CC: m...@bilobit.com; james.buchan...@gmail.com; cisco-voip@puck.nether.net

That page specifically says you don't have to disable LRO if you are above 4.1 
esxi with8.6 CUCM.
Martin, which version of 5.5 are you on?  GA, update 1 or 2?  I recently ran 
into an issue with a customer on 5.5 u1 and too new of raid drivers on the card 
and had to downgrade the drivers.  Wasn't causing 100% cpu, but very slow drive 
access times.
So checked your vmware version and compatibility on the hardware you are 
running.
On Mon, Nov 17, 2014 at 11:59 AM, Ryan Huff ryanh...@outlook.com wrote:



Martin,





Have you disabled Large Receive Offset (LRO) on all the Elastic Sky hosts? 
http://docwiki.cisco.com/wiki/Disable_LRO





Thanks,





Ryan Huff


CCIE Collaboration (Written), CCNP Voice, CCNA Voice,


CCNA Route  Switch, CCNA Wireless


UCCX Specialist

From: m...@bilobit.com
To: james.buchan...@gmail.com
Date: Mon, 17 Nov 2014 18:06:51 +
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] VMware 5.5









So 5.5 is supported with CUCM 9.1.x


 

So the question is: Why do I have CUCM at 100% CPU? Only hard reset helps at 
this point.

 

Thanks, Martin

 




From: James Buchanan [mailto:james.buchan...@gmail.com]


Sent: Monday, November 17, 2014 1:10 PM

To: Martin Schmuker

Cc: Cisco VoIP Mailing List

Subject: Re: [cisco-voip] VMware 5.5



 


Hello,



Whenever you are looking for answers on UC virtualization, look here: 
http://docwiki.cisco.com/wiki/Virtualization_for_Cisco_Unified_Communications_Manager_%28CUCM%29



Thanks,

James



 


On Mon, Nov 17, 2014 at 5:36 AM, Martin Schmuker m...@bilobit.com wrote:



Guys,

 

since 3 weeks we are running our UC Environment (CUCM, Unity Cxn, IMP) on ESXi 
5.5. At Friday we setup new vCenter 5.5 and added the esx hosts.

 

Since saturday, all machines are stuck in 100% CPU after a few hours. Sometimes 
all machines at the *same* time! They don’t reply Ping (ICMP Echo), and
 CPU is at 100%.

 

CUCM and Cxn are on 9.1(2)SU2a (9.1.2.12901-3).

 

Someone has any idea? Is 9.1.x not supported on vSphere 5.5?

 

Thanks, Martin





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Re: [cisco-voip] CUCM 10.5.1 adding subscriber rebooted phones?

2014-11-21 Thread Ryan Huff
What version of CUCM?

From: jason.aar...@dimensiondata.com
To: cisco-voip@puck.nether.net
Date: Fri, 21 Nov 2014 17:46:14 +
Subject: [cisco-voip] CUCM 10.5.1 adding subscriber rebooted phones?






I had a coworker tell me he added a subscriber during the day and all phones 
reset during the day. 



Perhaps this was ITL update?



 He wasn't expecting all phone in cluster to reset until he added new sub to a 
callmanger group after hours.



Any one else seen this? Is it expected? 







Sent from my Verizon Wireless 4G LTE Smartphone




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Re: [cisco-voip] CUCM 10.5.1 adding subscriber rebooted phones?

2014-11-21 Thread Ryan Huff
Never mind  just saw the subject line!

So when the phones reset, did any of them register to the new subscriber or was 
it just a phone reset? If it isn't in the CM group then no phones should have 
registered.

Is this 10.5(1) or SU1 or SU2?

An ITL reload sounds plausible.

From: ryanh...@outlook.com
To: jason.aar...@dimensiondata.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] CUCM 10.5.1 adding subscriber rebooted phones?
Date: Fri, 21 Nov 2014 13:00:40 -0500




What version of CUCM?

From: jason.aar...@dimensiondata.com
To: cisco-voip@puck.nether.net
Date: Fri, 21 Nov 2014 17:46:14 +
Subject: [cisco-voip] CUCM 10.5.1 adding subscriber rebooted phones?






I had a coworker tell me he added a subscriber during the day and all phones 
reset during the day. 



Perhaps this was ITL update?



 He wasn't expecting all phone in cluster to reset until he added new sub to a 
callmanger group after hours.



Any one else seen this? Is it expected? 







Sent from my Verizon Wireless 4G LTE Smartphone




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Re: [cisco-voip] AD Sync

2014-11-26 Thread Ryan Huff
What version of CUCM and What version of AD?

Has the AD sync ever worked correctly? 

Is this a break/fix?What changed from when it was working?Is the distinguished 
user the same, did that user's AD permissions changeDoes the distinguished user 
have the delegate control privilege on the domain?Is this a new install?Were 
any changes made to AD after the original full sync the first time?Has either 
the domain name of the CUCM cluster or the AD server changed since the first 
time the LDAP full sync was ran?Does the BIND authentication work correctly?
Have you completely removed an existing user account and then re-synced from AD 
to see if that account re-appears?



Date: Wed, 26 Nov 2014 16:09:51 +0530
From: sknt...@gmail.com
To: shabbar_babraw...@hotmail.com
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] AD Sync

Hi Shabbar,

What is the CUCM version?
So the users go into Inactive mode every 6 hours? Or once everyday? If once, 
what time does that happen and is that during a sync?

Have you taken a look at the DirSync logs during the period of failure?
What about a packet capture to see if this could be an issue due to the network?

Thanks
Sreekanth

On 26 November 2014 at 11:37, shabbar babrawala shabbar_babraw...@hotmail.com 
wrote:



Hi
Have a strange problem where the sync with AD has broken , everyday morning we 
have to keep performing a full sync as the users show inactive even though the 
setting is to sync every 6 hours. Have even deleted the LDAP configuration and 
redone but no luck. any help is appreciated.
Shabbar   

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Re: [cisco-voip] AD Sync

2014-11-26 Thread Ryan Huff
Shabbar,

Sounds like the LDAP Manger Distinguished User or the LDAP search base is 
acting strangely. Have you made any OU/permisison changes in AD? 

Try using a new/different distinguished manager user in the LDAP directory 
configuration. Also, are you using any custom filters?

Thanks,

Ryan

From: shabbar_babraw...@hotmail.com
To: ryanh...@outlook.com; sknt...@gmail.com
CC: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] AD Sync
Date: Wed, 26 Nov 2014 14:42:30 +




Hi
Cucm 9.1 win 2012
It was working before on win 2003 broken after upgrade to 2012
Regards
Shabbar

From: ryanh...@outlook.com
To: sknt...@gmail.com; shabbar_babraw...@hotmail.com
CC: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] AD Sync
Date: Wed, 26 Nov 2014 09:38:59 -0500




What version of CUCM and What version of AD?

Has the AD sync ever worked correctly? 

Is this a break/fix?What changed from when it was working?Is the distinguished 
user the same, did that user's AD permissions changeDoes the distinguished user 
have the delegate control privilege on the domain?Is this a new install?Were 
any changes made to AD after the original full sync the first time?Has either 
the domain name of the CUCM cluster or the AD server changed since the first 
time the LDAP full sync was ran?Does the BIND authentication work correctly?
Have you completely removed an existing user account and then re-synced from AD 
to see if that account re-appears?



Date: Wed, 26 Nov 2014 16:09:51 +0530
From: sknt...@gmail.com
To: shabbar_babraw...@hotmail.com
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] AD Sync

Hi Shabbar,

What is the CUCM version?
So the users go into Inactive mode every 6 hours? Or once everyday? If once, 
what time does that happen and is that during a sync?

Have you taken a look at the DirSync logs during the period of failure?
What about a packet capture to see if this could be an issue due to the network?

Thanks
Sreekanth

On 26 November 2014 at 11:37, shabbar babrawala shabbar_babraw...@hotmail.com 
wrote:



Hi
Have a strange problem where the sync with AD has broken , everyday morning we 
have to keep performing a full sync as the users show inactive even though the 
setting is to sync every 6 hours. Have even deleted the LDAP configuration and 
redone but no luck. any help is appreciated.
Shabbar   

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Re: [cisco-voip] IMP external database front-end

2014-11-26 Thread Ryan Huff
Hi Charles,

I got a super simple front end written in PHP running on the same LAMP stack 
the the PostGreSQL database is running on (benefit of being a former 
programmer). It isn't 100% complete but will give you a real-time flow of 
persistent chat messages and p 2 p messages. The code is logical that someone 
with basic programming skill could follow along and finish out the programming.

Interested?

Thanks,

Ryan

From: wo...@justfamily.org
Date: Wed, 26 Nov 2014 09:06:53 -0700
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] IMP external database front-end

Before I go down the road of re-creating the wheel, has anyone written a nice 
front end to the postgresql database for simple reports for a customer?
While pgadmin will do the trick, a non-technical person will struggle with 
getting reports out as needed.
I'm thinking about a simple search for person(s), date/time range, probably PHP 
would be easiest since if you installed the phppgadmin on a linux install, you 
have php and apache already there, or maybe on a 2nd box for management 
purposes if it's a big install.
I'm not a coder, but I can probably hack something together, would be a lot 
easier if someone else has something to share :)
Thanks!Happy and safe Holidays to everyoneCharles

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Re: [cisco-voip] Jabber contact disappears consistently

2014-11-26 Thread Ryan Huff
I call BS! 

I am running a fully AD integrated 9.1.2 CUCM with IMPresence 9.1.2 and I have 
the Cisco Jabber for Windows 10.5 client deployed and I have URI Dialing 
running in the Jabber client, ALL. DAY. LONG and it works flawlessly.

In the deployment docs, it says to use EnableSIPUriDialing in the 
jabber-config.xml HOWEVER; if you look at the PRT logs generated from one of 
the 10.5 Jabber clients I bet you it is looking for the EnableSIPURIDialing 
node and not the EnableSIPUriDialing node.

Notice the casing difference in URI Vs. Uri . I couldn't get URI in the 
client for weeks, got the same junk answer from TAC as well  then I 
troll'ed the PRT logs and found that to be my issue, a flippin' casing issue.



Date: Wed, 26 Nov 2014 09:09:50 -0600
From: pav.c...@gmail.com
To: leslie.me...@lvs1.com
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Jabber contact disappears consistently

Just to close this out.

Official word from Cisco even though every piece of Jabber doc says its 
supported in J4W 10.5 
==


Currently Directory URI is only supported on the
server.  The client has not yet been updated to fully support this. 
So no, Jabber 10.5 does NOT support directory URI

==


On Sat, Nov 22, 2014 at 2:15 PM, Leslie Meade leslie.me...@lvs1.com wrote:








I also have seen the same issue, and put everyone onto the same server
 





Leslie Meade 









..


Mobile:778.228.4339 |
Main: 604.676.5239

Email:
leslie.me...@lvs1.com

 


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Jason Aarons (AM)

Sent: Saturday, November 22, 2014 11:33 AM

To: Josh Warcop; Pavan K; cisco-voip@puck.nether.net

Subject: Re: [cisco-voip] Jabber contact disappears consistently


 
I complete agree, put all Jabberusers on single server.  Saw the same bugs in 
10x.
 


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Josh Warcop

Sent: Friday, November 21, 2014 3:48 PM

To: Pavan K; cisco-voip@puck.nether.net

Subject: Re: [cisco-voip] Jabber contact disappears consistently


 
 

Do to the bugs I've seen I have been pinning all users to a single node in the 
cluster and just having HA failover.

 




Date: Fri, 21 Nov 2014 11:31:39 -0600

From: pav.c...@gmail.com

To: cisco-voip@puck.nether.net

Subject: [cisco-voip] Jabber contact disappears consistently
We have an interesting problem on a new jabber deployment that has us stumped. 
Wonder if anybody else saw this.

Two jabber servers with ha enabled and balanced users. Using 10.5su1 for jabber 
windows and ucm/imp.

Leveraging sip directory uri as the IM scheme due to a multi forest environment 
with duplicate Samaccountnames across domains.

UserA contact list has userB on it. Folks can im each other without any 
problem. If we move userB from his imp server to another server in the same 
subcluster, userB disappears from userA's contact list.

Repeatable across multiple users with different userA and userB and happens 
every time regardless of moving them from server1 to server2 or vice versa.

Using router to router communication between nodes and default jabber-config.

Any ideas ?

Have a TAC case open but its going nowhere.






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itevomcid 





-- 
- Pavan


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Re: [cisco-voip] Extension Mobility login is unavailable (23)

2014-11-27 Thread Ryan Huff
Is the phone subscribed to the Extension Mobility Service?
Is Extension Mobility capabilities enabled on the phone that the user is trying 
to log into? 

From: leslie.me...@lvs1.com
To: cisco-voip@puck.nether.net
Date: Thu, 27 Nov 2014 16:21:59 +
Subject: [cisco-voip] Extension Mobility login is unavailable (23)









Client is getting this error when they try to use EM.
The error tells me that it is to do with EMCC and they are not configured for 
this, any ideas ?
 
I have restarted the EM service in the cluster and it was working before hand.
 
Cheers
 
Leslie
 




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Re: [cisco-voip] Extension Mobility login is unavailable (23)

2014-11-27 Thread Ryan Huff
Any database replication issues at the moment?
Are the phones still able to resolve the address of the TFTP server at the 
moment (either by IP address or hostname)
Is the URL that you are specifying in the Extension Mobility Service still 
valid and resolvable by the phones?

Thanks,

Ryan
From: leslie.me...@lvs1.com
To: ryanh...@outlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Extension Mobility login is unavailable (23)
Date: Thu, 27 Nov 2014 16:31:57 +









Yes to all. It is not just one site. I have tried other phones in the cluster 
and it is also happening to them as well.
 
 
 


From: Ryan Huff [mailto:ryanh...@outlook.com]


Sent: Thursday, November 27, 2014 8:31 AM

To: Leslie Meade; cisco-voip cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] Extension Mobility login is unavailable (23)


 

Is the phone subscribed to the Extension Mobility Service?

Is Extension Mobility capabilities enabled on the phone that the user is trying 
to log into?





From:
leslie.me...@lvs1.com

To: cisco-voip@puck.nether.net

Date: Thu, 27 Nov 2014 16:21:59 +

Subject: [cisco-voip] Extension Mobility login is unavailable (23)

Client is getting this error when they try to use EM.
The error tells me that it is to do with EMCC and they are not configured for 
this, any ideas ?
 
I have restarted the EM service in the cluster and it was working before hand.
 
Cheers
 
Leslie
 



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[cisco-voip] Cisco Presence External Database Front End Application

2014-12-01 Thread Ryan Huff
I have had a few requests from folks for help with a front-end GUI application 
for the external PostgreSQL database that can be used with the Cisco Presence 
and IM server.

Using something like PgAdmin is a great tool to use in an administrative 
function but not very user friendly for the non tech. I have written a PHP 
based application (developed on a LAMP stack). That is a great front-end GUI 
for basic functionality. The application has 2 of the 4 features finished and 
can be a great learning tool or a head start to writing/expanding your own app.

Please download at: http://ryanthomashuff.com/downloads/

Let me know if you have any questions/need assistance getting it set up.

Thanks,

Ryan Huff
ryanthomashuff.com
CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist
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Re: [cisco-voip] Jabber 10.5.1 for Windows shows In a Meeting when he isn't

2014-12-02 Thread Ryan Huff
Is this local client integration or is the CPIM server integrated with an 
on-prem Exchange server?

Thanks,

Ryan Huff
ryanthomashuff.com
CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist


From: jason.aar...@dimensiondata.com
To: cisco-voip@puck.nether.net
Date: Tue, 2 Dec 2014 16:31:49 +
Subject: [cisco-voip] Jabber 10.5.1 for Windows shows In a Meeting when he  
isn't









We have a user who’s Jabber for Windows 10.5.1 client shows “In a Meeting” when 
his calendar is clear. I looked in outlook if he has multiple calendars which 
he did not, when he logs into Jabber his status shows as “available” for around
 15 seconds before changing to “In a meeting”. If I clear the local Jabber CSF 
folders it still displays the same behavior.
 
IMP 10.5.1
 






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Re: [cisco-voip] 7941 SIP Endpoints Rejected - CUCM 10.5

2014-12-05 Thread Ryan Huff
Is the phone's config pre-built on CCM or is the phone trying to auto register?

Thanks,

Ryan Huff
ryanthomashuff.com
CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist


Date: Thu, 4 Dec 2014 20:49:50 -0500
From: joel.dav...@gmail.com
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] 7941 SIP Endpoints Rejected - CUCM 10.5

Trying to get some 7941 handsets registered as SIP endpoints for a customer 
that is coming back via a s2s IPSEC VPN tunnel. We are able to see the traffic 
come across and the validated it did upgrade but the current phone we are 
testing with just shows as rejected. Besides pulling a pcap on the 
subscriber/tftp nodes anyone else run into this or seen similar issues? There 
is firewall traversal but SIP ALG and protocol inspection has been disabled.
-- 
Joel Davila




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Re: [cisco-voip] Replace 7940 with 7942

2014-12-17 Thread Ryan Huff
Add a new phone device using the 7942 device type. Then build out the device 
config line for line compared with the 7940 config. Pay attention the phone 
template/softkey templates ... etc.

Thanks,

Ryan Huff
ryanthomashuff.com
CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist


Date: Wed, 17 Dec 2014 09:10:56 -0500
From: dzh...@gmail.com
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Replace 7940 with 7942

I have a two line 7940 setup but the internal mic is not working.  SO i bought 
a 7942 replacement.  If I were doing a 7940 for 7940 replacement, I would just 
login to CM to do a MAC Address change.  What is the best/easiest way to do 
this since I am replacing it with a 7942?

Thanks.


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Re: [cisco-voip] Call track

2014-12-18 Thread Ryan Huff
Are these recent calls or historical?
Do you currently have any type of call accounting system where your CDR's are 
currently sent?

If these are recent calls you can start with the Real Time Data in the RTMT.

Thanks,

Ryan Huff
ryanthomashuff.com
CCNA R/S, CCNA Wireless, CCNP Voice, UCCX Specialist


Date: Fri, 19 Dec 2014 00:12:32 +0530
From: dharambi...@gmail.com
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Call track

Hi Guys, I am using CUCM 7 cluster and with multi VG MGCP at different 
location. My Telecom provider has given list of unauthorized calls made from 
us. So where we can investigate in cucm.I collected CDR a nd cucm trace thert 
are showing some unsuccessful call attempts. Is there any strong methodology to 
collect all reports  
-- 
 Regards,
 Dharambir Kumar
  
  




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[cisco-voip] 10.5 Prime License Manager HA Deployment

2014-12-26 Thread Ryan Huff
Looking to deploy two PLM servers in an HA fashion. I assume that I just need 
to add a reference to the second PLM in the CCM Server section like a typical 
CCM subscriber. Is that correct?

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[cisco-voip] 10.5 BAT Error/Question

2014-12-26 Thread Ryan Huff
I think I know what the issue is but can't seem to verify ...

- 4 node 10.5 Cluster with DNS enabled.
- Currently, I have a known DNS resolution problem in the cluster, I know that 
and am working on it.

What currently happens is I'll upload a file in BAT then run a job (like 
import) against the file I uploaded and the Job Scheduler log comes back with a 
'success' result but N/A items processed. When I open the log I get Error; 
cannot reference /./filename.tar.

I'm guessing that BAT is using the DNS resolver to reference node/path/file and 
since DNS is bunk right now, it can't resolve the node. I can't find in the 
Googles and docs where it definitively says BAT uses DNS when enabled.

Is that my issue or should I be looking elsewhere?

Thanks,

Ryan
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Re: [cisco-voip] 10.5 BAT Error/Question

2014-12-26 Thread Ryan Huff
Interesting you mention that, cause I am trying to import the exports from a 
lower version

Thanks,

Ryan Huff

Date: Fri, 26 Dec 2014 19:17:07 +0100
Subject: Re: [cisco-voip] 10.5 BAT Error/Question
From: florian.kroessbac...@gmail.com
To: ryanh...@outlook.com
CC: cisco-voip@puck.nether.net

Hy
we have some Problems with BAT on 10.5, but the Proble is, that there are Null 
Values in the file. In 8.6 we can export, change something or add something and 
then reimport that.In 10.5 this isn't working (cause of Null Values in the 
Export)--Florian Kroessbacher 
florian.kroessbac...@gmail.com   

2014-12-26 18:18 GMT+01:00 Ryan Huff ryanh...@outlook.com:



I think I know what the issue is but can't seem to verify ...

- 4 node 10.5 Cluster with DNS enabled.
- Currently, I have a known DNS resolution problem in the cluster, I know that 
and am working on it.

What currently happens is I'll upload a file in BAT then run a job (like 
import) against the file I uploaded and the Job Scheduler log comes back with a 
'success' result but N/A items processed. When I open the log I get Error; 
cannot reference /./filename.tar.

I'm guessing that BAT is using the DNS resolver to reference node/path/file and 
since DNS is bunk right now, it can't resolve the node. I can't find in the 
Googles and docs where it definitively says BAT uses DNS when enabled.

Is that my issue or should I be looking elsewhere?

Thanks,

Ryan
  

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[cisco-voip] Device pack installation methodology question

2015-01-03 Thread Ryan Huff
I need to install a device pack on a 2 node 9.1(2) CCM cluster to get support 
for some 88xx phones but I do not want to update the loads for anything else.

The approach I am going to use is:

Drop the publisher out of the CM Group, forcing all phones to the subscriber. 
Install the device pack on the publisher and reboot the publisher. Once the 
publisher is backup, set all the device defaults back to what I want them to be 
then add the publisher back to the CM Group. Then drop the subscriber from the 
CM Group forcing all the phones on the publisher and start the install process 
over for the subscriber. Once everything is back up add the subscriber back to 
the CM Group.

Does that sound reasonable or is there an easier way?

Thanks,

Ryan
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Re: [cisco-voip] Device pack installation methodology question

2015-01-03 Thread Ryan Huff
Correct, ordinarily that would be what I would do as well. In this case though, 
I am trying to avoid upgrading the firmware on anything, other than the devices 
I am trying to get support for.

Thanks,

Ryan

Date: Sat, 3 Jan 2015 14:20:43 -0600
From: b...@brezworks.com
To: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Device pack installation methodology question


  

  
  
On 1/3/2015 2:11 PM, Ryan Huff wrote:



  
  I need to install a device pack on a 2 node 9.1(2)
CCM cluster to get support for some 88xx phones but I do not
want to update the loads for anything else.



The approach I am going to use is:



Drop the publisher out of the CM Group, forcing all phones to
the subscriber. Install the device pack on the publisher and
reboot the publisher. Once the publisher is backup, set all the
device defaults back to what I want them to be then add the
publisher back to the CM Group. Then drop the subscriber from
the CM Group forcing all the phones on the publisher and start
the install process over for the subscriber. Once everything is
back up add the subscriber back to the CM Group.



Does that sound reasonable or is there an easier way?

  



Any time I've needed device support I just install it on all the
nodes then reboot them one at a time during a maintenance window. 
As long as both nodes are running call processing, they'll fail over
when the node reboots.  Not sure about on 9.X, but on 10.X this just
shows as a blip on the phones of them reregistering, not a full
reboot.  Obviously if you install new software for those phones,
they'll upgrade software when they reboot.



Jeremy TheBrez Bresley

b...@brezworks.com

  


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Re: [cisco-voip] CUCMC 10.5.1 to CXN 10.5.1 with SCCP number of rings 4 before answer

2015-02-02 Thread Ryan Huff
Is it one phone or all phones?

It sounds like an interdigit timeout or extended hunt; I know you said there 
are no overlaps though. Could be on the connections side too if it is system 
wide.

Anything change with the CSS/Partion of the ports? Distribution algorithm of 
the line group change? I assume all the SCCP ports are still registered.

What does RTMT say about connection's health, CPU spike?

Thanks,

Ryan

From: jason.aar...@dimensiondata.com
To: cisco-voip@puck.nether.net
Date: Mon, 2 Feb 2015 17:52:56 +
Subject: [cisco-voip] CUCMC 10.5.1 to CXN 10.5.1 with SCCP number of rings 4
before answer









Customer states that it’s taking too long for Unity Connection to answer when 
he presses the Messages button.  It’s SCCP integrated.  The Hunt Pilot looks 
good, no duplicate or unassigned DNs, no overlapping number range.  Been in 
production
 and was working fine previously.
 
I did notice the checkbox in the Hunt List “For Voice Mail Usage” is unchecked. 
  After hours we can check that.
 
 
CUCMC 10.5.1.1-7
 
CXN 10.5.1.11900-13
 
Jason Aarons 
Consultant
Dimension Data
904-338-3245 mobile
 
 






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Re: [cisco-voip] Web directory from CUCM

2015-01-15 Thread Ryan Huff
Terry,

I am literally moving to Georgia today and tomorrow! I'll save this email  
look for something from me early next week.

It's all my code so I can and will share, no problem.

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Re: [cisco-voip] Web directory from CUCM

2015-01-15 Thread Ryan Huff
I have always made mine in the past. I use PHP for my scripting needs.

If you have a way to run PHP, I can share - just ping me.
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[cisco-voip] A new web GUI for Cisco Call Manager

2015-01-22 Thread Ryan Huff
Hello all,
I thought I'd share a PHP based web GUI for the Corporate Directory feature of 
Cisco Call Manager. I made it a while ago and have seen a few recent requests 
for something like it, so I thought I put it out here.
There is a README in the ZIP archive with all the particulars. It is a basic 
design with a little CSS (Cascading Style Sheet) so you'll probably want to 
change the interface around a little to fit your needs, but all the backend 
code is solid.
http://ryanthomashuff.com/2015/01/new-web-gui-for-call-manager-corporate-directory/

Thanks,

Ryan
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Re: [cisco-voip] Unity connection - Voicemail playback

2015-02-16 Thread Ryan Huff
Jose,

If SIP integrated, easy fix, just point the trunk at the pub (and then reset 
the trunk in CUCM).

If SCCP integrated, you have a few different options and the best option 
depends on how you're setup and if this is a perm. change or just temporary?

If this is just temporary and assuming you have the max amount of ports per CUC 
node, In CUCM I would go to the line group of the SCCP ports and drop out the 
ports that are registered to the CUC subscriber node. Once your link is no 
longer saturated, add the ports back to the line group.

There are some other design strategies that you can look at to help mitigate 
this in the future. For example, you could add an additional CUC Subscriber 
node in the same DC as the publisher (so you end up with a subscriber in both 
data centers). This would allow you to have additional ports in the same DC as 
the publisher.


Thanks,

Ryan

From: chrw...@cisco.com
To: jcolon...@gmail.com; cisco-voip@puck.nether.net
Date: Mon, 16 Feb 2015 18:52:43 +
Subject: Re: [cisco-voip] Unity connection - Voicemail playback









You have to route calls to that node. Right now, either you SCCP ports, or SIP 
trunk are pointing to the Sub. You have to change the priority of the trunk or
 the ports so it uses the Sub first.
 
+Chris
TME - Unity Connection and MediaSense
 
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Jose Colon II

Sent: Monday, February 16, 2015 1:44 PM

To: Cisco VOIP

Subject: [cisco-voip] Unity connection - Voicemail playback
 

I have a Unity 10.5 HA setup and the subscriber is at our COLO. Currently the 
link between the two is being saturated so voicemail playback is very choppy. 

 


The question is, how do i force the publisher to handle those requests so that 
voicemail playback is handled inside this location and not at the colo?


 


Thanks


Jose






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Re: [cisco-voip] CCNA/CCNP Collaboration certs

2015-02-17 Thread Ryan Huff
Dennis,

The conversion tool shows that CIPTV2 is the only test that CCNP Voice needs to 
convert to CCNP Collaboration. This is really new info though so it could 
change.

As of this writing the CCNP Voice retirement date is listed as Q4 2015 and 
PearsonVue doesn't even have 300-075 (CIPTV2) listed as a schedue-lable test 
yet.

From what I understand, the CIPTV2 test covers video related material. Given 
that the CCIE Collaboration is based on the 9.x stack I assume the topics will 
cover things like Collaboration Edge, Jabber Video, Telepresence ... etc

Thanks,

Ryan

From: dennis.h...@wwt.com
To: ryanh...@outlook.com; ben.st...@gmail.com; wo...@justfamily.org
CC: cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] CCNA/CCNP Collaboration certs
Date: Tue, 17 Feb 2015 18:50:05 +









What exam is required to convert CCNP voice to CCNP collaboration?
 

Dennis Heim | Emerging Technology Architect (Collaboration)
World Wide Technology, Inc. | +1 314-212-1814


Innovation happens on project squared --
http://www.projectsquared.com
 
Click here to join me in my Collaboration Meeting Room
 
 

 


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Ryan Huff

Sent: Tuesday, February 17, 2015 11:58 AM

To: Ben Story; Charles Goldsmith

Cc: voip puck

Subject: Re: [cisco-voip] CCNA/CCNP Collaboration certs


 

What gets me is that a CCNP Voice needs one test to upgrade to CCNP 
Collaboration 
(http://www.cisco.com/web/learning/tools/ccnp_collab/ccnp_collab_tool.html).




Really? Only one test's worth of info is what changes between Voice and 
Collaboration?



If you are going to retire and invalidate a credential and move to another 
tract altogether, then it should be a top-down revamp, not just one test of a 
few new questions about video.



This feels like it is nothing more than re branding/marketing; and I shouldn't 
have to pay for that.



My grumpy two cents,



Ryan








Date: Tue, 17 Feb 2015 10:10:24 -0600

From: ben.st...@gmail.com

To: wo...@justfamily.org

CC: cisco-voip@puck.nether.net

Subject: Re: [cisco-voip] CCNA/CCNP Collaboration certs

Is it just me or does it seem weird for a CCNA level to now require two tests.  
Money grab?











--
Ben Story 
CCNP, CCNA, CCNA Wireless, CCDA
ben.st...@gmail.com


@ntwrk80
http://showbrain.blogspot.com
http://rand0mw0rds.blogspot.com

 
 
From sour-faced saints and silly devotions, good Lord, preserve us!. -- St.
 Teresa of Avila








 

On Tue, Feb 17, 2015 at 9:58 AM, Charles Goldsmith wo...@justfamily.org wrote:


https://learningnetwork.cisco.com/community/ccna-ccnp-collaboration 



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Re: [cisco-voip] CCNA/CCNP Collaboration certs

2015-02-17 Thread Ryan Huff
What gets me is that a CCNP Voice needs one test to upgrade to CCNP 
Collaboration 
(http://www.cisco.com/web/learning/tools/ccnp_collab/ccnp_collab_tool.html). 

Really? Only one test's worth of info is what changes between Voice and 
Collaboration?

If you are going to retire and invalidate a credential and move to another 
tract altogether, then it should be a top-down revamp, not just one test of a 
few new questions about video.

This feels like it is nothing more than re branding/marketing; and I shouldn't 
have to pay for that.

My grumpy two cents,

Ryan


Date: Tue, 17 Feb 2015 10:10:24 -0600
From: ben.st...@gmail.com
To: wo...@justfamily.org
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CCNA/CCNP Collaboration certs

Is it just me or does it seem weird for a CCNA level to now require two tests.  
Money grab?--Ben Story CCNP, CCNA, CCNA Wireless, ccdaben.st...@gmail.com
@ntwrk80http://showbrain.blogspot.comhttp://rand0mw0rds.blogspot.com


From sour-faced saints and silly devotions, good Lord, preserve us!. -- St. 
Teresa of Avila

On Tue, Feb 17, 2015 at 9:58 AM, Charles Goldsmith wo...@justfamily.org wrote:
https://learningnetwork.cisco.com/community/ccna-ccnp-collaboration 


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[cisco-voip] Scratching my head, need a different set of eyes please cucm 10.5 ...

2015-01-29 Thread Ryan Huff
CUCM 10.5 (New Build with DNS)

Testing out phone registration with IP Communicator 8.6.4. Phone auto registers 
just fine and is assigned all the auto registered defaults I have specified 
(DN, partition, CSS ... etc) and the IP communicator is able to resolve all the 
URLs by DNS. Everything appears to be fine.

However, the Services menu keeps poping up. I haven't configured any services 
beyond the shipped services so the service menu is blank. I hit the exit button 
on the services menu and it goes away for a second or two, then it comes right 
back.

The phone can still process digits, I can dial, I can get dialtone ... etc  
everything seems to work but the services menu. I feel like I have dealt with 
this before but I can't recall what the solution was.

Thanks,

Ryan
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Re: [cisco-voip] Scratching my head, need a different set of eyes please cucm 10.5 ...

2015-01-29 Thread Ryan Huff
Winner winner chicken dinner! Thank you, that was driving me nuts. Learning all 
the new things in 10 ... coming from the 8/9 world doesn't seem that long ago 
but in Cisco UC land I might as well be a dinosaur. 

Thanks,

Ryan

Date: Thu, 29 Jan 2015 11:03:22 -0600
Subject: Re: [cisco-voip] Scratching my head, need a different set of eyes 
please cucm 10.5 ...
From: ad...@adman.net
To: ryanh...@outlook.com
CC: cisco-voip@puck.nether.net

The default Universal Device Template includes an idle URL that tries to access 
the self-provisioning IP Phone Service (/cucm-uds/xps/selfProvision). I'm 
guessing this is what you're seeing.

On Thu, Jan 29, 2015 at 10:52 AM, Ryan Huff ryanh...@outlook.com wrote:



CUCM 10.5 (New Build with DNS)

Testing out phone registration with IP Communicator 8.6.4. Phone auto registers 
just fine and is assigned all the auto registered defaults I have specified 
(DN, partition, CSS ... etc) and the IP communicator is able to resolve all the 
URLs by DNS. Everything appears to be fine.

However, the Services menu keeps poping up. I haven't configured any services 
beyond the shipped services so the service menu is blank. I hit the exit button 
on the services menu and it goes away for a second or two, then it comes right 
back.

The phone can still process digits, I can dial, I can get dialtone ... etc  
everything seems to work but the services menu. I feel like I have dealt with 
this before but I can't recall what the solution was.

Thanks,

Ryan
  

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Re: [cisco-voip] Scratching my head, need a different set of eyes please cucm 10.5 ...

2015-01-29 Thread Ryan Huff
So I just changed the default phone template in the Cisco IP Communicator 
Device Default to solve this particular issue. Thanks for the bug id reference 
Brian, that helps!

Thanks,

Ryan


Date: Thu, 29 Jan 2015 12:16:35 -0500
Subject: Re: [cisco-voip] Scratching my head, need a different set of eyes 
please cucm 10.5 ...
From: bmead...@vt.edu
To: ad...@adman.net
CC: ryanh...@outlook.com; cisco-voip@puck.nether.net

Also changing it there will remove it for new IP Phones added but you need to 
go into any phone that already auto-registered and remove the idle URL on the 
device configuration.
This feature only works on phones that support SBD/TVS since it works over 
HTTPS only.  That's why it doesn't work with IP communicator.  I opened a doc 
bug on this- https://tools.cisco.com/bugsearch/bug/CSCun13382
Brian
On Thu, Jan 29, 2015 at 12:03 PM, Adam Blomfield ad...@adman.net wrote:
The default Universal Device Template includes an idle URL that tries to access 
the self-provisioning IP Phone Service (/cucm-uds/xps/selfProvision). I'm 
guessing this is what you're seeing.

On Thu, Jan 29, 2015 at 10:52 AM, Ryan Huff ryanh...@outlook.com wrote:



CUCM 10.5 (New Build with DNS)

Testing out phone registration with IP Communicator 8.6.4. Phone auto registers 
just fine and is assigned all the auto registered defaults I have specified 
(DN, partition, CSS ... etc) and the IP communicator is able to resolve all the 
URLs by DNS. Everything appears to be fine.

However, the Services menu keeps poping up. I haven't configured any services 
beyond the shipped services so the service menu is blank. I hit the exit button 
on the services menu and it goes away for a second or two, then it comes right 
back.

The phone can still process digits, I can dial, I can get dialtone ... etc  
everything seems to work but the services menu. I feel like I have dealt with 
this before but I can't recall what the solution was.

Thanks,

Ryan
  

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[cisco-voip] BE7K Unity Connections 5,000 User

2015-01-30 Thread Ryan Huff
Looks like the DocWiki says I can run Unity Connections 5,000 user on the 
BE7K-K9 SKU but states an OVA mod needed for 8.x and 9.x. I am guessing the 
same holds true in 10.5?

Thanks,

Ryan
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Re: [cisco-voip] Device pack installation methodology question

2015-01-06 Thread Ryan Huff
I like that approach - seems simpler, does mean running the phones without 
valid tftp for a few (ringtones, phone desktops, corp. directory ...) but 
that's what maintenance windows are for.

Thanks for your time!

Thanks,

Ryan

From: avholloway+cisco-v...@gmail.com
Date: Sat, 3 Jan 2015 20:58:09 +
Subject: Re: [cisco-voip] Device pack installation methodology question
To: ryanh...@outlook.com; cisco-voip@puck.nether.net

I just did this recently too, to add support for an 8831 on an 8x cluster.

Trying to recall from memory, here's what I recall doing:
1. Deactivate TFTP on both nodes2. BAT Export the Device Defaults3. Install the 
Dev Pack on Pub4. Restart Pub5. Install Dev Pack on Sub6. Restart Sub7. BAT 
Import the Device Defaults8. Activate TFTP on both nodes9. Add new 8831's
For step 7, I also did a BAT export of device defaults post dev pack, and ran a 
diff on the two.  I saw the changes to the existing phone firmware as well as 
the addition of the new phone models.  When you import the old device defaults 
back in at this stage, note that the absence of the new phone models simply 
tells CUCM BAT process to ignore them and leave them alone, while reverting the 
firmware changes on all of the existing phones.
On Sat Jan 03 2015 at 2:15:25 PM Ryan Huff ryanh...@outlook.com wrote:



I need to install a device pack on a 2 node 9.1(2) CCM cluster to get support 
for some 88xx phones but I do not want to update the loads for anything else.

The approach I am going to use is:

Drop the publisher out of the CM Group, forcing all phones to the subscriber. 
Install the device pack on the publisher and reboot the publisher. Once the 
publisher is backup, set all the device defaults back to what I want them to be 
then add the publisher back to the CM Group. Then drop the subscriber from the 
CM Group forcing all the phones on the publisher and start the install process 
over for the subscriber. Once everything is back up add the subscriber back to 
the CM Group.

Does that sound reasonable or is there an easier way?

Thanks,

Ryan
  
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[cisco-voip] Where is the Unified Attendant Console ISO download?

2015-01-10 Thread Ryan Huff
Hello,

I can find the OVA to download in the portal but I can't seem to find the 
actual ISO. Any clues?

Thanks,

Ryan
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[cisco-voip] Quick question about Unified Attendant Console

2015-01-10 Thread Ryan Huff
Not really familiar with installing U-AC  from what I am gathering (version 
10.5); it is still Windows dependant, correct? Use the OVA to create the 
machine that you install Windows on, then use the AC binary to install.

Is that right?

Thanks,

Ryan
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[cisco-voip] Advanced Attendant Console 10.5, which Win OS?

2015-01-12 Thread Ryan Huff
Anyone know where I can find which Win OS is required for the Advanced 
Attendant Console Server 10.5?

Thanks,

Ryan Huff
r...@ryanthomashuff.com
http://www.ryanthomashuff.com

CCNA R/S, CCNA Wireless, CCNA Voice
CCNP Voice, CCIE Collaboration (Written)
UCCX Specialist
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Re: [cisco-voip] Intercluster Trunks

2015-01-09 Thread Ryan Huff
How are you doing your media termination? Which side of them trunk is 
terminating?

Thanks,

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[cisco-voip] Unity Connections - COBRAS import and a question

2015-01-06 Thread Ryan Huff
So I have no issue using COBRAS to import vm accounts, call handlers and such 
...

How about some of the more obscure elements though, like restriction tables, 
COS ... etc. Does all that need to be hand rebuilt from the MS Access backup or 
is their an easier way that I'm just missing

Thanks,

Ryan Huff

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Re: [cisco-voip] CUCILYNC Voicemail login issues

2015-01-13 Thread Ryan Huff
Another thing to look at / consider is client DNS issues (if the client is 
trying to resolve the directory server by hostname). Which, should be evident 
in the p-caps.

Thanks,

Ryan


From: jason.aar...@dimensiondata.com
To: ryanh...@outlook.com; george.hend...@l-3com.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] CUCILYNC Voicemail login issues
Date: Tue, 13 Jan 2015 14:46:50 +









Capture the packets via Wireshark/network.
 


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Ryan Huff

Sent: Tuesday, January 13, 2015 9:19 AM

To: george.hend...@l-3com.com; cisco-voip@puck.nether.net

Subject: Re: [cisco-voip] CUCILYNC Voicemail login issues


 
 

Have you reconfirmed all your integration credentials? You might not get a 
directory error because the bind request might not even be making it to your 
directory server.



Did anything change about the directory server (ip address, bind account 
credentials, password expire ... etc)? Can the clients still talk to the 
directory server over the required ports?



Thanks,



Ryan




From: 
george.hend...@l-3com.com

To: cisco-voip@puck.nether.net

Date: Tue, 13 Jan 2015 14:03:18 +

Subject: [cisco-voip] CUCILYNC Voicemail login issues

Hey Guys,
 
  I’m seeing an issue where I am getting “invalid username or password” when 
trying to authenticate in CUCILYNC for Voicemail, but I don’t get an error for 
the Directory services.  However, I can login to the Cisco PCA page just fine
 with my credentials.  Nothing has changed with the voicemail system and all 
the registry entries are correct for pointing to the voicemail system for 
CUCILYNC.  Any ideas what could cause this?
 
Thanks,
Bill



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Re: [cisco-voip] CUCILYNC Voicemail login issues

2015-01-13 Thread Ryan Huff
You can have it either way (depending on your configuration), whatever is being 
advertised to your client for directory auth is what your client is trying to 
contact. I would fire up wireshark as suggested and see what is going on in 
packet land, could just be a silly network issue.

Thanks,

Ryan


From: george.hend...@l-3com.com
To: ryanh...@outlook.com; jason.aar...@dimensiondata.com; 
cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] CUCILYNC Voicemail login issues
Date: Tue, 13 Jan 2015 14:54:12 +









Does the CUCILYNC client go straight to LDAP server (the one set in the 
registry) for Voicemail authentication or to the voicemail server?
 
Service account for Unity Connection in AD is fine. 

 


From: Ryan Huff [mailto:ryanh...@outlook.com]


Sent: Tuesday, January 13, 2015 9:51 AM

To: Jason Aarons AM; Hendrix, George (Bill) @ NSS; cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] CUCILYNC Voicemail login issues


 

Another thing to look at / consider is client DNS issues (if the client is 
trying to resolve the directory server by hostname). Which, should be evident 
in the
 p-caps.



Thanks,



Ryan








From:
jason.aar...@dimensiondata.com

To: ryanh...@outlook.com; 
george.hend...@l-3com.com; cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] CUCILYNC Voicemail login issues

Date: Tue, 13 Jan 2015 14:46:50 +

Capture the packets via Wireshark/network.
 


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of Ryan Huff

Sent: Tuesday, January 13, 2015 9:19 AM

To: george.hend...@l-3com.com;
cisco-voip@puck.nether.net

Subject: Re: [cisco-voip] CUCILYNC Voicemail login issues


 
 

Have you reconfirmed all your integration credentials? You might not get a 
directory error because the bind request might not even be making it to your 
directory server.



Did anything change about the directory server (ip address, bind account 
credentials, password expire ... etc)? Can the clients still talk to the 
directory server over the required ports?



Thanks,



Ryan




From: 
george.hend...@l-3com.com

To: cisco-voip@puck.nether.net

Date: Tue, 13 Jan 2015 14:03:18 +

Subject: [cisco-voip] CUCILYNC Voicemail login issues

Hey Guys,
 
  I’m seeing an issue where I am getting “invalid username or password” when 
trying to authenticate in CUCILYNC for Voicemail, but I don’t get an error for 
the Directory services.  However,
 I can login to the Cisco PCA page just fine with my credentials.  Nothing has 
changed with the voicemail system and all the registry entries are correct for 
pointing to the voicemail system for CUCILYNC.  Any ideas what could cause this?
 
Thanks,
Bill



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[cisco-voip] Cisco Unity Connections 10.5 Intersite Networking Methodology

2015-01-09 Thread Ryan Huff
Can someone pass me some me some links/info on Intersite/Intrasite networking? 
I know what they are are ... etc but I have still have some 
methodology/strategy questions.
 
Like why would I consider using it? What advantages do I get by linking to CUC 
clusters together , what can I do with it ... etc

Thanks,

Ryan
 
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[cisco-voip] LDAP Search Filter Syntax reference

2015-01-12 Thread Ryan Huff
Does anyone have any links / info for LDAP Search Syntax? I am trying to write 
this filter:
 
Sync all children in ABC OUExcept for children in XYZ OUANDAll children of ABC 
OU WHERE IpPhone IS NOT null

Thanks,

Ryan Huff
r...@ryanthomashuff.com
http://www.ryanthomashuff.com

CCNA R/S, CCNA Wireless, CCNA Voice
CCNP Voice, CCIE Collaboration (Written)
UCCX Specialist
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[cisco-voip] Strange routing behavior cucm 10.5 and int'l pattern

2015-02-13 Thread Ryan Huff
Try this one on;

Was working fine ...

Standard Int'l route pattern 9.011!# Discard set to PreDot (again, this was 
working ... no issue on gateway ... etc)

So today it stops working, just rings busy. I debug the ISDN and it shows 
called party as the last 7 digits. I go over to DNA and use an int'l pattern 
with the css I was using and it blocks pattern for unallocation.

I create a new test partition with a new 9.011!# pattern in it and a new css 
with only the new partition in it. I go back over to DNA and try the int'l 
pattern with the new test css and it blocks for unallocated.

Now I scratch my head, so I take off the octothorpe on the pattern (9.011!) and 
BOOM, DNA routes and everything is happy. I move to production and it works 
just fine without the octothorpe.

What does this sound like? Do you think I may have competing patterns somewhere 
in the dial plan?

Thanks,

Ryan
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Re: [cisco-voip] Strange routing behavior cucm 10.5 and int'l pattern

2015-02-13 Thread Ryan Huff
Andrew,

Since international numbers are varying lengths, the # is used to signal the 
end of the number string.

You can also just wait for the interdigit timeout to expire and then routing 
occurs.

Usually as Brian mentioned, both are supported because it usually comes down to 
a user preference.

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Re: [cisco-voip] Strange routing behavior cucm 10.5 and int'l pattern

2015-02-13 Thread Ryan Huff
Brian,

I suppose I wasn't clear on that. The octothorpe was dialed at the end of the 
international pattern in production and with DNA.

Only when I removed the octothorpe from the pattern did routing occur.

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Re: [cisco-voip] CallerID not showing on 8941

2015-01-08 Thread Ryan Huff
What version of CCM Leslie? 
 
If you're on 8.x+ You could try to create a 'Route next hop by calling party 
number translation and prefix the + in the on the xlate pattern with the 
anticipated CLID and see if you can catch it. If you prove that it is the + 
then you can translate/transform and remove.
 
Again, this would be a stop-gap work around.

Thanks,

Ryan Huff
r...@ryanthomashuff.com
http://www.ryanthomashuff.com

CCNA R/S, CCNA Wireless, CCNA Voice
CCNP Voice, CCIE Collaboration (Written)
UCCX Specialist

 
Date: Thu, 8 Jan 2015 11:24:04 -0500
From: bmead...@vt.edu
To: leslie.me...@lvs1.com
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] CallerID not showing on 8941

Probably has to do with the special character.  Here was the only bug I found 
related- https://tools.cisco.com/bugsearch/bug/CSCuo91983
Might be caused by the same root issue but might need a new bug opened.
On Thu, Jan 8, 2015 at 11:12 AM, Leslie Meade leslie.me...@lvs1.com wrote:








CallerID from the PSTN to H323 is correct. Sip trunk to CUCM is correct. CUCM 
to 8841 shows correct. But is it not displaying on the screen.
 
Same call to another model is fine. 
 
I suspect the issue lies with the “±” in front of the Displayname ?
 
16301690.001 |10:27:08.571 |AppInfo  |localizeCgpn: StationSIPCdfc on device 
SEP6CFA89720901 , CSS = ,useDevicePoolCgpnCss =1 
AlternateCgpn(global)=705264 cgpn=705264
16301690.002 |10:27:08.571 |AppInfo  |SIPStationCdfc:CcSetupReq - 
unicodeConnectedUnicodeDisplayName='' asciiConnectedDisplayName='±TRUE NORTH 
TRAN'
16301690.003 |10:27:08.571 |AppInfo  |SIPStationCdfc:CcSetupReq - 
unicodeCallingPartyName='' asciiCallingPartyName='±TRUE NORTH TRAN' 
callingParty='705264'
 
Has anyone seen something like this before ?
 
Leslie
 
 




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Re: [cisco-voip] Extension that hangs up on the user?

2015-03-18 Thread Ryan Huff
So is CUC just so you can use the after action hang up technique or are you 
playing a greeting first?

If your just hanging up and not playing a greeting, could you just catch the 
ingress call on a translation then just dump the call (or play the reorder 
tone)?

Thanks,

Ryan

 Original Message 
From: James Andrewartha jandrewar...@ccgs.wa.edu.au
Sent: Wednesday, March 18, 2015 02:34 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Extension that hangs up on the user?

Hi list,

Is there a way in CUCM to make an extension that hangs up on the other
end? Currently we have a Unity Connection AA that does that, but it's
literally the only thing CUC is being used for and I want to get rid of
it. Currently we have our AAs (and voicemail) in Exchange 2007, which is
being upgraded to 2013 soon, but as far as I can tell there's no way to
have it hang up on the caller, so I transfer to the AA in Unity.

Thanks,

-- 
James Andrewartha
Network  Projects Engineer
Christ Church Grammar School
Claremont, Western Australia
Ph. (08) 9442 1757
Mob. 0424 160 877
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Re: [cisco-voip] Migration strategy

2015-03-18 Thread Ryan Huff
Any elements that you are going to pre-stage that are a part of the dial plan 
(translations, route patterns, DNs, transformations ...etc) wilk all need to be 
isolated from your currently migrated phones.

So the DNs on your device profiles would likely need to be in an isolated 
partition, but the device profile itself probably doesn't need to be in an  
isolated device pool.

If you do this, remember to use a CLLI or some unique descriptor in the 
description field of all your pre-staged elements, it will make it very easy 
for BAT to find things later on.

Thanks,

Ryan

 Original Message 
From: 秀王 kiwi.vo...@gmail.com
Sent: Tuesday, March 17, 2015 11:32 PM
To: Ryan Huff ryanh...@outlook.com
Subject: Re: [cisco-voip] Migration strategy
CC: cisco-voip@puck.nether.net

Hi Ryan,

let's say my phones are on a temp partition not reachable by other CSS. My
UDP (user device profile) are on a valid partition shared by others ( Ie.
P_Internal) but they are not logged in anywhere.

Will this confused the CUCM? Or i shall place the UDP on temp partition as
well. If so, can BAT assist me in migrating from TEMP partition to actual
partition (P_Internal)?

Cheers,
Ki Wi

On Wed, Mar 18, 2015 at 9:26 AM, Ryan Huff ryanh...@outlook.com wrote:

 I'm not, sure I completely understand your questions but I'll attempt to
 answer based on my understanding.

 Yes, you can pre-config devices and users in CCM prior to migration. If
 they are Cisco IP phones, you'll need the MAC address and model of the
 phone at a minimum.

 If they are non Cisco IP phones, you'll need to pre configure 3rd party
 sip devices (which is a different license requirement than a Cisco IP
 phone).

 Place the preconfigured dial plan that isnt migrated yet (on CCM), in a
 temp. partition that the already migrated phones cannot access. As you
 migrate, change that partition using BAT, to the correct partition for the
 portion of phones you migrated.

 Thanks,

 Ryan


  Original Message 
 From: 秀王 kiwi.vo...@gmail.com
 Sent: Tuesday, March 17, 2015 09:15 PM
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] Migration strategy

 Currently the client have avaya and cisco linked together using SIP.

 Cisco UCM cluster have users in the production environment.

 I'm are going to cutover more sites from avaya to cisco. Is it possible to
 preconfigure the users, extension number (let's say 87XXX range), phones
 and the user device profiles in advance?

 I'm thinking that if I preconfigure those information, the cucm will think
 that those extension number (87XXX) are local and unregistered.

 Is there a way to make CUCM thinks that in order to reach 87XXX range, it
 will still reach out to Avaya using the SIP trunk? Is there any setting in
 the route pattern can do that?

 I thinking that CUCM will always find a more exact match locally instead
 of through other source like translation pattern or route pattern.


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Re: [cisco-voip] Transcoding question

2015-03-16 Thread Ryan Huff
Thanks Dainel! I probably knew that at some point,  but I couldn't remember for 
the life of me!  Makes total sense.

Thanks,

Ryan

 Original Message 
From: Daniel Pagan dpa...@fidelus.com
Sent: Monday, March 16, 2015 09:16 PM
To: Daniel Pagan dpa...@fidelus.com,Ryan Huff 
ryanh...@outlook.com,cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Transcoding question

For clarity, by “higher bandwidth codec” I meant to say higher bit-rate codec, 
or codec of higher bandwidth consumption.

- Dan

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
Daniel Pagan
Sent: Monday, March 16, 2015 9:08 PM
To: Ryan Huff; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Transcoding question

Hey Ryan how’s it going? Transcoder allocated by CUCM comes from the side 
using a higher bandwidth codec, regardless if it’s the calling or called 
party, with the intention to avoid streaming a high bandwidth consuming codec 
over a WAN connection – keeping it local to the LAN. Of course, this isn’t 
always true, such as due to a local transcoding resource being entirely 
nonexistent or a misconfiguration of the MRG/MRGLs.

Hope this helps answer your question.

- Dan

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Monday, March 16, 2015 8:11 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Transcoding question


When xcoding is required in the call setup, which side is transcoded? The 
called party or the calling party?

Thanks
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[cisco-voip] Transcoding question

2015-03-16 Thread Ryan Huff
When xcoding is required in the call setup, which side is transcoded? The 
called party or the calling party?

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Re: [cisco-voip] Extension that hangs up on the user?

2015-03-18 Thread Ryan Huff
So if the CUC AA IS NOT playing a greeting and the AA is doing NOTHING but the 
after action of hang-up; you could just create a translation pattern that 
matches the called number (presumably, the called number is currently a CTI 
route point/DN that is forwarding to CUC, you would need to remove it before 
creating the translation). 

In the translation pattern, set block this pattern, call rejected.

You can also explore CCM ANI based call blocking; I discuss it here. 
http://ryanthomashuff.com/2014/11/call-blocking-by-caller-id/

Thanks,

Ryan

 From: jandrewar...@ccgs.wa.edu.au
 To: ryanh...@outlook.com; cisco-voip@puck.nether.net
 Date: Wed, 18 Mar 2015 22:10:36 +0800
 Subject: RE: [cisco-voip] Extension that hangs up on the user?
 
 Yeah, CUC has Callers Hear: Nothing then After Greeting/Call Action: Hang Up. 
 How would I dump the call? Select Route Option/Block this pattern: Call 
 Rejected in the translation pattern?
 
 Thanks,
 
 James Andrewartha
 Network  Projects Engineer
 Christ Church Grammar School
 Claremont, Western Australia
 Ph. (08) 9442 1757
 Mob. 0424 160 877
 
 From: Ryan Huff [ryanh...@outlook.com]
 Sent: Wednesday, 18 March 2015 8:12 PM
 To: James Andrewartha; cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] Extension that hangs up on the user?
 
 So is CUC just so you can use the after action hang up technique or are you 
 playing a greeting first?
 
 If your just hanging up and not playing a greeting, could you just catch the 
 ingress call on a translation then just dump the call (or play the reorder 
 tone)?
 
 Thanks,
 
 Ryan
 
 
  Original Message 
 From: James Andrewartha jandrewar...@ccgs.wa.edu.au
 Sent: Wednesday, March 18, 2015 02:34 AM
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] Extension that hangs up on the user?
 
 
 Hi list,
 
 Is there a way in CUCM to make an extension that hangs up on the other
 end? Currently we have a Unity Connection AA that does that, but it's
 literally the only thing CUC is being used for and I want to get rid of
 it. Currently we have our AAs (and voicemail) in Exchange 2007, which is
 being upgraded to 2013 soon, but as far as I can tell there's no way to
 have it hang up on the caller, so I transfer to the AA in Unity.
 
 Thanks,
 
 --
 James Andrewartha
 Network  Projects Engineer
 Christ Church Grammar School
 Claremont, Western Australia
 Ph. (08) 9442 1757
 Mob. 0424 160 877
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Re: [cisco-voip] Extension that hangs up on the user?

2015-03-18 Thread Ryan Huff
Very true. 

Just leave it unallocated and CCM doesn't have to do anything, but the caller 
will get annunciation.

Thanks,

Ryan

 Original Message 
From: Daniel Pagan dpa...@fidelus.com
Sent: Wednesday, March 18, 2015 11:17 AM
To: Ryan Huff ryanh...@outlook.com,James Andrewartha 
jandrewar...@ccgs.wa.edu.au,cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Extension that hangs up on the user?

The main question I have... if CUC is being used simply to hang-up on the 
calling party, what's the purpose of needing this migrated to CUCM instead of 
simply leaving the number unallocated? Correct me if I'm wrong, but it seems 
to me that you're specifically looking for a method in CUCM where the call is 
answered and then disconnected.

Is this true? Are you hoping to have the call actually connected before the 
disconnect? Or does a simple rejection of the call work fine for you? Only 
situation I can think of where this is needed would be not wanting callers to 
hear a rejection or error recording due to unallocated number.

- Dan

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Wednesday, March 18, 2015 10:23 AM
To: James Andrewartha; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Extension that hangs up on the user?

So if the CUC AA IS NOT playing a greeting and the AA is doing NOTHING but the 
after action of hang-up; you could just create a translation pattern that 
matches the called number (presumably, the called number is currently a CTI 
route point/DN that is forwarding to CUC, you would need to remove it before 
creating the translation).

In the translation pattern, set block this pattern, call rejected.

You can also explore CCM ANI based call blocking; I discuss it here. 
http://ryanthomashuff.com/2014/11/call-blocking-by-caller-id/

Thanks,

Ryan
 From: jandrewar...@ccgs.wa.edu.aumailto:jandrewar...@ccgs.wa.edu.au
 To: ryanh...@outlook.commailto:ryanh...@outlook.com; 
 cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
 Date: Wed, 18 Mar 2015 22:10:36 +0800
 Subject: RE: [cisco-voip] Extension that hangs up on the user?

 Yeah, CUC has Callers Hear: Nothing then After Greeting/Call Action: Hang 
 Up. How would I dump the call? Select Route Option/Block this pattern: Call 
 Rejected in the translation pattern?

 Thanks,

 James Andrewartha
 Network  Projects Engineer
 Christ Church Grammar School
 Claremont, Western Australia
 Ph. (08) 9442 1757
 Mob. 0424 160 877
 
 From: Ryan Huff [ryanh...@outlook.com]
 Sent: Wednesday, 18 March 2015 8:12 PM
 To: James Andrewartha; 
 cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
 Subject: Re: [cisco-voip] Extension that hangs up on the user?

 So is CUC just so you can use the after action hang up technique or are you 
 playing a greeting first?

 If your just hanging up and not playing a greeting, could you just catch the 
 ingress call on a translation then just dump the call (or play the reorder 
 tone)?

 Thanks,

 Ryan


  Original Message 
 From: James Andrewartha 
 jandrewar...@ccgs.wa.edu.aumailto:jandrewar...@ccgs.wa.edu.au
 Sent: Wednesday, March 18, 2015 02:34 AM
 To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
 Subject: [cisco-voip] Extension that hangs up on the user?


 Hi list,

 Is there a way in CUCM to make an extension that hangs up on the other
 end? Currently we have a Unity Connection AA that does that, but it's
 literally the only thing CUC is being used for and I want to get rid of
 it. Currently we have our AAs (and voicemail) in Exchange 2007, which is
 being upgraded to 2013 soon, but as far as I can tell there's no way to
 have it hang up on the caller, so I transfer to the AA in Unity.

 Thanks,

 --
 James Andrewartha
 Network  Projects Engineer
 Christ Church Grammar School
 Claremont, Western Australia
 Ph. (08) 9442 1757
 Mob. 0424 160 877
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Re: [cisco-voip] Migration strategy

2015-03-17 Thread Ryan Huff
I'm not, sure I completely understand your questions but I'll attempt to answer 
based on my understanding.

Yes, you can pre-config devices and users in CCM prior to migration. If they 
are Cisco IP phones, you'll need the MAC address and model of the phone at a 
minimum.

If they are non Cisco IP phones, you'll need to pre configure 3rd party sip 
devices (which is a different license requirement than a Cisco IP phone).

Place the preconfigured dial plan that isnt migrated yet (on CCM), in a temp. 
partition that the already migrated phones cannot access. As you migrate, 
change that partition using BAT, to the correct partition for the portion of 
phones you migrated.

Thanks,

Ryan

 Original Message 
From: 秀王 kiwi.vo...@gmail.com
Sent: Tuesday, March 17, 2015 09:15 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Migration strategy

Currently the client have avaya and cisco linked together using SIP.

Cisco UCM cluster have users in the production environment.

I'm are going to cutover more sites from avaya to cisco. Is it possible to
preconfigure the users, extension number (let's say 87XXX range), phones
and the user device profiles in advance?

I'm thinking that if I preconfigure those information, the cucm will think
that those extension number (87XXX) are local and unregistered.

Is there a way to make CUCM thinks that in order to reach 87XXX range, it
will still reach out to Avaya using the SIP trunk? Is there any setting in
the route pattern can do that?

I thinking that CUCM will always find a more exact match locally instead
of through other source like translation pattern or route pattern.

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[cisco-voip] Unified Attendant Console - Advanced Server

2015-03-17 Thread Ryan Huff
Is there a way to tell if I have the resilience option? Looks like my SKU is 
L-CUAC10X-ADV-HA, so I assume the HA is the designator for resilience / high 
availability? Can anyone confirm?

Thanks,

Ryan
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Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...

2015-03-10 Thread Ryan Huff
Jonathan,

Have you applied any other cop or engineering special files besides the version 
3 keys?

Is your cluster based on the unrestricted export version or the restricted 
export version? Does the update your applying match (restricted versus 
unrestricted)

Can you list out the specific version of 10.0.1 you are using and the specific 
version version of 10.5.1 that you want to go to?

If you do a show version active on all the cluster nodes; what do you see? Is 
it the same on all nodes?

I assume you tried to download a new image file from CCO and use that?

Thanks,

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Re: [cisco-voip] 7841 SIP Phone audio issues on some calls

2015-03-10 Thread Ryan Huff
is the MCB onnet or offnet?

Thanks,

Ryan 


 Date: Tue, 10 Mar 2015 08:20:16 -0500
 From: erick...@gmail.com
 To: cisco-voip@puck.nether.net
 Subject: [cisco-voip] 7841 SIP Phone audio issues on some calls
 
 Anyone noticing problems on 7841 SIP phones when calling outbound with
 a SIP provider?
 
 When we call Microsoft Conference bridge it connects but we can't hear
 the bridge.
 It works fine from a SCCP phone on same Call Manager and same SIP trunk.
 All other calls work on the 7841 outbound.
 
 CUCM Version 10.5.1.1-7
 
 Firmware: sip78xx.10-1-1SR1-4
 
 Thanks
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Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

2015-03-10 Thread Ryan Huff
Could you tell me what version of CCM you are dealing with? If you go under 
Call Routing - Translation Pattern and click Add New, do you see Route Next 
Hop By Calling Party Number?

Thanks,

Ryan


From: norm.nichol...@kitchener.ca
To: cisco-voip@puck.nether.net
Date: Tue, 10 Mar 2015 12:43:04 +
Subject: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0









 
I have been asked to send an outside caller to a voicemail box so when the DNIS 
or ANI shows 519-XXX- I want that caller to go to a specific mailbox. Is 
this possible ?
 
 
 
 
Thanks
 
 
 
 
 
 
 
Norm Nicholson
Telecom Analyst
City of Kitchener
(519) 741-2200 x 7000
 
 




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Re: [cisco-voip] DNIS Question

2015-03-06 Thread Ryan Huff
Not automatically, I agree.

Thanks,

Ryan

 Original Message 
From: Daniel Pagan dpa...@fidelus.com
Sent: Friday, March 6, 2015 09:34 AM
To: Ryan Huff 
ryanh...@outlook.com,dennis.h...@wwt.com,norm.nichol...@kitchener.ca,cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] DNIS Question

!-- /* Font Definitions */ @font-face {font-family:Cambria Math;
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span.MsoHyperlink {mso-style-priority:99; color:blue; 
text-decoration:underline;} a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99; color:purple;   text-decoration:underline;} p  
 {mso-style-priority:99; mso-margin-top-alt:auto;
margin-right:0in;   mso-margin-bottom-alt:auto; margin-left:0in;   
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span.EmailStyle18 {mso-style-type:personal-reply; 
font-family:Calibri,sans-serif;   color:#404040;  font-weight:normal;
 font-style:normal;  text-decoration:none none;} .MsoChpDefault  
{mso-style-type:export-only;font-family:Calibri,sans-serif;} @page 
WordSection1   {size:8.5in 11.0in; margin:1.0in 1.0in 1.0in 1.0in;} 
div.WordSection1   {page:WordSection1;} -- 

But this doesn’t provide the logic required for the call acceptance rule of 
“accept only the first (regardless of ToD), reject all others, reset after 24 
hours”. Aside from using some form of UCCX scripting or using a custom app 
integrated through the JTAPI or Routing Rules API, I too don’t see CUCM 
natively accomplishing this.

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Ryan 
Huff
Sent: Friday, March 06, 2015 9:23 AM
To: dennis.h...@wwt.com; norm.nichol...@kitchener.ca; 
cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] DNIS Question

 

If youre on a modern version of ccm you could use a combo of route to next 
hop with calling party I'd and hunts.

That would be a fun one to play around with.

Thanks,

Ryan



 Original Message 
From: Heim, Dennis dennis.h...@wwt.com
Sent: Friday, March 6, 2015 09:13 AM
To: norm.nichol...@kitchener.ca,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] DNIS Question

Out of the box no. You could leverage the Routing Rules API to build a policy 
and database lookup.

 

Dennis Heim | Emerging Technology Architect (Collaboration)

World Wide Technology, Inc. | +1 314-212-1814





Innovation happens on project squared -- http://www.projectsquared.com

 

Click here to join me in my Collaboration Meeting Room

 

 

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
norm.nichol...@kitchener.ca
Sent: Friday, March 06, 2015 6:07 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] DNIS Question

 

 

I have a user request for the following:

 

 

We are having an issue with a nuisance caller calling dispatch 20-30 times a 
day for the past month.  Can we set up the phone system to only allow the 
following number to call once a day on our   and  extension numbers ?

 

519 XXX  

 

After the one call is used for that day, the following call goes to this 
message:

“If this is an emergency, please hang up and call 911 (hang up)”

 

 

 

 

The question is it possible to accept one call a day then route them to a 
message or is it all or nothing ?

 

 

 

 

 

 

 

Thanks

 

 

 

 

 

 

Norm Nicholson

Telecom Analyst

City of Kitchener

(519) 741-2200 x 7000

 

 

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Re: [cisco-voip] DNIS Question

2015-03-06 Thread Ryan Huff
If youre on a modern version of ccm you could use a combo of route to next hop 
with calling party I'd and hunts.

That would be a fun one to play around with.

Thanks,

Ryan

 Original Message 
From: Heim, Dennis dennis.h...@wwt.com
Sent: Friday, March 6, 2015 09:13 AM
To: norm.nichol...@kitchener.ca,cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] DNIS Question

!-- /* Font Definitions */ @font-face {font-family:Cambria Math;
panose-1:2 4 5 3 5 4 6 3 2 4;} @font-face   {font-family:Calibri;   
panose-1:2 15 5 2 2 2 4 3 2 4;} /* Style Definitions */ p.MsoNormal, 
li.MsoNormal, div.MsoNormal{margin:0in;margin-bottom:.0001pt;  
font-size:11.0pt;   font-family:Calibri,sans-serif;} a:link, 
span.MsoHyperlink{mso-style-priority:99; color:blue; 
text-decoration:underline;} a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99; color:purple;   text-decoration:underline;} 
span.EmailStyle17   {mso-style-type:personal;   
font-family:Calibri,sans-serif;   color:windowtext;} span.EmailStyle18   
 {mso-style-type:personal-reply; font-family:Calibri,sans-serif; 
  color:#1F497D;} .MsoChpDefault  {mso-style-type:export-only;
font-size:10.0pt;} @page WordSection1   {size:8.5in 11.0in; margin:1.0in 
1.0in 1.0in 1.0in;} div.WordSection1   {page:WordSection1;} -- 

Out of the box no. You could leverage the Routing Rules API to build a policy 
and database lookup.

 

Dennis Heim | Emerging Technology Architect (Collaboration)

World Wide Technology, Inc. | +1 314-212-1814





Innovation happens on project squared -- http://www.projectsquared.com

 

Click here to join me in my Collaboration Meeting Room

 

 

 

From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of 
norm.nichol...@kitchener.ca
Sent: Friday, March 06, 2015 6:07 AM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] DNIS Question

 

 

I have a user request for the following:

 

 

We are having an issue with a nuisance caller calling dispatch 20-30 times a 
day for the past month.  Can we set up the phone system to only allow the 
following number to call once a day on our   and  extension numbers ?

 

519 XXX  

 

After the one call is used for that day, the following call goes to this 
message:

“If this is an emergency, please hang up and call 911 (hang up)”

 

 

 

 

The question is it possible to accept one call a day then route them to a 
message or is it all or nothing ?

 

 

 

 

 

 

 

Thanks

 

 

 

 

 

 

Norm Nicholson

Telecom Analyst

City of Kitchener

(519) 741-2200 x 7000

 

 

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Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...

2015-03-10 Thread Ryan Huff
Martin,

In your case, it may also be worth checking, have you installed the refresh 
upgrade COP file 
(http://www.cisco.com/web/software/282204704/18582/ciscocm.refresh_upgrade_v1.5.pdf)?

Thanks,

Ryan

From: ryanh...@outlook.com
To: m...@bilobit.com; jonv...@gmail.com
Date: Tue, 10 Mar 2015 12:44:16 -0400
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...




Martin,

I have done a 7.1.3 to 10.5.1 before; I bit the bullet and setup PCD, and it 
went off without a hitch. 

Thanks,

Ryan


From: m...@bilobit.com
To: jonv...@gmail.com; ryanh...@outlook.com
CC: cisco-voip@puck.nether.net
Subject: AW: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...
Date: Tue, 10 Mar 2015 16:32:28 +









At the moment i have the same problem with a jump upgrade from 7.1.3.1-11 
to 10.5.2. It also does not work to 10.5.1.
 
The same: Removed all subs, all network etc. is ok, NTP reachable and 
synchronized and so on.
 
*arghl*
 



Von: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
Im Auftrag von Jonathan Charles

Gesendet: Dienstag, 10. März 2015 16:23

An: Ryan Huff

Cc: cisco-voip@puck.nether.net

Betreff: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...


 

It is a jump upgrade (we are moving from a pizza box to vmware), I am doing the 
upgrade offbox to then import data to a 10.5.1 BE6K...

 


I have rebooted it dozens of times... there is just one server (I have removed 
the other subs)...


 


 


Jonathan



 

On Tue, Mar 10, 2015 at 9:59 AM, Ryan Huff ryanh...@outlook.com wrote:



In your email below it seemed you were trying an update from 10.0.1 to 10.5.1



If you are coming FROM 9.1.2, you may be hitting this bug 
https://tools.cisco.com/bugsearch/bug/CSCup45923



On page 3 of the 9.1.2 SU2 release notes 
(http://www.cisco.com/web/software/282074295/113937/cucm-readme-912su2-Rev2.pdf)
 it warns of this bug. The
 reported work around is to go directly to 10.5.1. 



f you are suggesting that you are still receiving the unknown error when trying 
to directly upgrade 9.1.2
TO 10.5.1, then I'd go ahead and raise a TAC case. 



If this isn't a production cluster or I didn't mind taking it down, I'd reboot 
the cluster then try a 9.1.2 - 10.5.1 upgrade and if I still got the error, 
I'd raise a TAC case.

 

Thanks,



Ryan 








Date: Tue, 10 Mar 2015 09:41:10 -0500

Subject: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...

From: jonv...@gmail.com

To: ryanh...@outlook.com

CC: agrec...@gmail.com; 
cisco-voip@puck.nether.net


 

I applied the RSA cop file... the version from is 9.1.2.11900-12 

 


I have tried going to:


 


10.0.1.11900-2


10.5.1.1-7


 


Same errors every time.


 


All of them are the non-bootable ISO.


 


 


 


Jonathan



 

On Tue, Mar 10, 2015 at 6:22 AM, Ryan Huff ryanh...@outlook.com wrote:

Jonathan,

Have you applied any other cop or engineering special files besides the version 
3 keys?

Is your cluster based on the unrestricted export version or the restricted 
export version? Does the update your applying match (restricted versus 
unrestricted)

Can you list out the specific version of 10.0.1 you are using and the specific 
version version of 10.5.1 that you want to go to?

If you do a show version active on all the cluster nodes; what do you see? Is 
it the same on all nodes?

I assume you tried to download a new image file from CCO and use that?

Thanks,

Ryan







 Original Message 

From: Jonathan Charles jonv...@gmail.com

Sent: Tuesday, March 10, 2015 01:30 AM

To: Andrew Grech agrec...@gmail.com

Subject: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...

CC: Cisco VoIP Group cisco-voip@puck.nether.net

Well, that didn't work... SFTP'd the ISO to the box, same error...

 


Same error message...


 


 


 


Jonathan



 

On Mon, Mar 9, 2015 at 11:45 PM, Jonathan Charles jonv...@gmail.com wrote:


It is re-running after SFTPing from a Windows box (FreeFTPd)...

 


 


Jonathan





 

On Mon, Mar 9, 2015 at 11:31 PM, Andrew Grech agrec...@gmail.com wrote:

Hi john, I've had a ISO from Linux to prime with bad permissions and had this 
error




On 10/03/2015 1:13 PM, Jonathan Charles jonv...@gmail.com wrote:


OK, let me try via SFTP (Windows box)... see what happens...

 


 


Jonathan



 

On Mon, Mar 9, 2015 at 9:26 PM, Charles Goldsmith wo...@justfamily.org wrote:


Given that it is complaining about accessing the upgrade file, are you using 
dvd image/dvd on the host or sftp?  I'd try a different method to see.

 


Not a whole lot of information, but I have seen SFTP cause similar issues, I 
think it was a permissions problem on a linux setup using openssh.



 



On Mon, Mar 9, 2015 at 8:16 PM, Jonathan Charles jonv...@gmail.com wrote:






Getting error: 

 



  


Error encountered: An unknown error occurred while accessing the upgrade file. 
Warning: A system reboot is required when the upgrade process completes or is 
canceled. This will ensure services affected

Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...

2015-03-10 Thread Ryan Huff
Martin,

I have done a 7.1.3 to 10.5.1 before; I bit the bullet and setup PCD, and it 
went off without a hitch. 

Thanks,

Ryan


From: m...@bilobit.com
To: jonv...@gmail.com; ryanh...@outlook.com
CC: cisco-voip@puck.nether.net
Subject: AW: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...
Date: Tue, 10 Mar 2015 16:32:28 +









At the moment i have the same problem with a jump upgrade from 7.1.3.1-11 
to 10.5.2. It also does not work to 10.5.1.
 
The same: Removed all subs, all network etc. is ok, NTP reachable and 
synchronized and so on.
 
*arghl*
 



Von: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
Im Auftrag von Jonathan Charles

Gesendet: Dienstag, 10. März 2015 16:23

An: Ryan Huff

Cc: cisco-voip@puck.nether.net

Betreff: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...


 

It is a jump upgrade (we are moving from a pizza box to vmware), I am doing the 
upgrade offbox to then import data to a 10.5.1 BE6K...

 


I have rebooted it dozens of times... there is just one server (I have removed 
the other subs)...


 


 


Jonathan



 

On Tue, Mar 10, 2015 at 9:59 AM, Ryan Huff ryanh...@outlook.com wrote:



In your email below it seemed you were trying an update from 10.0.1 to 10.5.1



If you are coming FROM 9.1.2, you may be hitting this bug 
https://tools.cisco.com/bugsearch/bug/CSCup45923



On page 3 of the 9.1.2 SU2 release notes 
(http://www.cisco.com/web/software/282074295/113937/cucm-readme-912su2-Rev2.pdf)
 it warns of this bug. The
 reported work around is to go directly to 10.5.1. 



f you are suggesting that you are still receiving the unknown error when trying 
to directly upgrade 9.1.2
TO 10.5.1, then I'd go ahead and raise a TAC case. 



If this isn't a production cluster or I didn't mind taking it down, I'd reboot 
the cluster then try a 9.1.2 - 10.5.1 upgrade and if I still got the error, 
I'd raise a TAC case.

 

Thanks,



Ryan 








Date: Tue, 10 Mar 2015 09:41:10 -0500

Subject: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...

From: jonv...@gmail.com

To: ryanh...@outlook.com

CC: agrec...@gmail.com; 
cisco-voip@puck.nether.net


 

I applied the RSA cop file... the version from is 9.1.2.11900-12 

 


I have tried going to:


 


10.0.1.11900-2


10.5.1.1-7


 


Same errors every time.


 


All of them are the non-bootable ISO.


 


 


 


Jonathan



 

On Tue, Mar 10, 2015 at 6:22 AM, Ryan Huff ryanh...@outlook.com wrote:

Jonathan,

Have you applied any other cop or engineering special files besides the version 
3 keys?

Is your cluster based on the unrestricted export version or the restricted 
export version? Does the update your applying match (restricted versus 
unrestricted)

Can you list out the specific version of 10.0.1 you are using and the specific 
version version of 10.5.1 that you want to go to?

If you do a show version active on all the cluster nodes; what do you see? Is 
it the same on all nodes?

I assume you tried to download a new image file from CCO and use that?

Thanks,

Ryan







 Original Message 

From: Jonathan Charles jonv...@gmail.com

Sent: Tuesday, March 10, 2015 01:30 AM

To: Andrew Grech agrec...@gmail.com

Subject: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...

CC: Cisco VoIP Group cisco-voip@puck.nether.net

Well, that didn't work... SFTP'd the ISO to the box, same error...

 


Same error message...


 


 


 


Jonathan



 

On Mon, Mar 9, 2015 at 11:45 PM, Jonathan Charles jonv...@gmail.com wrote:


It is re-running after SFTPing from a Windows box (FreeFTPd)...

 


 


Jonathan





 

On Mon, Mar 9, 2015 at 11:31 PM, Andrew Grech agrec...@gmail.com wrote:

Hi john, I've had a ISO from Linux to prime with bad permissions and had this 
error




On 10/03/2015 1:13 PM, Jonathan Charles jonv...@gmail.com wrote:


OK, let me try via SFTP (Windows box)... see what happens...

 


 


Jonathan



 

On Mon, Mar 9, 2015 at 9:26 PM, Charles Goldsmith wo...@justfamily.org wrote:


Given that it is complaining about accessing the upgrade file, are you using 
dvd image/dvd on the host or sftp?  I'd try a different method to see.

 


Not a whole lot of information, but I have seen SFTP cause similar issues, I 
think it was a permissions problem on a linux setup using openssh.



 



On Mon, Mar 9, 2015 at 8:16 PM, Jonathan Charles jonv...@gmail.com wrote:






Getting error: 

 



  


Error encountered: An unknown error occurred while accessing the upgrade file. 
Warning: A system reboot is required when the upgrade process completes or is 
canceled. This will ensure services affected by the upgrade process are 
functioning
 properly.



 


Tried upgrade from to 10.0.1 and 10.5.1, same errors...


 


I have applied the ciscocm.version3-keys.cop.sgn patch, no change in behavior...


 


 


Any ideas?


 


 


 


Jonathan


 


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Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

2015-03-10 Thread Ryan Huff
Daniel,

There you go trying to make things slick and cool, lol! I like that approach!

So, it seems his need is that if only 1 particular DNIS dialed one particular 
DN (a uccx trigger from my understanding) in ccm, that it then route the call 
to a specific VM account; but that it would also allow that DNIS to dial any 
other DN in the cluster without going to that specific VM account..

Being that I love to learn new things constantly; in your approach, is there a 
way to accomplish that requirement?

Thanks,

Ryan 


From: dpa...@fidelus.com
To: norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net
Date: Tue, 10 Mar 2015 15:09:32 +
Subject: Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail
8.0









For routing based on ANI, Ryan Huff’s suggestion will certainly help from the 
perspective of CUCM. You can use this alongside CUC routing rules – route calls 
from the 519 area code to a specific mailbox. If you
 know these callers from area code 519 are dialing the same DNIS, then routing 
by ANI in CUCM won’t be needed – simply route the DNIS to voicemail, add a 
routing rule matching the 519 calls, and route to a mailbox.
 


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of norm.nichol...@kitchener.ca

Sent: Tuesday, March 10, 2015 8:43 AM

To: cisco-voip@puck.nether.net

Subject: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0


 
 
I have been asked to send an outside caller to a voicemail box so when the DNIS 
or ANI shows 519-XXX- I want that caller to go to a specific mailbox. Is 
this possible ?
 
 
 
 
Thanks
 
 
 
 
 
 
 
Norm Nicholson
Telecom Analyst
City of Kitchener
(519) 741-2200 x 7000
 
 






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Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

2015-03-10 Thread Ryan Huff
Agreed. 

CCM ANI call based routing is fantastic, works flawlessly once setup and then 
you go a month and forget it was setup (because it works so flawlessly) and you 
make an arbitrary change and BLAMO  no ingress in the cluster.

Long story short, the CCM approach is great if you're committed to remembering 
it is there and the role it plays with your dial plan (because it has to insert 
itself in the ingress path right before the dial plan).

Not trying to scare you off of ccm ani routing Norm, it works great and really 
isn't that hard to setup, but Daniel reminded me of a good point. If you go 
with the CCM approach, document how you have it setup (names of the partitions 
and css's ... etc).

Thanks,

Ryan


From: dpa...@fidelus.com
To: ryanh...@outlook.com; norm.nichol...@kitchener.ca; 
cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0
Date: Tue, 10 Mar 2015 17:50:48 +









Doc on ANI-based call routing in UCCX:
 
http://www.scribd.com/doc/197424933/Cisco-UCCX-ANI-Based-Call-Routing#scribd
 
Norm can use this as a starting point assuming the call is routing into UCCX 
already. But use the call redirect step instead of the set step specified in the
 article. Use call redirect to send the call (which returned TRUE based on the 
previous IF step looking at calling party) to a DN in CUCM configured for 
CFwdAll to Unity Connection. Then have a dedicated voice mailbox for these 
calls with a matching DN.

 
What I’m not sure of is whether CUC will match the voicemail user based on 
first redirecting number, which might use the CTI port performing the redirect… 
Not
 sure… would need to test… but it’s probably the cleanest solution IMO.
 


From: Daniel Pagan


Sent: Tuesday, March 10, 2015 1:11 PM

To: 'Ryan Huff'; norm.nichol...@kitchener.ca; cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0


 
HAH! Slick and cool… Two things I’m far, far from :) Unfortunately it took my 
email below a full hour to be posted!
 
That’s tricky if the desired behavior is to dial a UCCX trigger and, if the ANI 
contains area code 519, only then should CUCM route the call to a specific 
voicemail
 box. If this is correct, then I think routing based on calling-number in CUCM 
is certainly going to be required along with dialed number based routing… if 
scripting isn’t desired...

 
But since it seems they’re dialing a UCCX trigger anyway, assuming it’s not 
just an unregistered CTI RP and calls are actually routing to UCCX, why not 
just edit
 the existing script to read the calling number and redirect the call to a 
predetermined DN that’s configured for CFWD all into voicemail?
 
Something straight forward like…
 
If calling number begins with 516
  True
 Call Redirect
à 56789
  False
  continue
 
In CUCM… DN 56789 - CFwdAll to Voicemail - Unity CH or User w/ mailbox
 
Add the partition of 56789 to the CSS of the UCCX CTI ports performing the 
redirect.
 
- Dan
 
 


From: Ryan Huff [mailto:ryanh...@outlook.com]


Sent: Tuesday, March 10, 2015 12:23 PM

To: Daniel Pagan; norm.nichol...@kitchener.ca;
cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0


 

Daniel,



There you go trying to make things slick and cool, lol! I like that approach!



So, it seems his need is that if only 1 particular DNIS dialed one particular 
DN (a uccx trigger from my understanding) in ccm, that it then route the call 
to a specific VM account; but that it would also allow that DNIS to dial any 
other DN in the cluster
 without going to that specific VM account..



Being that I love to learn new things constantly; in your approach, is there a 
way to accomplish that requirement?



Thanks,



Ryan 




From:
dpa...@fidelus.com

To: norm.nichol...@kitchener.ca;
cisco-voip@puck.nether.net

Date: Tue, 10 Mar 2015 15:09:32 +

Subject: Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

For routing based on ANI, Ryan Huff’s suggestion will certainly help from the 
perspective of CUCM. You can use this alongside CUC routing rules – route calls 
from the 519 area
 code to a specific mailbox. If you know these callers from area code 519 are 
dialing the same DNIS, then routing by ANI in CUCM won’t be needed – simply 
route the DNIS to voicemail, add a routing rule matching the 519 calls, and 
route to a mailbox.
 


From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]
On Behalf Of norm.nichol...@kitchener.ca

Sent: Tuesday, March 10, 2015 8:43 AM

To: cisco-voip@puck.nether.net

Subject: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0


 
 
I have been asked to send an outside caller to a voicemail box so when the DNIS 
or ANI shows 519-XXX- I want that caller to go to a specific mailbox. Is 
this possible ?
 
 
 
 
Thanks
 
 
 
 
 
 
 
Norm Nicholson
Telecom Analyst
City of Kitchener
(519) 741-2200

Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

2015-03-10 Thread Ryan Huff
I'm not sure I correctly understand your question. Are you asking if instead of 
routing the calling number directly to a VM account, route to a UCCX queue 
trigger?

Thanks,

Ryan 


From: norm.nichol...@kitchener.ca
To: ryanh...@outlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0
Date: Tue, 10 Mar 2015 14:06:21 +









Great info and one more question….
 
 
Can we do this on specific CCX pilot # s ( they are DID’s ) verses every call 
this number makes to our call manager?.

 
 
 
 
 
Thanks
 
 
 
 


From: Ryan Huff [mailto:ryanh...@outlook.com]


Sent: Tuesday, March 10, 2015 9:26 AM

To: Norm Nicholson; cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0


 


If you have a modern CCM version (8.x or above), which I'm guessing you are 
indicating such by the subject line of the email, you can create a scenario 
where CCM can perform routing decisions based upon the
 calling party number.



Please review this post: 
http://ryanthomashuff.com/2014/11/call-blocking-by-caller-id/ and follow the 
steps.




The variation that you are going to need to do though is once CCM makes a match 
on the
calling party number you'll need to translate the called party number to a CTI 
Route Point or something that forwards into the Unity VM account that you want.



Thanks,



Ryan




From: 
ryanh...@outlook.com

To: norm.nichol...@kitchener.ca;
cisco-voip@puck.nether.net

Date: Tue, 10 Mar 2015 08:57:27 -0400

Subject: Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

Could you tell me what version of CCM you are dealing with? If you go under 
Call Routing - Translation Pattern and click Add New, do you see Route Next 
Hop By Calling Party Number?



Thanks,



Ryan








From: 
norm.nichol...@kitchener.ca

To: cisco-voip@puck.nether.net

Date: Tue, 10 Mar 2015 12:43:04 +

Subject: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

 
I have been asked to send an outside caller to a voicemail box so when the DNIS 
or ANI shows 519-XXX- I want that caller to go to a specific mailbox. Is 
this possible ?
 
 
 
 
Thanks
 
 
 
 
 
 
 
Norm Nicholson
Telecom Analyst
City of Kitchener
(519) 741-2200 x 7000
 
 



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Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

2015-03-10 Thread Ryan Huff

Yes you can do that, but it gets a bit more tricky, and since your dealing with 
a CCX trigger, you could probably script something in uccx just as easy. 

If you want to do this in CCM though, put on a hard hat and lets get to it ...

Assuming you've already read and deployed the previous example ...

In the initial Partition that contains the ! pattern that routes everything by 
calling party, you would create another translation in that same partition and 
the pattern would be the called party number. So the partition would contain a 
translation for ! and a translation for the CCX trigger. Both translations 
would route to next hop by calling party. The translation that contains your 
CCX trigger would have to use a CSS that called a different partition than the 
! translation. 

In the new partition there would be a ! translation pattern that would call out 
with a CSS that can access the DN's partition and then a translation pattern 
that matches the called party and then either block the route or translate the 
calling party to your CTI Route Point that forwards to your VM account in Unity.

In the end, you end up with a scenario where the CCX trigger is called and 
routed to the next hop by calling party and once at the next hop, if the 
calling party matches the defined pattern, is blocked or routed. However, if 
that calling party called any other DN besides the CCX trigger, it would route 
to the next hop in the original filter partition and be allowed to call 
(provided I wasn't blocking that calling party in that partition).

The reason all this works is because of CCM's route matching matrix; wherein 
the most specific route match always wins routing decisions. A specified number 
pattern will always be more specific than ! which essentially matches anything.

Let me know if you need addition help offline, this is sort of complicated in 
writing but when you see it in production you're like really, that's it?

Thanks,

Ryan Huff


From: norm.nichol...@kitchener.ca
To: ryanh...@outlook.com; cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0
Date: Tue, 10 Mar 2015 15:00:48 +









 
Sorry the incoming phone number we want to route to a voicemail box, but only 
when it calls in on a specific CCX pilot # .  They should be allowed to call all
 other numbers and DID’s within our Call Manager.
 
 
 
Thanks
 
 
 


From: Ryan Huff [mailto:ryanh...@outlook.com]


Sent: Tuesday, March 10, 2015 10:45 AM

To: Norm Nicholson; cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0


 

I'm not sure I correctly understand your question. Are you asking if instead of 
routing the calling number directly to a VM account, route to a UCCX queue 
trigger?



Thanks,



Ryan 








From:
norm.nichol...@kitchener.ca

To: ryanh...@outlook.com; 
cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

Date: Tue, 10 Mar 2015 14:06:21 +

Great info and one more question….
 
 
Can we do this on specific CCX pilot # s ( they are DID’s ) verses every call 
this number makes to our call manager?.

 
 
 
 
 
Thanks
 
 
 
 


From: Ryan Huff [mailto:ryanh...@outlook.com]


Sent: Tuesday, March 10, 2015 9:26 AM

To: Norm Nicholson; cisco-voip@puck.nether.net

Subject: RE: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0


 


If you have a modern CCM version (8.x or above), which I'm guessing you are 
indicating such by the subject line of the email, you can create a scenario 
where CCM can perform routing decisions
 based upon the calling party number.



Please review this post: 
http://ryanthomashuff.com/2014/11/call-blocking-by-caller-id/ and follow the 
steps.




The variation that you are going to need to do though is once CCM makes a match 
on the
calling party number you'll need to translate the called party number to a CTI 
Route Point or something that forwards into the Unity VM account that you want.



Thanks,



Ryan




From: 
ryanh...@outlook.com

To: norm.nichol...@kitchener.ca;
cisco-voip@puck.nether.net

Date: Tue, 10 Mar 2015 08:57:27 -0400

Subject: Re: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

Could you tell me what version of CCM you are dealing with? If you go under 
Call Routing - Translation Pattern and click Add New, do you see Route Next
 Hop By Calling Party Number?



Thanks,



Ryan




From: 
norm.nichol...@kitchener.ca

To: cisco-voip@puck.nether.net

Date: Tue, 10 Mar 2015 12:43:04 +

Subject: [cisco-voip] Forwarding an inbound caller straight to Voicemail 8.0

 
I have been asked to send an outside caller to a voicemail box so when the DNIS 
or ANI shows 519-XXX- I want that caller to go to a specific mailbox. Is 
this possible ?
 
 
 
 
Thanks
 
 
 
 
 
 
 
Norm Nicholson
Telecom Analyst
City of Kitchener
(519) 741-2200 x 7000

Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...

2015-03-10 Thread Ryan Huff
In your email below it seemed you were trying an update from 10.0.1 to 10.5.1

If you are coming FROM 9.1.2, you may be hitting this bug 
https://tools.cisco.com/bugsearch/bug/CSCup45923

On page 3 of the 9.1.2 SU2 release notes 
(http://www.cisco.com/web/software/282074295/113937/cucm-readme-912su2-Rev2.pdf)
 it warns of this bug. The reported work around is to go directly to 10.5.1. 

f you are suggesting that you are still receiving the unknown error when trying 
to directly upgrade 9.1.2 TO 10.5.1, then I'd go ahead and raise a TAC case. 

If this isn't a production cluster or I didn't mind taking it down, I'd reboot 
the cluster then try a 9.1.2 - 10.5.1 upgrade and if I still got the error, 
I'd raise a TAC case.
 
Thanks,

Ryan 


Date: Tue, 10 Mar 2015 09:41:10 -0500
Subject: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...
From: jonv...@gmail.com
To: ryanh...@outlook.com
CC: agrec...@gmail.com; cisco-voip@puck.nether.net

I applied the RSA cop file... the version from is 9.1.2.11900-12 
I have tried going to:
10.0.1.11900-210.5.1.1-7
Same errors every time.
All of them are the non-bootable ISO.


Jonathan
On Tue, Mar 10, 2015 at 6:22 AM, Ryan Huff ryanh...@outlook.com wrote:
Jonathan,
Have you applied any other cop or engineering special files besides the version 
3 keys?
Is your cluster based on the unrestricted export version or the restricted 
export version? Does the update your applying match (restricted versus 
unrestricted)
Can you list out the specific version of 10.0.1 you are using and the specific 
version version of 10.5.1 that you want to go to?
If you do a show version active on all the cluster nodes; what do you see? Is 
it the same on all nodes?
I assume you tried to download a new image file from CCO and use that?
Thanks,
Ryan


 Original Message 
From: Jonathan Charles jonv...@gmail.com
Sent: Tuesday, March 10, 2015 01:30 AM
To: Andrew Grech agrec...@gmail.com
Subject: Re: [cisco-voip] Upgrade 9.1.2 to 10.0 fails...
CC: Cisco VoIP Group cisco-voip@puck.nether.net

Well, that didn't work... SFTP'd the ISO to the box, same error...
Same error message...


Jonathan
On Mon, Mar 9, 2015 at 11:45 PM, Jonathan Charles jonv...@gmail.com wrote:
It is re-running after SFTPing from a Windows box (FreeFTPd)...

Jonathan
On Mon, Mar 9, 2015 at 11:31 PM, Andrew Grech agrec...@gmail.com wrote:
Hi john, I've had a ISO from Linux to prime with bad permissions and had this 
error
On 10/03/2015 1:13 PM, Jonathan Charles jonv...@gmail.com wrote:
OK, let me try via SFTP (Windows box)... see what happens...

Jonathan
On Mon, Mar 9, 2015 at 9:26 PM, Charles Goldsmith wo...@justfamily.org wrote:
Given that it is complaining about accessing the upgrade file, are you using 
dvd image/dvd on the host or sftp?  I'd try a different method to see.
Not a whole lot of information, but I have seen SFTP cause similar issues, I 
think it was a permissions problem on a linux setup using openssh.
On Mon, Mar 9, 2015 at 8:16 PM, Jonathan Charles jonv...@gmail.com wrote:
Getting error: 
Error encountered: An unknown error occurred while accessing the 
upgrade file. Warning: A system reboot is required when the upgrade process 
completes or is canceled. This will ensure services affected by the upgrade 
process are functioning properly.
Tried upgrade from to 10.0.1 and 10.5.1, same errors...
I have applied the ciscocm.version3-keys.cop.sgn patch, no change in behavior...

Any ideas?


Jonathan

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Re: [cisco-voip] vMotion w/o Shared Storage

2015-03-10 Thread Ryan Huff
Shutdown, yes. Running, no.

Thanks,

Ryan

 Original Message 
From: Daniel Pagan dpa...@fidelus.com
Sent: Tuesday, March 10, 2015 05:06 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] vMotion w/o Shared Storage

Quick question...

vMotion of a shut down UCM between two hosts without shared storage using 
vCenter. Is this supported? The virtualization document for UC platforms says 
vMotion is supported on shared storage, so I figured to ask. Is this migration 
method supported?

- Dan



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[cisco-voip] Sip design question

2015-03-30 Thread Ryan Huff
I have a pair of cubes on 4000 series ISRs. I want to do cube-ha on the ccm 
facing side and the itsp facing side.

1.) Am I better off just doing HSRP on both sides (which is 70% of cube-ha 
anyway) or is it practical to do the connected call failover portion?

2.) If I include the connected call failover,  which side would I do that one, 
1 or both (ccm facing side or itsp facing side)?

Thanks,

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Re: [cisco-voip] Temp Fail since upgrade

2015-03-30 Thread Ryan Huff
Lisa,

SIP or TDM/PRI?

Have you gandered into RTMT and taken a look at any active / recent alerts?

Thanks,

Ryan

 Original Message 
From: Lisa Notarianni lisa.notaria...@scranton.edu
Sent: Monday, March 30, 2015 04:48 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Temp Fail since upgrade

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text-indent:-9.0pt;} ol {margin-bottom:0in;} ul 
{margin-bottom:0in;} -- 

A few weeks ago we upgraded Call Manager and Unity Connection from 8.6.2 to 
10.5.1.

 

We have experienced 2 intermittent outbound calling issues:

 

1.   “Temp Fail” shows on phone display and only option on phone button is 
“End Call”

2.   Outbound callers dial a number and hear nothing but notice the phone 
display says “Connected”.  Caller cannot hear anything.  Person they called 
cannot hear anything.

***These issues are only intermittent.***

 

We had no issues before the upgrade.

 

Is anyone experiencing the similar issues?

 

Thank you in advance.

 

 



 

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Re: [cisco-voip] Temp Fail since upgrade

2015-03-30 Thread Ryan Huff
I'm not aware of any known caveats of that nature regarding that upgrade path.

Any recent network changes between ccm and the gateways since the upgrade?

I have seen situations with MGCP where the ccm side of the gateway config had 
to be rebuilt after a CCM upgrade.

Are you running an ED code version on the gateway? Check for caveats in your 
gateway code version with ccm 10.5.1

Have you restarted the MGCP process on the gateways (inside a maintenance 
window)?

In a more drastic approach, you could try and reintegrate the gateways as h.323 
and see if the issue persists.

Thanks,

Ryan

 Original Message 
From: Lisa Notarianni lisa.notaria...@scranton.edu
Sent: Monday, March 30, 2015 05:01 PM
To: Ryan Huff ryanh...@outlook.com,cisco-voip@puck.nether.net
Subject: RE: [cisco-voip] Temp Fail since upgrade

2 PRIs one each Communication Media Module– MGCP gateways. 2 Call Managers set 
up as redundant.

Cisco TAC looked at RTMT  with me today.  They want us to set up packet 
captures and try to duplicate the problem tomorrow.  I am just wondering if 
this is an upgrade issue that anyone else may be experiencing since we had no 
problems before the upgrade.



From: Ryan Huff [mailto:ryanh...@outlook.com]
Sent: Monday, March 30, 2015 4:52 PM
To: Lisa Notarianni; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Temp Fail since upgrade


Lisa,

SIP or TDM/PRI?

Have you gandered into RTMT and taken a look at any active / recent alerts?

Thanks,

Ryan


 Original Message 
From: Lisa Notarianni 
lisa.notaria...@scranton.edumailto:lisa.notaria...@scranton.edu
Sent: Monday, March 30, 2015 04:48 PM
To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net
Subject: [cisco-voip] Temp Fail since upgrade
A few weeks ago we upgraded Call Manager and Unity Connection from 8.6.2 to 
10.5.1.

We have experienced 2 intermittent outbound calling issues:


1.   “Temp Fail” shows on phone display and only option on phone button is 
“End Call”

2.   Outbound callers dial a number and hear nothing but notice the phone 
display says “Connected”.  Caller cannot hear anything.  Person they called 
cannot hear anything.
***These issues are only intermittent.***

We had no issues before the upgrade.

Is anyone experiencing the similar issues?

Thank you in advance.


[LNsignatureFile]

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Re: [cisco-voip] Sip design question

2015-04-01 Thread Ryan Huff
Charles,

I guess that is a better place to start; I may be going down this road in a 
near future. I have been reading 
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-high-availability.html#concept_5013D60352C446769D62736C8CDE87E8
 which seems to suggest that L2 box to box is possible on the 4451.

Are you saying it is not?

Thanks,

Ryan

 From: wo...@justfamily.org
 Date: Tue, 31 Mar 2015 23:59:34 -0600
 Subject: Re: [cisco-voip] Sip design question
 To: ryanh...@outlook.com
 CC: cisco-voip@puck.nether.net
 
 Please correct me if I'm wrong, but I thought cube-ha was missing from
 the code on these?  same as the ASR's since they are all running
 ios-xe.
 
 I have not tested it myself, just doing a lot of reading in
 preparation of deploying these.
 
 On Mon, Mar 30, 2015 at 6:29 AM, Ryan Huff ryanh...@outlook.com wrote:
  I have a pair of cubes on 4000 series ISRs. I want to do cube-ha on the ccm
  facing side and the itsp facing side.
 
  1.) Am I better off just doing HSRP on both sides (which is 70% of cube-ha
  anyway) or is it practical to do the connected call failover portion?
 
  2.) If I include the connected call failover,  which side would I do that
  one, 1 or both (ccm facing side or itsp facing side)?
 
  Thanks,
 
  Ryan
 
 
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Re: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2

2015-03-23 Thread Ryan Huff
I've used the ATA190 and it's family of 18X pretty often. In the case of the 
SIP devices like the 190 or 187 I have always been able to use the default SIP 
Profile settings; of course I'm generally dealing with vanilla networks and 
CCM's built very closely to the SRND.

Here is where I would start (if you haven't already):

What do you have the device security mode set to? If you have it 
set to Authenticated or Encrypted but it has not received a CTL file, the 
phone will attempt registration up to four times to make a secure 
connection.

Are the MAC addresses for Phone 1 and Phone 2 correct (Phone 2 will usually 
loose the first two characters and append a 01 at the end of the device name). 
If you support auto registration, remove the devices and let them auto register 
and see if they stabilize.

Any layer 1/2 issues (change out the patch cable, check the connection into the 
switchport ... etc)? Do you see any interface drops on the switchport? Is the 
switchport full duplex 10/100 or 10/1000? Have you shut/no shut the port just 
to see if clearing the port helps?

Go through the IVR menu on the ATA itself and verify that it reports all the 
same settings that you CCM device configs show. 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cata/190/1_0/english/administration/guide/sip/ATA190/a190_agBcd.html

Thanks,

Ryan 

From: bertacco.alessan...@alice.it
To: cisco-voip@puck.nether.net
Date: Mon, 23 Mar 2015 11:28:44 +0100
Subject: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2

Hi Guys,   ATA190 FW version 1.1.2.(005) with CUCM 10.5.2.1-5, de-register 
randomly, and need to reboot the device to re-register again on the CUCM. 
Anyone as the same issue? Do you use a special SIP Device profile changing some 
timings, or you are using standard SIP Device Profile? Thank you all Regards  
Alessandro Bertacco 
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Re: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2

2015-03-23 Thread Ryan Huff
I should also add if you have a spare ATA190, swap it out and see if the issue 
stabilizes or continues on with the different device. If the issue persists 
with a different device, likely not the ATA.

If you don't have a spare 190, then try your 190 on a different cluster if you 
have one (I don't think the 190 is officially supported on any version of CME 
yet, but you could try that if you don't have another cluster).

Thanks,

Ryan 


From: ryanh...@outlook.com
To: bertacco.alessan...@alice.it; cisco-voip@puck.nether.net
Date: Mon, 23 Mar 2015 09:19:36 -0400
Subject: Re: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2




I've used the ATA190 and it's family of 18X pretty often. In the case of the 
SIP devices like the 190 or 187 I have always been able to use the default SIP 
Profile settings; of course I'm generally dealing with vanilla networks and 
CCM's built very closely to the SRND.

Here is where I would start (if you haven't already):

What do you have the device security mode set to? If you have it 
set to Authenticated or Encrypted but it has not received a CTL file, the 
phone will attempt registration up to four times to make a secure 
connection.

Are the MAC addresses for Phone 1 and Phone 2 correct (Phone 2 will usually 
loose the first two characters and append a 01 at the end of the device name). 
If you support auto registration, remove the devices and let them auto register 
and see if they stabilize.

Any layer 1/2 issues (change out the patch cable, check the connection into the 
switchport ... etc)? Do you see any interface drops on the switchport? Is the 
switchport full duplex 10/100 or 10/1000? Have you shut/no shut the port just 
to see if clearing the port helps?

Go through the IVR menu on the ATA itself and verify that it reports all the 
same settings that you CCM device configs show. 
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cata/190/1_0/english/administration/guide/sip/ATA190/a190_agBcd.html

Thanks,

Ryan 

From: bertacco.alessan...@alice.it
To: cisco-voip@puck.nether.net
Date: Mon, 23 Mar 2015 11:28:44 +0100
Subject: [cisco-voip] ATA190 Random De-Register from CUCM 10.5.2

Hi Guys,   ATA190 FW version 1.1.2.(005) with CUCM 10.5.2.1-5, de-register 
randomly, and need to reboot the device to re-register again on the CUCM. 
Anyone as the same issue? Do you use a special SIP Device profile changing some 
timings, or you are using standard SIP Device Profile? Thank you all Regards  
Alessandro Bertacco 
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Re: [cisco-voip] Sip design question

2015-04-01 Thread Ryan Huff
CUBE-HA on a 4k doesn't seem very battle tested yet. Clearly it shouldn't go 
the way of CUBE-SP on an ASR1k which got dumped.

Some of those are significant caveats though (SDP passthru being a possible 
deal killer for me); almost makes just doing plain old HSRP and setting the 
client expectation for failover seem just as reasonable.

Thanks,

Ryan


 From: wo...@justfamily.org
 Date: Wed, 1 Apr 2015 08:17:45 -0600
 Subject: Re: [cisco-voip] Sip design question
 To: ryanh...@outlook.com
 CC: cisco-voip@puck.nether.net
 
 Per this: 
 http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_mgmt/configuration/xe-3s/cube-mgmt-xe-3s-book/voi-stateful-switchover.html
 it says it is on 3.2 or later, but it does have a list of caveats,
 perhaps that is what I was thinking about.
 
 Sorry for the false alarm.
 
 
 
 On Wed, Apr 1, 2015 at 5:33 AM, Ryan Huff ryanh...@outlook.com wrote:
  Charles,
 
  I guess that is a better place to start; I may be going down this road in a
  near future. I have been reading
  http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-high-availability.html#concept_5013D60352C446769D62736C8CDE87E8
  which seems to suggest that L2 box to box is possible on the 4451.
 
  Are you saying it is not?
 
  Thanks,
 
  Ryan
 
  From: wo...@justfamily.org
  Date: Tue, 31 Mar 2015 23:59:34 -0600
  Subject: Re: [cisco-voip] Sip design question
  To: ryanh...@outlook.com
  CC: cisco-voip@puck.nether.net
 
 
  Please correct me if I'm wrong, but I thought cube-ha was missing from
  the code on these? same as the ASR's since they are all running
  ios-xe.
 
  I have not tested it myself, just doing a lot of reading in
  preparation of deploying these.
 
  On Mon, Mar 30, 2015 at 6:29 AM, Ryan Huff ryanh...@outlook.com wrote:
   I have a pair of cubes on 4000 series ISRs. I want to do cube-ha on the
   ccm
   facing side and the itsp facing side.
  
   1.) Am I better off just doing HSRP on both sides (which is 70% of
   cube-ha
   anyway) or is it practical to do the connected call failover portion?
  
   2.) If I include the connected call failover, which side would I do that
   one, 1 or both (ccm facing side or itsp facing side)?
  
   Thanks,
  
   Ryan
  
  
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Re: [cisco-voip] FXS voltages / POTS compatibility

2015-03-02 Thread Ryan Huff



Lelio,

You could remove the mgcp service from the fxs port, then create your dial 
peers (assuming your alarm only wants tone and doesn't need inward).

Thanks,

Ryan


From: le...@uoguelph.ca
Date: Mon, 2 Mar 2015 08:21:51 -0500
To: jandrewar...@ccgs.wa.edu.au
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] FXS voltages / POTS compatibility

James,
Does this mean you have an h323 gateway? Right now, I have MGCP, which I'm 
guessing, precludes me from doing this. 

Sent from my iPhone
On Mar 2, 2015, at 7:51 AM, James Andrewartha jandrewar...@ccgs.wa.edu.au 
wrote:

With our security systems I have to remove the call manager from the call path 
for the system to complete due to the nonstandard tones they send. From my 
notes on how to configure this:
On the VG224s:
voice class h323 1  h225 timeout tcp establish 3voice-port 2/13  no 
timeoutsdial-peer voice 23 pots  service stcappdial-peer voice 99 voip  
description h323 direct to voip1 for alarm number  destination-pattern 13451015 
 session target ipv4:10.101.0.5  voice-class h323 1  codec g711ulaw  no 
vaddial-peer voice 98 voip  description h323 direct to voip2 for alarm number  
preference 1  destination-pattern 13451015  session target ipv4:10.101.0.6  
voice-class h323 1  codec g711ulaw  no vad
On the 2921s:
voice service voipip address trusted list  ipv4 10.100.0.10 255.255.255.255
The AVG is 10.100.0.10, the 2921s are 10.101.0.5 and .6, and dial-peer voice 23 
is for voice-port 2/13. These are GE security panels I think (which their MAC 
OUI confirms).
-- James AndrewarthaNetwork  Projects EngineerChrist Church Grammar 
SchoolClaremont, Western AustraliaPh. (08) 9442 1757Mob. 0424 160 877
From:  Justin Steinberg jsteinb...@gmail.com
Date:  Monday, 2 March 2015 2:14 am
To:  chris tknch...@gmail.com
Cc:  Cisco VOIP cisco-voip@puck.nether.net
Subject:  Re: [cisco-voip] FXS voltages / POTS compatibility

Are you using local h323 or sip 'pots' dialpeers to route directly between your 
FXS and T1 port?  Or is call manager in between the call due to MGCP or VOIP 
dialpeers involved in the dialplan ?I doubt your issue is line voltage, since 
you can see the call being placed.  My guess is the DSP is processing the call 
and causing issues.  I've seem alarm boxes use nonstandard DTMF transmission 
that isn't properly recognized by the DSP.The 2800 supports DSP bypass by 
default when you route directly between ports using POTS dialpeers.   You do 
need to have properly configured network clock configuration.Can you send a 
copy of your config along with the output of 'show controller t1' and 'show 
network-clock'JustinOn Feb 28, 2015 10:33 PM, chris tknch...@gmail.com 
wrote:
Hey Ryan,
We have a channelized T1 with channels split between voice/data so the voice 
path is TDM. We have a VIC-4FXS/DID and for each of the two ports we have a 
single copper pair with rj11 on both ends, one side going to the FXS port and 
the other is going into alarm panel. The total distance from the 2800 to the 
alarm panel is around 20-30 feet and its a direct run, no 66 blocks or anything 
in between.Don't know model of the panel (this is another location) 
From what I've read I think the problem is the default idle-voltage the 
VIC-4FXS/DID is only -24V but based on the link I sent in the first email I 
thought this could be reconfigured through the idle-voltage option but this 
doesnt seem to be available when I try to enter it under the voice-port. When 
I talked to the alarm company and told them I see the calls going through the 
guy told me the alarm doesn't check the line state based on the dialtone and 
he said that it uses the voltage to see when the line is idle, ringing, etc 
and I think this is where the problem lies.
Someone recommend this adapter offlist which looks interesting but the price is 
a little nuts as it costs more than all the equipment we have installed at this 
site combined. 
http://www.homedepot.com/p/Viking-1-Line-Long-Loop-Adapter-VK-LLA-1/20435

Chris

On Sat, Feb 28, 2015 at 8:18 PM, Ryan Huff ryanh...@outlook.com wrote:
Chris,Can you diagram the connections for me? Are the copper pairs swinging off 
a 66 block before terminating to the alarm panel or is there a direct copper 
path between the fxs port and the alarm? Are you using an RJ-11 or RJ-14 
configuration?Could you estimate the copper distance between the termination 
points?Is the pstn path for the VG SIP or TOM?Also, I would be curious to know 
if the alarm panel is a Simplex Grinnell?Thanks,Ryan

 Original Message 
From: chris tknch...@gmail.com
Sent: Saturday, February 28, 2015 07:52 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] FXS voltages / POTS compatibility

HelloWe have a location with a 2800 acting as a voice gateway where we have 2 
FXS voice ports going to an alarm system. We are using the vic-4fxs/did line 
card.We have the alarm company saying they are seeing the panel reporting Comm 
trouble so we checked the call records

[cisco-voip] 79xx and the ITL, without access to the existing phone system

2015-03-02 Thread Ryan Huff
About 1K of 79xx phones on a ccm 8.X going to ccm 10.5. So CTL/ITL is an issue. 
There are lots of ways to solve this with access to the previous phone system, 
I know.

The catch for me is, I DO NOT have access to the previous phone system or the 
previous phone certificates (would be so much easier if I did, I know).

Anyone have any dark magic I can script though the SSH / Web Server client on 
the phone that doesn't require me to bind to a valid ccm user from the previous 
system or in any other way, reference the previous system?

Any other ideas?

Thanks,

Ryan
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Re: [cisco-voip] FXS voltages / POTS compatibility

2015-03-02 Thread Ryan Huff
Lelio,

You could do that, but you would have to split out the channels on your T1.

Thanks,

Ryan

 Original Message 
From: Lelio Fulgenzi le...@uoguelph.ca
Sent: Monday, March 2, 2015 10:26 AM
To: Ryan Huff ryanh...@outlook.com
Subject: Re: [cisco-voip] FXS voltages / POTS compatibility
CC: Cisco VOIP cisco-voip@puck.nether.net,James Andrewartha 
jandrewar...@ccgs.wa.edu.au

p { margin: 0; }

OK, so have a few FXS ports reserved for alarm outbound calling. That could 
work, but I was hoping to capitalize on the existing PRI connection.

---
Lelio Fulgenzi, B.A.
Senior Analyst, Network Infrastructure
Computing and Communications Services (CCS)
University of Guelph

519‐824‐4120 Ext 56354
le...@uoguelph.ca
www.uoguelph.ca/ccs
Room 037, Animal Science and Nutrition Building
Guelph, Ontario, N1G 2W1


From: Ryan Huff ryanh...@outlook.com
To: Lelio Fulgenzi le...@uoguelph.ca, James Andrewartha 
jandrewar...@ccgs.wa.edu.au
Cc: Cisco VOIP cisco-voip@puck.nether.net
Sent: Monday, March 2, 2015 8:48:20 AM
Subject: RE: [cisco-voip] FXS voltages / POTS compatibility

!-- .hmmessage P { margin:0px; padding:0px } body.hmmessage { font-size: 
12pt; font-family:Calibri } -- 

Lelio,

You could remove the mgcp service from the fxs port, then create your dial 
peers (assuming your alarm only wants tone and doesn't need inward).

Thanks,

Ryan


From: le...@uoguelph.ca
Date: Mon, 2 Mar 2015 08:21:51 -0500
To: jandrewar...@ccgs.wa.edu.au
CC: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] FXS voltages / POTS compatibility

James,


Does this mean you have an h323 gateway? Right now, I have MGCP, which I'm 
guessing, precludes me from doing this. 

Sent from my iPhone


On Mar 2, 2015, at 7:51 AM, James Andrewartha jandrewar...@ccgs.wa.edu.au 
wrote:

With our security systems I have to remove the call manager from the call path 
for the system to complete due to the nonstandard tones they send. From my 
notes on how to configure this:


On the VG224s:


voice class h323 1

  h225 timeout tcp establish 3

voice-port 2/13

  no timeouts

dial-peer voice 23 pots

  service stcapp

dial-peer voice 99 voip

  description h323 direct to voip1 for alarm number

  destination-pattern 13451015

  session target ipv4:10.101.0.5

  voice-class h323 1

  codec g711ulaw

  no vad

dial-peer voice 98 voip

  description h323 direct to voip2 for alarm number

  preference 1

  destination-pattern 13451015

  session target ipv4:10.101.0.6

  voice-class h323 1

  codec g711ulaw

  no vad


On the 2921s:


voice service voip

ip address trusted list

  ipv4 10.100.0.10 255.255.255.255


The AVG is 10.100.0.10, the 2921s are 10.101.0.5 and .6, and dial-peer voice 
23 is for voice-port 2/13. These are GE security panels I think (which their 
MAC OUI confirms).


-- 

James Andrewartha

Network  Projects Engineer

Christ Church Grammar School

Claremont, Western Australia

Ph. (08) 9442 1757

Mob. 0424 160 877


From: Justin Steinberg jsteinb...@gmail.com
Date: Monday, 2 March 2015 2:14 am
To: chris tknch...@gmail.com
Cc: Cisco VOIP cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] FXS voltages / POTS compatibility


Are you using local h323 or sip 'pots' dialpeers to route directly between 
your FXS and T1 port?  Or is call manager in between the call due to MGCP or 
VOIP dialpeers involved in the dialplan ?

I doubt your issue is line voltage, since you can see the call being placed.  
My guess is the DSP is processing the call and causing issues.  I've seem 
alarm boxes use nonstandard DTMF transmission that isn't properly recognized 
by the DSP.

The 2800 supports DSP bypass by default when you route directly between ports 
using POTS dialpeers.   You do need to have properly configured network clock 
configuration.

Can you send a copy of your config along with the output of 'show controller 
t1' and 'show network-clock'

Justin

On Feb 28, 2015 10:33 PM, chris tknch...@gmail.com wrote:

Hey Ryan,


We have a channelized T1 with channels split between voice/data so the voice 
path is TDM. We have a VIC-4FXS/DID and for each of the two ports we have a 
single copper pair with rj11 on both ends, one side going to the FXS port and 
the other is going into alarm panel. The total distance from the 2800 to the 
alarm panel is around 20-30 feet and its a direct run, no 66 blocks or 
anything in between.

Don't know model of the panel (this is another location) 


From what I've read I think the problem is the default idle-voltage the 
VIC-4FXS/DID is only -24V but based on the link I sent in the first email I 
thought this could be reconfigured through the idle-voltage option but this 
doesnt seem to be available when I try to enter it under the voice-port. When 
I talked to the alarm company and told them I see the calls going through the 
guy told me the alarm doesn't check the line state based on the dialtone and 
he said that it uses the voltage to see when the line is idle, ringing, etc 
and I think this is where

Re: [cisco-voip] errors with IOS transcoder

2015-03-03 Thread Ryan Huff
So in order to configure the transcoder, you must first initialize the DSP 
Farm. If you plan to use the dspfarm with your T1 card, then yes, you would 
need the card installed first.

If you are not going to use the dspfarm with your T1 card, then you need to 
initialize the dspfarm on the voice-card that the DSP shows up on (doing a 
show inventory should reveal that to you).

Thanks,

Ryan

Date: Tue, 3 Mar 2015 12:07:25 -0500
From: bhowser5...@gmail.com
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] errors with IOS transcoder

I am getting ready to add a VWIC3 T1 (and PRI) to my 2900 series ISR router 
running 15.3(2) and I'm trying to configure an IOS transcoder, I have PVDM3-64 
(2 32 modules). When I get into the transcode sub configuration it gives me an 
Unrecognised Command error when trying to preference the codecs. What I am 
doing wrong?

router(config)#dspfarm profile 6 transcode 
router(config-dspfarm-profile)#codec ?
% Unrecognized command
router(config-dspfarm-profile)#codec 

I've been reading that I may need to actually have the T1 card in the router 
first but I can't find anything to confirm that. I've been reading here: 
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cminterop/configuration/15-mt/dia-15-mt-book/vc-enh-confr-vgr.html#GUID-2E7DACC4-090A-4E89-BC1B-006E79BF0BC0
 which seems to suggest that PVDM3 might not be able to support transcoding?

Can anyone shed a little light my way?

thank you


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Re: [cisco-voip] FXS voltages / POTS compatibility

2015-02-28 Thread Ryan Huff
Chris,

Can you diagram the connections for me? 

Are the copper pairs swinging off a 66 block before terminating to the alarm 
panel or is there a direct copper path between the fxs port and the alarm? Are 
you using an RJ-11 or RJ-14 configuration?

Could you estimate the copper distance between the termination points?

Is the pstn path for the VG SIP or TOM?

Also, I would be curious to know if the alarm panel is a Simplex Grinnell?

Thanks,

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Re: [cisco-voip] Auto Attendant Web Access Imges Issue

2015-02-22 Thread Ryan Huff
Are talking about attendant console or a unity connection system call handler?

I assume you're talking about attendant console; did you double check your IIS 
setup from the deployment docs?

Thanks,

Ryan

 Original Message 
From: AbdusSaboor Khan saboor.k...@gmail.com
Sent: Sunday, February 22, 2015 02:28 PM
To: Cisco VoIP List cisco-voip@puck.nether.net
Subject: [cisco-voip] Auto Attendant Web Access Imges Issue

Hi,

Can anyone tell me what I am missing here , I have install the Auto
Attendant, as it says if you dont have SQL server willl install the express
version, but install the 32bit version of standard SQL, so far everything
goes well while I am accessing through webpage i am getting everything on
the webpage except no images is coming which shows i am missing something
in setup, so web page without images.

Thanks if someone reply me quick will be great.

Abdul

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Re: [cisco-voip] Cisco unity connection call handler

2015-02-22 Thread Ryan Huff
Not sure which version of CUC you're dealing with but this is a link to the 
System Administration Guide for Cisco Unity Connection Release 10.x

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/administration/guide/10xcucsagx.html

If you can explain the specific issue you're having, this list has a wealth of 
resources that could assist with CUC and system call handlers.

Thanks,

Ryan

From: claitoncam...@gmail.com
Date: Mon, 23 Feb 2015 01:08:34 +
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Cisco unity connection call handler

Anyone with knowledge in unity connection call handler? If you can show me a 
document. I am implementing but I have had some problems.

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