Configuring Dial-Plan in CME
Have to configure the dial-plan in CME version 10
Is ther any kind of timer setting as in Call Manager to wait for all the
digits and to route the call properly
If user dial 0 to reception
if user dial 024 to internal extension
if user dial 02433 to other site going through the voip-dialpeer
Can We do this please help
For sccp phones under Telephony Service for sip phone I guess under voice register global
This is the max time to wait for another digit
timeouts interdigit (telephony-service)
To set the interdigit timeout value for all Cisco IP phones in a Cisco Unified CME system, use the timeouts interdigit command in telephony-service configuration mode. To return to the default value, use the no form of this command.
timeouts interdigit seconds
no timeouts interdigit
Hope it helps
Similar Messages
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Trouble configuring Dial Plan in CallManager 7.1.3
Hello all.
Been trying to make my internal phones to communicate to the PSTN line through a VIC2-2FXO card on my Cisco 2911 router.
Problem is that in past tests, I managed to configure CME to direct PSTN call through an FXO card, but it was a different scenario. I had at the time a Cisco 2801 router with CallManager Express installed on its flash, and an FXO card same as this one.
I could also configure my scenario using SDM and other Cisco tools (graphical) which I can't now, because this Cisco 2911 won't accept SDM 2.5 to be installed.
Anyway, I have my phones communicating with eachother internally, both the IP phones, SIP softphones and 2 analog phones connected to the network through a VG202 voice gateway.
I'm here to ask for your help on the best way, or which steps i have to pass in order to allow my communications to be delivered to the exterior of my organization.
I already tried all that I could find, and understand, such as Route Pattern, Application Dial Rules and several others, but still no luck.
Can anyone help me with some guidelines on what I need to configure, or point me on some good documentation about this?
Thank you in advance.Hi
Since you are using FXO ports, I would normally configure the gateway as a H.323 router. This config should be familiar to you if you have played with CME previously.
Basically you would need carry out these steps:
1) Configure dial-peers pointing to your FXO ports on the gatway (here I'm assuming you want to dial 9 for an outside line, so change the 9 if you use something different - also use whatever voice port numbers you have on your gateway):
dial-peer voice 1 pots
destination-pattern 9T
port 0/0/0
dial-peer voice 2 pots
destination-pattern 9T
port 0/0/1
2) Set the ports to use PLAR to route any incoming calls to an internal extension:
voice-port 0/0/0
connection plar 1234 (set an internal extension where I have 1234)
3) Configure dial-peers pointing to CCM for inbound calls to the number you are sending the calls to:
dial-peer voice 10 voip
session target ipv4:x.x.x.x (CCM IP address)
destination pattern 1234
codec G711a
no vad
4) In CCM, add the gateway as a H.323 gateway. Set:
Device name: IP address of the gateway.
Device Pool: something suitable or default
Inbound Calling Search Space: Set to a CSS including your internal phones, or leave at if you haven't configured partitions
5) In CCM, add a route pattern to match your dial-peer. E.g.
Pattern : 9!
Partition : a partition which your phones have in their CSS, or if you haven't configured CSS/Partitions
Gateway or Route LIst : your H.323 gateway
Leave other properties as-is for now.
This should then allow anything you dial starting with 9 to be routed to the gateway, which will then strip the 9 and send the call out.
This is a very basic setup.... there's a world of other stuff you may want to do, but it should get you started...
Regards
Aaron
Please rate helpful posts -
How do I configure Dial-plan redundancy - Priority
Hello
I am setting up Dial Plan to single Destination PREFIX to 2 seperate gateway (not under my control).
1st Gateway as 8 PSTN line
2nd Gateway has 4 PSTN line
how do I configure my cisco 3620 gateway that will allow me to route calls to 1st Gateway as long as all 8 PSTN lines are available.
I know if you configure preference under dial plan like the one below
dial-peer voice 88021 voip
destination-pattern 8802T
session target ivp4:x.x.x.x
preference 10
dial-peer voice 88022 voip
destination-patter 8802T
session target ipv4:y.y.y.y
preference 9
but this does not work as long as IP network is fine. I need as solution that will allow me to route calls to 2nd gateway if PSTN interfaces are all full/busy
These 2 destination gateways are not under my control therefore, I can't request them to use my gatekeeper.
Any other solution that I can try in my gateway
FaisalThis will work. The clal will be routed to the first gateway with prefernce 9 and then if the call fails, the call will be routed to the second gateway with prefernce 10.
The only trick is the call has to fail with an error message which will tell the gateway to hunt. For example user busy will not cause the gateway to hunt. No cirucit or channle, no resource, no route to destination etc will all make the gateway hunt.
You can on the other hand use advanced busyout on the remote gateways to tell the originating when the interfaces are busy.
Taimoor -
Hi Cisco techs,
I was recently asked this question at work from the rest of my team as they know I am doing studies for cisco ICND 1 & 2.
And couldn't answer them.
Or does this require more information like hardware used etc?
Had searched Google to ambiguous results.
Can someone give me a link to advise please
Thanks in advance again..Dear,
For small business (up to max 450 users) we use Call Manager Express (CME) which is configured on Cisco Voice router.
If no dial plan is used or configured then none of callmanager can do external outgoing/internal calls. Dial plan is basic thing which must be configured on cisco voice router (e.g. CME).
If you have configured Phones (ephone & ephone dn) and if they are registered on callmanager express then for internal calls you dont need to configure dial plan because callmnager have all the information of its phones.
If you want to have external incoming/outgoing calls for CME then you will have to create dial peers/dial plan.
For a CME lab setup, you can visit below link.
http://cisco.jjc.edu/cnt208/PDF/CCNP4_lab_2_1_en.pdf
Suresh -
Time based dial plan configuration
Hello experts!
We're trying to maximize security on our VOIP Gateway to avoid being victimized by long distance/international toll fraud. In efforts to address this concern, we're looking to somehow deploy a time based dial plan on our gateway (2821) which based on that it automatically shutdown outbound international dial peer for any calls made during off work hours (i.e 7pm- 7am including weekends):
dial-peer voice 119 pots
destination-pattern 01T
port 0/0/0:23
forward-digits 16
I understand this can be done inside Callmanager (we're running Callmanager 4.1(3)) as well but for extra security pre-caution we'd like to have it on our gateway preferably. Is there any way to accomplish this task on a 2821 VOIP Gateway?
Thanks,You can use kron command.
This is an example:
kron occurrence NIGHT at 20:00 recurring
policy-list SHUT_DIALPEER
kron occurrence DAY at 8:00 recurring
policy-list NOSHUT_DIALPEER
kron policy-list SHUT_DIALPEER
cli dial-peer voice 119 pots
cli shutdown
kron policy-list NOSHUT_DIALPEER
cli dial-peer voice 119 pots
cli no shutdown
Regards. -
Need a little help with dial setup on CME
I've got a CME I'm using for testing and I think I need a little help figuring out the proper config to get the system to accept numbers I dial and have those numbers be passed on to an Avaya system (including the leading 9 for ARS in Avaya) via H.323 IP trunks. I have it working well for internal 5 digit extension calls across the H.323 trunks and I also have it working well for some types of outside calls that gets passed on to the Avaya and then the Avaya dials the call out to the PSTN. My only real problem is, I can't figure out how to correctly configure CME to examine the digits I'm dialing and only send the digits once I'm finished dialing....not as soon as it sees an initial match.
What's happening is this. I can dial local numbers in my area as 7 digits or 10 digits. The phone company doesn't yet force us to dial area code and number for local calls (10 digits). I can still dial 7 digits. But...if I put an entry in CME that looks like this....
(by the way, the 192.168.1.1 IP is not the real IP address, that's just an example, but the rest of this entry is what I really have entered in CME)
dial-peer voice 9 voip
description Outside 7 Dig Local Calls Via Avaya
destination-pattern 9.......
session target ipv4:192.168.1.1
dtmf-relay h245-alphanumeric
no vad
...Then it will always try to dial out immediately after seeing the first 8 digits I dial (9 plus the 7 digit number I called)...even though I have a speicifc entry in the system that accounts for calls to 9 plus area code 513. I would have assumed that if I put the specific entry in for 9513....... it would see that and wait to see if I was actually dialing something to match 9513....... instead of 9....... Understand what I mean? Because 9513....... is more specific than 9....... but it still tries to send the call out immediately after seeing the first 8 digits I dialed.
dial-peer voice 9513 voip
description Outside 10 Dig Local Calls Via Avaya
destination-pattern 9513.......
session target ipv4:192.168.1.1
dtmf-relay h245-alphanumeric
no vad
...BUT...here's the interesting thing. If I trace the 10 digit call in Avaya, I see that the number being presented to the Avaya PBX is only the first 7 digits of the number....not the full 10 digits...BUT I see a few more of the digits I dialed (like the 8th and 9th digits) after the call is already setup and sent to the PSTN. It's like the CME is trying to send the rest of the 10 digits I dialed, but at that point it's already too late. It setup the call as a 7 digit call (9 plus 7 digits), not 10 digit like I wanted.
I'm more familiar with how to setup dialing in the Avaya via ARS. My background is Avaya, not Cisco, so this dial-peer config is a little difficult for me until I understand the reasoning of how it examines the numbers and what I should do to make it wait for me to finish dialing....or to tell the system that what I'm dialing will be a minimum or a certain amount of digits and maximum of a certain amount of digits, like the Avaya does. I just need some pointers and examples to look at :-) I think I've almost got it....but I'm just missing something at the moment.
Just so you understand, the call flow should be like this: Cisco phone registered to CME > CME to Avaya via H.323 trunks > Avaya to PSTN via ISDN PRI trunks connected to Avaya. I have to be sure I send the 9 to the Avaya also, because 9 triggers ARS in the Avaya.
Thanks for your helpHere is a good document that explains how dial-peers are matched in the Cisco world:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#topic7
In your case, it is variable length dial plan you are trying to implenent. To fix it, you need to add a T to force the system to wait for more digits to be entered if there is any.
dial-peer voice 9 voip
description Outside 7 Dig Local Calls Via Avaya
destination-pattern 9.......T
session target ipv4:192.168.1.1
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 9513 voip
description Outside 10 Dig Local Calls Via Avaya
destination-pattern 9513.......
session target ipv4:192.168.1.1
dtmf-relay h245-alphanumeric
no vad
You can also configure the inter-digits timeout using the command timeouts interdigit under telephony-service.
Please rate helpful answers! -
Hi,
We are in the process of Migrating Cisco CUCM & Voice Gateway (From another vendor to Cisco).
The requirement is all internal calls between Cisco IP Phones & Lync to be flown through CUCM. Means internal extension to extension. Remaining all calls like Mobile, National, International, Toll Free, Emergency, Shared numbers calling to be routed
to Cisco Voice Gateway.
We created the test dial plan, Voice policies, Route and assigned it to couple of user from Lync (2 extensions) and from Cisco side we have taken 2 IP Phones which is pointed to new CUCM. We tested all below scenarios,everything was working fine.
Lync to Lync Call using internal Extension number – Routed through Cisco new CUCM
Lync to Cisco Call using internal Extension number – Routed through Cisco new CUCM
Cisco to Lync Call using internal Extension number – Routed through Cisco new CUCM
Lync to Hotline Numbers (66XX, 68XX Numbers) – Routed through Cisco Gateway
Lync to Shared Numbers starting with 600 (Verified the number 600535353) - Routed through Cisco Gateway
Lync to Emergency numbers & Toll Free Numbers (Not verified the emergency Number as we decided to do it at end) - Routed through Cisco Gateway
Lync to Landline Numbers – Any 7 digit numbers - Routed through Cisco Gateway
Lync to National Numbers – Starting with 3,4,6,7,8 followed by 7 digits - Routed through Cisco Gateway
Lync to Mobile Phones – Starting with 05 contains exactly 10 digits - Routed through Cisco Gateway
Lync to International Numbers – Starting 00 contains at least 11 digits - Routed through Cisco Gateway
All Incoming calls – From Landline, Mobiles, International Numbers - Routed through Cisco Gateway
Call Transfer – To another Lync Extension, Cisco Extension, Landline, Mobiles, International Number
Conference – with another Lync Extension, Cisco Extension, Landline, Mobiles, International Number
Call Forwarding – To another Number, Voice mail
Response Groups
Click to call – As if user try to place a call by directly click the number from Outlook, Websites will be in E.164 format
Dial in meeting – Conference calls are works fine
But when we roll out to the production we are facing issues listed below
1) The phones we used during testing are working which is using same dial plan, Voice policy, Route, PSTN Usage. But from production most of the phones are not working (using the same dial plan, voice policy, Route). Also Problem is only with external calls
as the internal calls are working fine between Cisco & Lync even in production (Routed through CUCM) NOTE: All incoming calls are working fine (From international, local, national, extension)
2) How long its going to take for Lync to push the new voice policies, Dial plans to the Phones?
3) Is there a way to forcefully update the policies, dial plans to the Phone?
4) Also the environment is using over 100 dial plans, so I just copied and pasted the Normalization rules that we tested and working fine. Most of the dial plans are assigned to individual users as every dial plan contains a normalization rule for
international calling with Unique Prefix (Example: User John international Normalization rules says #1234#00#CountrycodePhonenumber, means if John has to place the international call he need to dial #1234# followed by 00 and then country code, then actual
phone number). In this case how long its take for the users / phones to get updated with new dial plans?
6) Is it recommended to use multiple dial plans ? What are the best practices?
5) Also calls are working fine one & failing on subsequent tries. Means when I dial first 1 or 2 times. Call fails, but when I try 3rd time and subsequently it works. After some again there will be failure during 1 or 2 attempts. Why is it so?
6) After updating the dial policies, voice Route, Voice policies If i reboot all the phones from Switch, Will the changes take effect immediately?
7) Also when some one calling from mobile or external number to Lync extensions they cant here any Dial tones or caller tunes? Its working fine when they call Cisco Extensions. Also to Lync its working if we dial in E.164 Format, if we dial like 023XXXXX
format its not working. Any guess about this issue?
Waiting for some one to help,
Best regards
Krishna
Thanks & Regards Krishnakumar BHi,
1. As all incoming call worked normally, please double check outgoing ports for Lync FE Server and Mediation Server.
You can refer to the link of “Ports and protocols for internal servers in Lync Server 2013” below:
http://technet.microsoft.com/en-us/library/gg398833.aspx
2. When an administrator makes a change to Lync Server (for example, when an administrator creates a new voice policy or changes the Address Book server configuration settings) that change is recorded in the Central Management store.
In turn, the change must then be replicated to all the computers running Lync Server services or server roles.
So it may not replication completely immediately.
3. You can run the following cmdlet with Lync Server Management Shell on FE server to
forcibly replicate information to a computer: Invoke-CsManagementStoreReplication
4. As you used over 100 dial plans, it may be the issue of multiple dial plans. Would you please tell us why you created different dial plan for individual user with unique prefix?
5. Multiple dial plans and undue normalization rules may cause call fail. You can double check the normalization rule.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support -
Best practice to Change Dial plan?
Hi,
Customer has made plenty of misdialed 911 calls so they want to change the dial plan. They have CUCM, CUC and UCCX .. I will try to suggest putting a delay for 3 sec or so and blocking 911! or 911!# translation pattern .. but in case if they do want to change their dial out number.. what's the best practice for this? I tried looking for a suggestion or document but couldn't find it... at this point I can only think of copying existing RP's and change the dial out number to 8 and if required if they will have an H.323 gateway then might require configuration on dial peers... Any suggestion on this is appreciated? ThanksHi Vishal,
If 911 is being dialed accidentally, you can try configuring a 9.911 or 911# route pattern. 9.911 will require you to change the destination pattern and forward digit settings on dial-peer or you can actually strip it on cucm itself and will not require a change on dial-peers. Other than that you need to check your dial-plan and see if there are any router patterns that are overlapping with 911 you can try editing them as well by changing the first digit for those route patterns to something other than 9.
HTH
Manish -
Dial plan: Can we change the redirect number (sip-sip)?
Hello Scott Page/Alexei/Dirk Anyone.
I have a question on dial plan:
Call flow: A---cucm----pgw----voicemail
B-----|
Its a sip to sip call, what i need to do is change the redirect number or rather add a digit to the redirect number at the PGW side.
For instance:
A number: 902228990
B number: 902228996
Redirect number: 92228996 --->Here i want to change it to 902228996
pgw patch: 9.8.2 "Patch:"CSCOgs014/CSCOnn014""
bash-3.00$ uname -a
SunOS tcs-tza1 5.10 Generic_127127-11 sun4u sparc SUNW,Netra-440
I checked from the cisco website but i think its only for isup sip calls only but not sure.
numan-add:fullnumbertrans:svcname="Service Name",numbtype="Number Type", digstring="Original Digits",
translatednum="Translated Digits"
Where numtype can be choose 3 for redirect number.
numtype—Identifier for the number type (1-5), it is one of the following values: 1—called party number 2—calling party number 3—redirecting number 4—calling party number and redirecting number 5—original called number
Can some help me out with the dial plan configuration?.
Regards,
AbyGuys,
This configuration would work?, please let me know
1. mml> numan-add:service:custgrpid="DP00",name="BATMANredirect"
2. mml> numan-add:fullnumbertrans:svcname="BATMANredirect",digstring="92228996", translatednum="902228996",numtype="3"
numtype=3-->redirect number
3. Adding a result type of NUM_TRANS:
mml> numan-add:resulttable:custgrpid="DP00",name="results",resulttype="NUM_TRANS", dw1="BATMANredirect",dw2="3",dw3="3",setname="setname3"
dw2--->number type hence i put it 3
dw3--->NOA--->hence i put it as national ie 3,
dw4--->Dialplan?-- can i avoid this?
4. mml> numan-add:resulttable:custgrpid="DP00",name="noar",resulttype="R_NUMBER_TYPE", dw1="4",setname="setname3"
dw1===>NOA again, i put it as dw1="4" as national.
Can any of you help me if this would do the trick or anything else needs to be added?.
Thanks and appreciate your time.
Regards,
Aby -
911 Dial Plan for Specific Users
Hi all, We have a Lync 2013 deployment here.
I am trying to achieve something that I am not sure is even possible with 911 calling.
Out SIP provider is telling us that we MUST send all calls outbound as a specified number (let's use 555-4040 in this example).
What I would like to achieve, is to configure an outbound translation ONLY when a user dials "911". All other calls will go outbound as the user's regular DID number.
Is this even possible with the Lync dial plans/translations?
If not, then it appears we must ALWAYS translate outbound calls as "555-4040" no matter what number the user dials.
Thanks!Yes, you can do it. You have to create a new voice route for 911 calls only. When you create a route, there is an option to "Suppress caller ID". This is the place where you enter the caller ID that you want to have.
Another way to do it directly on your IP gateway. I have done it before on AudioCodes and on Dialogic gateways. You create a new manipulation rule and specify that if dialed number is 911, replace the caller's number.
Please “Vote As Helpful” and/or “Mark As Answer” if this post helped you. -
Analogue devices not using assigned dial plan
Hi,
We are using Tenor gateways for running analogue devices (fax, cordless phones) through Lync 2013. It works fine on our standard dial plan which all end-users are on. However, for a business reason, we need to do restricted dialing on the cordless phones
(e.g. dial a 5 digit access code to dial out).
The cordless phones exist in AD (created using new-csanalogdevice in the Lync Mgmt Shell) and I'm using the following command to assign the dial plan:
get-csanalogdevice "My test phone" | grant-csdialplan -policyname "restrictedCallDialPlan(77777)" -v
When I run get-csanalogdevice "My test phone" | Select DialPlan I can see I am using the restricted dialling dial plan but the calls are not restricted - e.g. I can call my mobile phone as normal with out having to preceed the number with
77777.
First thought was there was a configuration an issue with the dial plan. Checking it in the Lync Ctrl Panel it all looks fine, and it more importantly, it works as expected on Common Area Phones (Polycom CX500).
If I have to, I can do the restriction on the Tenor gateways but it's not a "nice" solution as the config gets messy. Ideally we'd like to do it in Lync so we can manage it all from one place.
Is anyone familiar with assigning dial plans to analogue extensions in Lync and know of a reason this wouldn't work?
Many thanksHello,
I experienced the same behaviour utilising analogue devices within Lync, during my research it appeared that while you can set a dial plan against an analogue device, this will never take affect and
will only inherit normalisation rules defined in the Global dial plan. Please see the following article below under the "Call Routing For Analogue Devices" section, around half way down this section it explains in details the grant-csdialplan behaviour
for an analogue device. In my case, I utilised a feature on the AudioCodes MediaPack devices which we were using for analogue connectivity in order to restrict dialling on a per port basis.
http://www.mylynclab.com/2013/04/microsoft-lync-facts-about-fax.html
Regards.
http://www.b4z.co.uk -
AA unable to transfer to Lync 2013, but only on User dial plans
The current environment has two Lync 2013 standard edition servers. Lync 2010 is still in existence, and both pools are connected to an Exchange 2010 SP2 UM server.
UserA is part of the default Global dial plan in Lync and part of a UM dial plan named Global. UserA has been moved to Lync 2013 pool.
UserB is part of a user dial plan in Lync named UserDialPlan and a UM dial plan named UserDialPlan. UserB has been moved to Lync 2013 pool.
Both dial plans have their own route/pstn usage tied to their own sip trunk from Intelepeer. Both UM dial plans have an AA identically configured with key mapping #1 tied to dial the 11 digit phone number of the user. Both users can successfully
be dialed directly via PSTN.
If I call Global’s AA and press 1, the call is successfully connected and rings through to UserA.
If I call UserDialPlan’s AA and press 1, I get “Sorry I couldn’t transfer you to the extension” and the UM server logs shows errors 1079 and 1136:
1079 The VoIP platform encountered an exception Microsoft.Rtc.Signaling.OperationFailureException: Failed to transfer, successful refer notification not received
1136 An error occurred while transferring a call to "15552735555". Additional information: The call transfer type is "Blind.", the transfer target is "phone number", and the caller
ID is: "0ed8114d-a068-41be-9790-9342d0a02d7b".
If I switch the ip address of our sip trunk back to the Lync 2010 mediation server and alter the Lync topology to connect that trunk back to the 2010 server, then these transfers start working again. This goes for other
user dial plans as well. What I don’t understand is why the Global dial plan would work on 2013 and the user dial plans will not. Refer is disabled on the trunk config (has been for
over 2 years). I assume if it was a general setting like that, the Global plan wouldn’t work either, but what is it that’s special about the Global plan vs. user plans?
Any thoughts would be great, thanks!I have the same issue. I can actually transfer to extensions when it is speech enabled and I say the person's name. However if I try a key mapping to transfer to the same user using the Extention I get the same "Sorry I couldn't transfer you to the extension"
An error occurred while transferring a call to "151". Additional information: The call transfer type is "Blind.", the transfer target is "phone number", and the caller ID is: "4813feae-2a10-4d33-9612-ee4bccf7c0f9".
The VoIP platform encountered an exception Microsoft.Rtc.Signaling.OperationFailureException: Failed to transfer, successful refer notification not received
at Microsoft.Rtc.Signaling.SipAsyncResult`1.ThrowIfFailed()
at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult result)
at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult result, String operationId)
at Microsoft.Rtc.Collaboration.Call.EndTransferCore(IAsyncResult result)
at Microsoft.Rtc.Collaboration.AudioVideo.AudioVideoCall.EndTransfer(IAsyncResult result)
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.BlindTransferSessionState.Call_TransferCompleted(IAsyncResult r)
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.SubscriptionHelper.<>c__DisplayClass5f`1.<>c__DisplayClass62.<WrapCallback>b__5e()
at Microsoft.Exchange.UM.UcmaPlatform.UcmaCallSession.<>c__DisplayClassd.<CatchAndFireOnError>b__9()
Detected at System.Environment.get_StackTrace()
at Microsoft.Rtc.Signaling.OperationFailureException..ctor(String message)
at Microsoft.Rtc.Collaboration.Call.CallTransferAsyncResult.Refer_StateChanged(Object sender, ReferStateChangedEventArgs e)
at Microsoft.Rtc.Signaling.ReferStateChangedEventArgs.Microsoft.Rtc.Signaling.IWorkitem.Process()
at Microsoft.Rtc.Signaling.WorkitemQueue.ProcessItems()
at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessing()
at Microsoft.Rtc.Signaling.SerializationQueue`1.ResumeProcessingCallback(Object state)
at Microsoft.Rtc.Signaling.QueueWorkItemState.ExecuteWrappedMethod(WaitCallback method, Object state)
at System.Threading.ExecutionContext.Run(ExecutionContext executionContext, ContextCallback callback, Object state)
at System.Threading._ThreadPoolWaitCallback.PerformWaitCallbackInternal(_ThreadPoolWaitCallback tpWaitCallBack)
at System.Threading._ThreadPoolWaitCallback.PerformWaitCallback(Object state)
FailureReason = 0 during the call with ID "4813feae-2a10-4d33-9612-ee4bccf7c0f9". This exception occurred at the Microsoft Exchange Speech Engine VoIP platform during an event-based asynchronous operation submitted by the Unified Messaging server. The Unified
Messaging server will attempt to recover from this exception. If this warning occurs frequently, contact Microsoft Product Support. -
Remove and install dial plan installer - the effects on route patterns etc
Hi All,
I am hoping someone could answer this one for me quickly.
I have a scenario during an refresh upgrade to CUCM 10.5(2) where it fails in the install logs with "component_install:807, failed to refresh_upgrade Infrastructure_post components | <LVL::Critical>".
The fix to this may be that I need to uninstall the AUNP dialplan before upgrading.
See (https://supportforums.cisco.com/printpdf/12334191)
What is the effects on route patterns etc when removing the AUNP dial plan?
Will the configuration be maintained and become operable once I upgrade and re-install the AUNP?
Thanks in advance
KentCheck your database replication status, this might help:
https://supportforums.cisco.com/docs/DOC-13672
HTH,
Chris -
UC320W Problems with the dial plan
Problems with the dial plan ... There is a method to modify the External Dial Plan, we can not dial some numbers (800 XXX XXX, 115, 118, 187, etc., are special numbers in Italy), happens on FXO lines and SIP,in external.
Example: The operator enters "on any phone" 0 (for outside line) then 800 000 000,on the composition of the second 0, the phone responds "invalid number" and does not allow the call
We think it is a problem of Italy dial plan, you can change them?,
our system is configured as a region of Italy and Italy dial plan
is a known problem? you know to help me resolve itPerfect, we wait for the update, this happens also for other special numbers (187, 191 customer service phone numbers and 118, 112, 115 emergency numbers). It happens with any line it is FXO, SIP, Shared. The problem is in the dial plan, the numbers are not included, the system does not accept them, and blocks the composition. Exists a method for change the file loaded into the configuration page "region" the system is set to Italy, I have seen that you can import the file from the computer, you can download the file "Italy", edit, and upload in the system '.
We are a telecommunications company, we install telephone systems, we would like to offer your product the 300 series our clients. We are testing in our lab your UC320W. -
Need UC320 Region Pack or dial plan modification for Egypt
We have bought a UC320W device from a Cisco local distributer in Egypt, but we are facing a problem with the configuration as the device doesn't have a Region Pack for Egypt so we used some other country's Region Pack, the problem with this configuration is that it only supports dialing external land line as it doesn't support dialing mobile phone or International numbers, is there a Region Pack for Egypt? Or a way to modify a dial plan?
Thanks.Hi Friedrich,
To answer your question about the authorized local Cisco Distributor, we have bought the UC320W from Metra Computer, I've configured everything correctly using the Brasil Region Pack except for the time-zone, each time I change the time on the phone-set (SPA504G) it reverts after a little while to the time set by the UC320W because it's being provisioned by it, I have found the ChangeNTPserver.pmf among my exploration, so I guess it's possible to provide a .pmf file that can add the GMT+2 time-zone to the Brasil Region Pack or disable the time-zone provisioning, can you help me with that? How can I request such support task adequately?
Thanks and regards.
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