do C20 to call outside 8.6 CUCM

Hi Experts,

I have in my project C20 telepresence and 8.6 CUCM, I want to be able to call remotely, and I want to know what I have to do to 8.6 CUCM and voice gateway if needed. Some people told me I should do NAT, but I don't want to do NAT. I want to make it through by CUCM 8.6. And what is the possibility to appeal through URI in CUCM 8.6.

Thank you for all

Mohammad Saeed

For voice calls, just register your C20 against CUCM and it'll work just like any standard Cisco VoIP phone. (And so use whatever the gateways/SIP trunks you got set for the external connectivity)

If you want to make video calls outside CUCM, then you must make some changes to the CUCM. Basically, you'll want to be able to make/receive the SIP (or maybe H.323) calls on the ' net. Exactly what are the changes will depend on what you want to do, but my money is on that you need to have somewhere of the VCS in the mixture.

GTG

Tags: Cisco Support

Similar Questions

java.lang.IllegalStateException: setPosition called outside the layout

Hello guys '

Good day. I try to spend PopupScreen and posted about this before.

But I can't solve. You can see what I do from here.

At the request of work, throws exception: java.lang.IllegalStateException: setPosition called outside the layout

I use following code:

import net.rim.device.api.ui.component.*;import net.rim.device.api.ui.container.HorizontalFieldManager;import net.rim.device.api.ui.container.MainScreen;import net.rim.device.api.ui.container.PopupScreen;import net.rim.device.api.ui.*;import net.rim.device.api.system.Bitmap;import net.rim.device.api.system.KeyListener;

public class CursorField extends UiApplication {

  private InfoPopupScreen progressPopup;    int x = Graphics.getScreenWidth() / 3;    int y = Graphics.getScreenHeight() / 2;

  public CursorField() {

      ManagedScreen managedScreen = new ManagedScreen();        pushScreen(managedScreen);    }

  private class InfoPopupScreen extends PopupScreen implements KeyListener {

      protected void applyTheme() {         //Nothing     }

      public boolean keyStatus(int keycode, int time) {         return false;     }

      public boolean keyRepeat(int keycode, int time) {         return false;     }

      public boolean keyUp(int keycode, int time) {         return false;     }

      public InfoPopupScreen(Manager manager){          super(manager);       }

      protected boolean navigationMovement(int dx, int dy, int status, int time) {

          return true;      }

      protected void paintBackground(Graphics graphics) {           graphics.setBackgroundColor(16777215);        }

      public void sublayout(int width, int height){         super.sublayout(width, height);           setPosition(x, y);        }

      public boolean keyDown(int keycode, int time) {

          char key =  net.rim.device.api.ui.KeypadUtil.getKeyChar(keycode, net.rim.device.api.ui.KeypadUtil.MODE_UI_CURRENT_LOCALE);

          //Letter 'I'          if(key == 73) {               x = x + 10;               try {                 this.setPosition(x, y);              } catch(Exception e) {                    System.out.println("Exception : " + e.toString());                }         }

          if(key == net.rim.device.api.system.Characters.ESCAPE) {              this.close();         }         return true;      }

      public boolean keyChar(char key, int status, int time) {          return false;     }

      public boolean trackwheelRoll(int amount, int status, int time) {         return false;     }

      public boolean trackwheelUnclick( int status, int time ) {            return false;     }

      public boolean trackwheelClick( int status, int time ) {          super.trackwheelClick(status, time);          return true;      } }

  private class ManagedScreen extends MainScreen {      private BitmapField _cursorField;     private Bitmap _bitmap;

      public ManagedScreen() {

          _bitmap = Bitmap.getBitmapResource("cursor.png");         int[] a = new int [10 * 10];          _bitmap.getARGB(a, 0, 10, 0, 0, 10, 10);

          _cursorField = new BitmapField(_bitmap) {             public void paint(Graphics graphics) {                    graphics.setBackgroundColor(16777215);                    super.paint(graphics);                }         };

          HorizontalFieldManager manager = new HorizontalFieldManager(USE_ALL_WIDTH) {              public void paint(Graphics graphics) {                    graphics.setBackgroundColor(16777215);                    super.paint(graphics);                }         };

          manager.add(_cursorField);            progressPopup = new InfoPopupScreen(manager);

          Bitmap pic = Bitmap.getBitmapResource("background.jpg");          BitmapField picField = new BitmapField(pic);          add(picField);

          ButtonField _closeButton = new ButtonField("Push screen", ButtonField.CONSUME_CLICK | ButtonField.FIELD_HCENTER);         _closeButton.setChangeListener(new FieldChangeListener() {                public void fieldChanged(Field field, int context) {                  pushScreen(progressPopup);                }         });

          add(_closeButton);        } }

  public static void main(String[] args) {      CursorField application = new CursorField();      application.enterEventDispatcher();   }}

I don't know what is the problem here.

Please help me guys

TNX in advance"

You can only call layout() setPosition or sublayout().

Try this...

Subclass sublayout() and put your good standing in position one. Once you update your x / call updateLayout().

Call outside the component
I have a mx:Repeater that repeats the same custom component repeatedly
How can I dynamically id and call an effect outside the effect.
The component contains a HBox I need to hide, but I don't know how to call outside the component of
If you give an element in a repeater an id, you can access instances repeated this element using array notation:
myId [n], where "n" is the index of the item dataPrvoider.

For a repeated complex content, I recommend create a custom with all controls component that it and repeat it. Pass the currentItem all. I have posted examples of code repeatedly, try seraching for it if you wish.

Tracy

Outgoing Caller ID question using CUCM 8.6, MGCP gateway & PRI lines

Hello

We have two ranges DID with our provider: 02825911XX & 02829212XX

02825911XX maps extension 11XX

02829212XX 12XX expansion cards.

We have 2 E1 lines 60 channels.

Current configuration:

CUCM <----MGCP---->Gateway (8.6.2) (2921) <======2 e1="====">carrier / Teleco.

Question:

When calling 11XX we get correct caller id 02825911XX.

But when the 12XX we receive caller ID 02825911XX instead of 02829212XX.

for example: when calling 1206 far end caller id watch 0282591106 instead of 0282921206.

No problem in incoming calls on the two beaches.

The external mask are configured correctly in the domain name.

The debug ISDN q931 shows the correct caller until the gateway ID:

ISDN Se0/0/0:15 Q931: TX-> INSTALLATION pd = callref 8 = 0x016A
Complete package
Carrying capacity I = 0x8090A3
Standard = CCITT
Ability to transfer = speech
Circuit transfer mode
Transfer rate = 64 kbit/s
The channel ID I have = 0xA98396
Exclusive, Channel 22
Display i = 'test '.
Calling party number i = 0 x 0081, '0282921206'
Plan: Unknown Type: unknown
Called number i = 0 x 80, '82353587'
Plan: Unknown Type: unknown

Fixing ccapi inout & ISDN q931 debug. & Gateway config.

Hello Gyanendra,

I checked the configuration debug and gateway.

These debugging come from gateway. The catwalk shows TX - Setup. Once we have sent installation information to the telephone company, I don't think that we can change the ID calls information. This means that we send correct calling Telco part number information (0282921206).

May be that Telco is changing the number on their own party? Have you checked with Telco yet? If not, ask them what is the number of parties calling they get. You can do a live test and track same call with the telephone company engineer.

The details of trace log:

00:49:25.410 Sep 10: ISDN Se0/0/0:15 Q931: TX-> INSTALLATION pd = callref 8 = 0x016A

Complete package

Carrying capacity I = 0x8090A3

Standard = CCITT

Ability to transfer = speech

Circuit transfer mode

Transfer rate = 64 kbit/s

The channel ID I have = 0xA98396

Exclusive, Channel 22

Display = 'Ludvík Aunedi' i

Calling party number i = 0 x 0081, '0282921206'

Plan: Unknown Type: unknown

Called number i = 0 x 80, '82353587'

Plan: Unknown Type: unknown

00:49:26.222 Sep 10: ISDN Se0/0/0:15 Q931: RX<- call_proc="" pd="8" callref="0x816A ">

The channel ID I have = 0xA98396

Exclusive, Channel 22

00:49:26.222 Sep 10: ISDN Se0/0/0:15 Q931: RX<- alerting="" pd="8" callref="0x816A">

Sep 10 00:49:26.258: / / 6444, 9E3008C18C96, CCAPI, ccCallModifyExtended:

Numerator = 0x2B305B60, Params = 0x2B304D78, Id = 6444 Call

Sep 10 00:49:26.258: / / 6445, 9E3008C18C96, CCAPI, ccCallModify:

Numerator = 0x18E30, Params = 0x2B304F80, Id = 6445 Call

Sep 10 00:49:26.258: / / 6444, 9E3008C18C96, CCAPI, cc_api_call_modify_done:

Result = 0, = 0x2AF52C6C, Id = 6444 Call Interface

Sep 10 00:49:26.262: / / 6445, 9E3008C18C96, CCAPI, cc_api_call_modify_done:

Result = 0, = 0x2AD55F80, Id = 6445 Call Interface

Sep 10 00:49:29.406: / / 6444, 9E3008C18C96, CCAPI, cc_handle_inter_digit_timer:

Generate inter-chiffre timeout CC_EV_CALL_DIGIT_END event

00:49:35.262 Sep 10: ISDN Se0/0/0:15 Q931: RX<- connect="" pd="8" callref="0x816A">

00:49:35.266 Sep 10: ISDN Se0/0/0:15 Q931: TX-> the CONNECT_ACK pd = callref 8 = 0x016A

Also, can you do another test call and join:

Debug mgcp packet with debug ISDN q931 and CCAPI inout

Kind regards

Amarjeet

My Variable is not consistent when called outside the Group
Dear all,

I stored a variable that was within the group as follows:
<? for-each-group: MarketingFundTransferOut; / FromFundId? >
<? sorting: current-group () / FromFundId; ' ascending '; data-type = "text"? >
<? sum (ATTRIB_01 [.! = "])? >
<? xdoxslt:set_variable ($_XDOCTX, 'INIT3', xdoxslt:to_number (sum (ATTRIB_01 [.! = "])))? >-> store for the variable INIT3
<? end for each group -? >

and he asked outside the group with these:

<? xdoxslt:to_number(xdoxslt:get_variable($_XDOCTX,'INIT3'))? >

Sometimes the true result and sometimes the wrong result (non-compliant)

Here is the example of one is not consistent:

1.2 regular TV
In Total total
10 0 10 (I've posted the out) 10 (I've posted the in)

Does anyone have an idea to solve this problem? It will be highly appreciated

Best regards
TSE
You can initialize this variable before this group, so that it does not have the previous value of groups

CUCME no calls incoming, outgoing calls okay

Hello everyone,

I'm setting up a CUCME with SIP trunk, I can make calls outside, but I can´t receive everything from the outside, it's my second time as a SIP configuration

I ve use debug command voice dialpeer all to check was happening, but I can´t find the problem.

This is my config:

IP server host sip - A.B.C.D

!

voip phone service

list of approved IP addresses

IPv4 A.B.C.D 255.255.255.252

!

translation of the voice-rule 1

rule 1 / 325277\ (\) / / 1\1 /.

!

voice translation-profile IN

translate 1 called

!

Dial-peer voice 1 voip

Description * incoming SIP trunk call *.

entrants IN translation-profile

session protocol sipv2

session target sip-Server

incoming called-number.

codec voice-class 1

voice-class sip dtmf-relay rtp - nte force

DTMF-relay rtp - nte

No vad

!

ePhone-dn 1

number 100

Description of RECEPTION

!

ePhone 2

address Mac YYYY. BENAMER. CCBC

ePhone-model 1

type 7942

Keep-Conference

button 1:1

NOTE: The IP address are hidden, just for safety

Here is the output from my debug/tests:

voice translation rule 1 32527700 #test

Matched with rule 1

Original number: 32527700 translated number: 100

Number of origin type: no number translation type: no

Original number plan: no number plan translated: no

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Number = 32527700, called number = 32527700, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Associate the rule of = DP_MATCH_DEST; Called number = 32527700

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Dial String = 32527700, expanded String = 32527700 number = 32527700T

Timeout = TRUE, incoming = FALSE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

No outbound dial-peer does not; Result = NO_MATCH(-1)

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

dialstring = 32527700, saf_enabled = 1 saf_dndb_lookup = 1, dp_result =-1

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

Result = NO_MATCH(-1)

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Number = 59513212, called number =, Voice-Interface = 0 x 0.

Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

Peer Type Info = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_ANSWER; Number = 59513212

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls = 59513212T

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_ORIGINATE; Number = 59513212

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls = 59513212T

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Result = NO_MATCH(-1) after all rules attempt Match

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Number = 59513212, called number = 32527700, Voice-Interface = 0 x 0.

Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

Peer Type Info = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_VIA_URI; URI = SIP:A.B.C.D:5060

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls =

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_REQUEST_URI; URI = sip:[email protected]/ * /: 5060; user = phone

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls =

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_TO_URI; URI = sip:[email protected]/ * /; user = phone

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls =

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_FROM_URI; URI = sip:[email protected]/ * /; user = phone

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Dial String =, expanded string =, number of calls =

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Result =-1

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Associate the rule of = DP_MATCH_INCOMING_DNIS; Called number = 32527700

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

Incoming = TRUE, expand = FALSE

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

Dial String = 32527700, expanded String = 32527700 number =

Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/MatchNextPeer:

Result = Success (0); Incoming dial-peer = 1 is set in correspondence

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

Result = Success (0) after DP_MATCH_INCOMING_DNIS; Incoming dial-peer = 1

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchSafModulePlugin:

dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerSPI:[email protected]/ * /.

Can someone help me?

Thanks in advance!

I looked on the other leg of the SIP messages appeal, here's the fault for the where incoming call is being failed because the session timer is too small, has received from the SBC (provider)

Call ID:

Call ID: [email protected]/ * /.

INVITE RECEIEVED SBC - SIP

* Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

GUEST sip: 32527700 @(WAN): 5060; user = phone SIP/2.0

Via: SIP/2.0/UDP (SIP_SERVER): 5060; Branch = z9hG4bK776928550196f0d843ca0b092

Call ID: [email protected]/ * /.

From:; tag = 6e8b9968-CC-25

TO:

CSeq: 1 INVITE

Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see

Max-Forwards: 70

Supported: 100rel, timer

User-Agent: Huawei SoftX3000 V300R601

Session time-out: 300

Min - SE: 90

Contact:

Content-Length: 376

Content-Type: application/sdp

v = 0

o = HuaweiSoftX3000 4507886 4507886 IN IP4 (SIP_SERVER)

s = call Sip

c = IN IP4 (SIP_SERVER)

t = 0 0

m = audio RTP 11554 / AVP 8 0 18 4 2 98 98 98

a = rtpmap:8 PCMA/8000

a = rtpmap:0 PCMU/8000

a G729/8000 rtpmap:18 =

a = rtpmap:4 G723/8000

a = rtpmap:2 G726-32/8000

a = rtpmap:98 G726-40/8000

a = rtpmap:98 G726-32/8000

a = rtpmap:98 G726-24/8000

a = ptime:20

a = fmtp:18 annex b = No.

In response to GUY sends 422

Envoy:

SIP/2.0 422 Session Timer too small

Via: SIP/2.0/UDP (SIP_SERVER): 5060; Branch = z9hG4bK776928550196f0d843ca0b092

From:; tag = 6e8b9968-CC-25

Up to:; tag = 4CD1E84-2094

Date: Wednesday, January 29, 2014 22:53:19 GMT

Call ID: [email protected]/ * /.

CSeq: 1 INVITE

Allow-events: telephone-event

Min - SE: 1800

Server: Cisco-SIPGateway/IOS-15.2.4.M

Content-Length: 0

According to rfc

If the Session time-out interval is too low for a proxy (i.e., lower)

that the value Min - SE that the proxy would argue), the

Proxy denies the request with a 422 response. This response

contains a header field in Min - TO identify the minimum session

meantime, she is ready to support. The UAC will try again, this time

including the header of Min - SE in the query field. The header field

contains the largest header field Min - SE that he observed in all 422

responses received previously. In this way, the minimum timer meets the

constraints of all proxies on the way.

http://www.Cisco.com/en/us/docs/iOS/voice/SIP/configuration/guide/sip_cg-msg_tmr_rspns.html#wp1056968

Response message 422

If the value of the Session header expires is too small, the UAS or proxy refuses the call with a response message 422 Session Timer too small . With 422 response, the proxy or the SAMU message includes a header of Min - SE, indicating the value of minimum session, he can accept. UAC can then try again the appeal with a higher value of session timer.

If a 422-response message is received after a GUEST query, the UAC can again INVITE him.

There is two way to fix this

1) asked the SBC (your SIP provider) the value change and the value of standards send the SIP invite session expires

(2) change the value of the Min - SE on the CME on demand

Run this Global Config on CME

voip phone service

allow sip to sip connection

SIP

90 min - to

BR,

Nadeem

Please note all the useful post.

Capture of calls by the digital code of Cisco IP Phones 3905 / 6921 / 7942 / 7962 (CUCM 8.6)

Hi team.

It is possible set up a calling code collection in CUCM 8.6 for users of IP Phone 3905, 6921, 7942 and 7962 to capture calls that belong to a pickup group call?

This means that the user enter a digital code and automatically captures the call without having to press the buttons on the phone?

Thank you.

FOR EXAMPLE

Yes you can, by dialing the extension of the phone that you can able to pick up that call.

Internet of Jabber Clients through VCSe calls

Hi Experts,

We have provisioned on CUCM 8.6.2 Jabber clients and Movi customers supplied on VCS control (X 7.0) and TMS (13.2.1)

We will deploy VCSexpressway soon to our society.

My request is that it will be possible for the Jabber Clients on CUCM to call outside our company. I know Movi would work, but not sure of Jabber clients.

Could someone help me? If it can work, how it is possible?

Thank you

Saurabh

Slim,

It should work very well. !! Once you make a call to jabber client that calls flow would be to track Express VCS de VCS control and then again in venture capital control of CUCM on the sip trunk.

the only thing is that you must develop with the numbering plan so that it should not conflict with other numbering on the vcs control plans.

Rgds,

Alok

Disable the redirection of call to landline when SNR is used

Hi all

I am looking for a solution for the following request:

A call is redirected to mobile phone in number SNU. When the user hangs up the call on mobile, the CUCM then sends the caller on the desk phone, where the call is for a few seconds. The caller hears Ministry of health during this period. => Works as expected.

Is it possible to disable this redirection to the desk phone. If the called party hangs up the call on the mobile, then the CUCM should end the call.

Thank you

BR
Axel

Have you tried to go to the user and Changingn page

Maximum waiting time for the office collection

Change the Authcode sequence in CUCM

Hello

I need a customer experience with CUCM match. Someone wants access PSTN should make up a code * 12A authcode then PSTN number.

I CUCM auth code always required after matcing routing model. Is it possible to change this sequence in CUCM?

Concerning

Saurabh

Hi Slim,

There is no option to have a FAC or CMC before dialing the number you are calling, please write in CUCM.

HTH

Manish

Time and calendar on the connection of the unit on CUCM

Hello

Today, it's Christmas and Friday, I configured delay on CUCM of identity week and holidays. However, today is Friday and holidays as well. Is there anyone who knows how to give a higher priority to route the call to CUC on CUCM vacation? When the call is transferred to the unit connection, I was faced with the same problem. How to give a higher priority to route the call to wish holiday? In other words, when the day is the day of the week and holidays. Holiday will all first check that the days of the week on CUCM and CUCM.

Thank you

Eric

don't think that there is adjustment to set the priority on the calendar, but if one defines a period of time with a date points, to this date, this period overrides other periods that are set on a weekly basis.

consult the SRND for clarity on the behavior at the time of the period

Behavior of time period

If you set a period of time with a specific date, at that date, this period overrides other periods that are set on a weekly basis.

Example of

Consider the following example:
- There is a time afterofficehours, which is defined as 00:00 and 08:00 Monday to Friday.
- There is a time, newyearseve, which corresponds to 14:00 and 17:00 on 31 December.
In this case, on 31 December, the period of afterofficehours will not because it is overridden by more specific newyearseve period.

Timetables

A calendar includes a group of defined periods that associates the administrator. Once the administrator has configured a period of time, the time period appears in Configuration window when the calendar in the list of available time periods. The administrator may choose a period of time and add it to the selected list of time periods.

Note

After the administrator selects a time for connection with a calendar, the deadline is available for binding with other calendars.

Once the administrator has configured a calendar, the administrator can use the window of the Configuration Partition to select the time zone of the original unit or any specific time zone to a defined schedule. The time zone selected gets checked against the calendar when the user places the call.

The Time-of-Day function filter the CallingSearchSpace string in time settings defined for each partition in the CallingSearchSpace.

After the time of routing is configured, if the time of an incoming call is within one of the time periods in the calendar, the partition is included in the list of partition filtered for the call.

Examples

You can set the USAholidays calendar as the Group of the following time periods: newyearsday, presidentsday, memorialday, independenceday, laborday, thanksgivingday, christmasday. The administrator must first configure the applicable time limits.

You can set the time schedule library_open_hours as the Group of the following time periods: Mon_to_Fri_hours, Sat_hours, Sun_hours. The administrator must first configure the applicable time limits.

http://www.Cisco.com/c/en/us/TD/docs/voice_ip_comm/CUCM/Admin/9_0_1/CCMS...

VINET.

-The rate of useful messages.

No Audio with call transfer to the CUE Script

Hello

I have a CUE script set up to dial several extensions and transfer them to a conference MeetMe hosted on the same 2911 router where the CUE ISM is installed. The call flow is:

CUCM 2911 <---> <--->CUE

with calls to the script being initiated by phones registered with the CUCM.

Dial the pilot script, calls to the specified extensions are undertaken and transferred to the DN MeetMe (7070) successfully, but there is no sound on calls. The script seems to complete successfully as well, because all of the extensions as well as the original one is transferred to the DN MeetMe. The command "display the compact active voice call" shows all participants connected to 7070 and the output of 'see the ephone-dn conf' displays the number of active sessions to 7070

Direct calls to the DN MeetMe (without going through the script) work very well.

Logging in the atrace.log seems to show that reactivate the QUEUE calls will fail. I tested with all three SIP call transfer methods CUE without result.

CUE and 2911 configs as well as the atrace.log CUE file is attached.

Any ideas would be very appreciated.

Hi Miroslav,

You can check the method of transfer in CUE - please define Bye-also and without h450 service of voip telephony services in the CME.

HTH,

Alex

Calling ID mapping

Can anyone give some information about the format of the caller ID card? How exactly does the replacement number formatting work? Also, is it possible to have the display shows something like a comma between the call outside pick number and entering the phone number?

Hello

[Q] can someone give some information about the format of the caller ID card? How exactly does the replacement number formatting work?

[A] sure. See for more information telephone administration Guide:

Direct link: http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa_wip_admin.pdf

List of related documents to: https://www.myciscocommunity.com/docs/DOC-2148

Caller ID card allows you to change what you see on the SPA phone during a call for you.

For example: a person makes a call on your phone in the SPA and you look at the screen of phone of the SPA and see something like 14445551212

You decide that you will not change your dial plan, but I would like to use this information to remind the appellant using a number of management 3. Normally, you would change your dial plan or change the number in your call history, then press the dial. You can use Caller ID card as follows:

1 look to see what your SPA phone displays on the incoming call: 14445551212 in this example

2. observe which line on the SPA phone the call comes on: Ext 4 in this example

3 change the configuration of the phone:
a. SPA phone web-interface user/admin/advanced > Ext 4 > Dial Plan > Caller ID card: (<1444:31444>xxxxxxx)

b. click on submit all changes

The phone does not restart.

4 test by calling your phone in the SPA of 14445551212.

5. the SPA phone will sound and the display will now show 314445551212

[Q] in addition, is it possible to have the screen to show something like a comma between the outside call choose the number and the incoming phone number?

[Has] Yes. Suppose that you want to change an incoming call on the phone in the SPA of 14445551212 to 3-1 444-5551212

1 SPA phone web-interface user/admin/advanced > Ext 4 > Dial Plan > Caller ID card: (<1444:3-1,444->xxxxxxx)

2. click on submit all changes

The phone does not restart.

3 test by calling your phone in the SPA of 14445551212.

4. the SPA phone will ring and displays now 3-1 444-5551212

Enjoy,

Patrick

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IPCC can pass the caller to a mobile phone ID?

I have a script that during off-peak hours, transfer callers to a mobile phone that is answered by a technician. I have a request to the technician to have ALI on the mobile phone. Demand occurred because he missed a call from our ISP and when he tried to call the missed call the number back, he saw was our main number and no voice message was left.

Is this possible? If so, how could I do it?

I know that we have currently to override Caller ID with our main number of the "call transform mask party" on the browse list.

Any help would be appreciated.

Thank you.

You can set your MAC to achieve this. There are two options:

1. change the RL not to transform the call number, however, it will affect everyone.

2. If you have a CCM 4.0 or higher you can proceed as follows:

Create a new list of road that points to the Gulf war, don't do any transformation

Create a new pattern of route for outgoing calls, place it in a new partition,

Create a new CSS that has access to this partition

Assign this CSS to ports CTI associated with this jtapi group used by the application of the src

This way when the CRS needs transfer a call outside it will use difrferent list of route, which does not perform the transformation.

I hope this helps.

Chris

Sx20 does not seek to make the IP to call with H323, after the first attempt failed with SIP

Hi all

We use the Sx20 SIP with CUCM registered devices and also register H323 with VCS - c... There are also a few registered with VCS - E external SX20.

Sx20 > CUCM > VCS - C > VCS-E > Sx20

the defaultcall Protocol is selected Auto (SIP, H323) on devices SX20.

SX20 on cucm cannot call IP when defaultcall is selected as auto, it is able to call IP through VSC - C only when defaultcall is selected H323 on Sx20.

We cannot change the defaultcall as H323 because we cannot make a call since sx20 on CUCM sx20 on vcs - e

someone has an idea to make the two whole scenerio? Perhaps there is a feature to do this on sx20?

your response will be appreciated.

Thank you.

Take a look at these examples of configuration - Dial IP addresses registered to CUCM endpoints with VCS/highway. This would work in your environment.

Kind regards

Acevirgil

do C20 to call outside 8.6 CUCM

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