Presenting Called Party with IVR?

We're building out a prepaid calling card system using IVR instances on Voice Gateways with a modified version of the Debit Card script. The Voice Gateways communicate back to billing software via radius, and so far everything has been going great! Last week we met with the customer and were thrown a bit of a curveball. They are requesting that when the called party picks up the line they are presented with a recorded message and an option to accept or deny the call. Something like: "This is a Prepaid Call from <name>. To accept press 1. To deny press 2, or hang up." To add a bit more complexity, they do not want the calling party to be able to speak with the called party until they have accepted the call. I've been looking around at a few different ways to do this but so far I'm drawing a blank. Any help would be greatly appreciated. Thanks!

Rolando,
Thanks so much for your help! I just reached out to the Cisco Partner Helpline about this exact feature in UCCX. If you don't mind me asking a quick question while I wait to hear back from them. The Voice Gateway running the IVR Debit Card scripts keeps the Billing System updated with call duration info via RADIUS. The Billing System uses this information to determine the call duration and the amount to deduct from the calling account. Once the outbound call has been authorized from the on-site Voice Gateway, would you then have the call forwarded to a UCCX instance that matches on inbound and then creates it's own outbound dial? Forgive my ignorance of UCCX. Would it be something like: match on inbound ANI & park call, place outbound dial to DNIS and run IVR script when called party picks up, after they press 1 connect the 2 calls together? Thanks!

Similar Messages

IP phone display name of the called party with grabage

Hi,
I am running CME with 12.4.14(t) ios. The problem which i am having is that when anyone dial any extension, the name is displayed of the called extension & some garbage charater are shown

Hi,
CSCsj18014 is:
Caller ID string received with extra characters
Symptoms: A caller ID may be received with extra characters.
Conditions: This symptom is observed when caller ID is enabled on both
routers and when the station ID and station name are configured on the FXS
side.
Workaround: There is no workaround.
The reference of FXS ports makes be believe that CSCsj18014 is not the applicable bug. However, in the same results page, we see another bug matching better a problem of garbage w/o FXS ports:
Garbage characters in called name displayed on calling phone
Symptom: Garbage characters in called name displayed on calling phone Conditions: When placing a call from one phone to another phone on CME using 12.4(11)T3, the displayed name on the calling phone gets some unidentified characters at the end of the displayed name after a few seconds have passed. The problem was NOT seen in 12.4(11)T2
Now, the bug toolkit does not give the number for this "related bug" and I was not able to localize it.

Calling Party Presentation

Hello,
I have CM4.1(3)Sr3c connecet to a cisco 1760 h323 gateway, when i make a call from a ipphone to a phone beyind the gateway I dont want to send the calling number, I have tryed in the gateway configuration on the callmanager to put restrict in Calling Party Presentation but with no sucess. Is there any other way to block this, all the other setting's are default.
Thanks

Try setting the Calling Party Number Presentation to Restricted at the Route Pattern or Route List level and see if that helps.
Regards,
Anup

Route pattern with called party transformations

HI All,
i wanto to add route pattern with transformation
i want to add RP with 9.001! predot
and want to convert to 9.01017! with called party transformations.
How we replace ! ? i've tried and it's error with message
Called Party Transform Mask - allowed characters are numeric (0-9),plus (+),asterisk (*),pound (#),X.
I've give screen shot for the configuration
Please anybody help.
Thanks in advance
Regards,
ATommy

You can create the RP with 9001.! and in the called party, discard predot and prefix 901017.
see below screenshot:

Help with Calling Party Transformation Patterns

Hello,I am struggling to figure out how to apply Calling Party Transformation Patterns on CUCM for MGCP gateways.
Basically, we use MGCP gateway everywhere so I need to apply CPTP on CUCM. However, I know how to do this using a gateway (CME) as an example.
Below is what I am trying to figure out using CUCM...Here as you can see, if the calling party number is 000xxxx then the calling Party xformation is set to 353xxxx and then egress to PSTN.
If the calling party number is 111xxxx, then the calling party xformation is set to 454xxxx and then gets sent to the PSTN;
voice translation-rule 1
rule 1 /^000$...$$/ /353\1/
rule 2 /^111$...$$/ /454\1/
voice translation-profile OutBound_CallerID
translate calling 1
dial-peer voice 101 pots
destination-pattern 9T
translation-profile outgoing OutBound_CallerID
port 0/0/0:15
The only way I know how to do this is by creating 2 sets of CSS and PTs. Then, create the same route pattern in both (example 9.!) but set the CPTM on each route pattern as 353xxxx on one and 454xxxx on the other. Then by putting some of the phones with one CSS and some with the other it will work. This is not ideal as I need to be able to do it before matching the Route Patterns and I need all phones to use the same CSS.
Is there a better way?
Thanks

@Anas, thanks for the response, great answer.One question. When creating 'Calling Party Transformation Patterns' should the pattern be the original calling party number i.e 111XXXX.
For example, in the below screenshot will CUCM match the pattern if extension 111XXXX tries to make an outbound call?
So the call flow on CUCM goes:
Ext 111XXXX makes call > Route Pattern>RouteList>ROuteGRoup> Gateway>CPT CSS Applied matching below rule which changes Calling PT to 454XXXX > PSTN
Would that be correct?
Thanks.

Called party number on phone display - updating with results of translation on GW, not wanted

Call Manager 9.x, IOS 15.1, H.323 gateways
Hi, I've got 2 questions regarding the called number display on handsets. Essentially, when a user dials an external number it's obviously shown on their phone handset screen - when that number is manipulated to add prefix for certain PSTN gateways etc. the updated number is shown on the phone display, which the users identify as "not the number I dialled" - can this be changed?
It a cosmetic issue essentially, but one I am being asked about and can't find an answer to:
1) I add a prefix on the gateway to the outbound dialled number (to add a carrier code / network function to all calls) - ie, \^9!\ \1666\ - Process looks like this:
user dials -> 912345
shows as dialled number on handset -> 912345
translation-rule on gateway (in IOS) converts number to -> 166612345
call connected
user handsets now updated to show dialled number as -> 166612345 (but still wanted it to show 912345)
2) Another seperate scenario is that I am doing called party transformation on a route pattern - here the modified number is shown instantly on the callers display. Presumably this in unavoidable? Or, can the original number dialled by the user be displayed on their phone, not the modified one?

Hi,
any calling or called transformation in the route pattern appears in the screen.
you can discard the 9 in the route pattern and add prefix 9 in the route list level.
for the 2nd point there is a service parameter in the call manager to keep the original dialed number
HTH
Anas
don't forget to rate the helpful posts

Play Prompt to Called Party after Placing Outbound call

Hi,
with IPCCX 5.0(IPIVR)- Prem License, is it possible to play a prompt by the IVR to the called party, after the IVR makes call to an external number and the call is connected?
If yes, how?which function is to be used?
thanks

The definition assumes it is an incoming call to IPCC not an "Place call" step, where the roles are reversed.
Yes, my logic was something like:
inboud call arrives
script looks thru DB for scheduled phone numbers to handle the call
IPCC places call to selected number
Menu is played to the called party to either access or reject the call
if call is accepted it is dropped and the original call is redirected to this number.
So, as you can see menu prompt was played to the called party.
Chris

Playing Annoucement to called party before conference call starts

Hi,
I have an interesting requirement here.
My customer wants to play annoucement to the participants of adhoc and meet me conference before bridging them in call. The announcement would be something like this "This call will be recorded, press 1 if you want to join the conference press 0 to hang up". If caller presses '1' then the he should be bridged.
Is it possible to use IVR server for this??
For the adhoc conference this is what I thought:
The conference initiator calls an a number to initiate IVR Script, the script asks him to enter the number to be called(conference participant's number), then the IVR script places call on this number(use "place call function"), once the recipient answers the call, IVR will announce the prompt to him "This call will be recorded, press 1 if you want to join the conference press 0 to hang up".
Questions:
1.Now how to merge this call with the conference initiator call(as conference call) if recipient presses 1??
2.What treatment can be given to the conference call initiator, for the suration when IVR is placing call out? Can we give him some hold music??
2.Is there any better way? Can I use "Call Consult Transfer" function instead of place "call function"??
3.Will it be easier/possible to use Unity instead of IVR server for this?
4.How can the same be done for meetme conference??
Thanks

yes that works to use the new contact which set by 'set contact info' step.
So now I am able to play prompt to called number after the call been connected.
But I found I couldn't collect entered digit by Called person if I use Menu step.
For instance, if I use 'Menu' to collect the digit from Called person, then when enter '1', redirect the original (or this new) call flow to reception.
My sub steps within Menu step (using new contact session) seems no working.
Any thoughts?
General idea is: I can use IVR to place outbound calls, play prompts, then collect digits from called party and make sub call flows decision.
Thanks

Calling Party translation internal and external

Hello all!
I have a problem regarding the external phone number mask and some special requests.
Basically is the feature "external phone number mask" enabled and in use.
Now we have the following request:
Several phones are member of a hunt list, with the extension 555. They want now two groups, one should show internal and external the 555. And the other one should show internal the 555 and external the "normal" extension of the user.
Overview, showing the numbers:
Phone group 1:
external: 555
internal: 555
Phone group 2:
external: line extension
internal: 555
Used is CUCM 8.6 and MGCP-Gateways to the PSTN.
Has anyone an idea to configure that?
Thanks a lot!
Kind regards,
DrMxxxxx

Ah, got it. Well here's how you do it then:
Create a Partition for e.g. PT-CallingParty and put it in CSS-CallingParty.
Create a Calling Party Transformation Mask as following:
4912345XXX with PT-CallingParty
Check Mark "Use Calling Party's External Phone Number Mask"
Under Calling Party Transformation Mask put "4912345555"
Now, for phones:
Go under each phone and under Number Presentation Transformation Uncheck "Use Device pool Calling Party Transformation CSS" and from dropdown just above it Select " CSS-CallingParty" (Apply config, I will suggest BAT for this if you have multiple phones)
Once done that, For User A put External Mask as 4912345555
Whereas for User B put 4912345456
Let me know how that works out for you.
Regards,
Vishal

Calls going out displaying extra digits to called party

I have calls going out of one of our offices that have extra digits appended to the end. Instead of caller id showing the 10 digits of the calling party, it shows 15 repeating the last 5 digits again. If my number was 1234567890, it goes out 123456789067890. Obviously that looks strange and the called party often times doesn't answer. I've been comparing with some of our other sites but I don't see anything obvious. I'm new to voice, I've looked at the dial peer and translation rule on our gateway for that site. Where else should I be looking for this problem. AT&T says they aren't sending the extra digits.
Thanks

Here is the log from a test call after enabling the debugs. Sequence# 095713 includes the extra digits. Thanks for the help.
095693: Sep 30 09:48:53.467 CST: //-1/00E60F465903/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=Dallas Main Number
   ----- ccCallInfo IE subfields -----
   cisco-ani=2147651234
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=917139074321
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=-1
   cisco-rdnplan=-1
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
095694: Sep 30 09:48:53.467 CST: //-1/00E60F465903/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x4A04DDA4, Call Info(
   Calling Number=2147651234,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=917139074321(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=101, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=27508
095695: Sep 30 09:48:53.467 CST: //-1/00E60F465903/CCAPI/ccCheckClipClir:
   In: Calling Number=2147651234(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
095696: Sep 30 09:48:53.467 CST: //-1/00E60F465903/CCAPI/ccCheckClipClir:
   Out: Calling Number=2147651234(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
095697: Sep 30 09:48:53.467 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
095698: Sep 30 09:48:53.467 CST: :cc_get_feature_vsa malloc success
095699: Sep 30 09:48:53.467 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
095700: Sep 30 09:48:53.467 CST: cc_get_feature_vsa count is 3
095701: Sep 30 09:48:53.467 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
095702: Sep 30 09:48:53.467 CST: :FEATURE_VSA attributes are: feature_name:0,feature_time:1309707448,feature_id:27508
095703: Sep 30 09:48:53.467 CST: //27508/00E60F465903/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=2147651234(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=917139074321(TON=Unknown, NPI=Unknown))
095704: Sep 30 09:48:53.471 CST: //27508/00E60F465903/CCAPI/cc_process_call_setup_ind:
   Event=0x4B77BFD0
095705: Sep 30 09:48:53.471 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 917139074321
095706: Sep 30 09:48:53.471 CST: //27508/00E60F465903/CCAPI/ccCallSetContext:
   Context=0x4E138418
095707: Sep 30 09:48:53.471 CST: //27508/00E60F465903/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 27508 with tag 101 to app "_ManagedAppProcess_Default"
095708: Sep 30 09:48:53.471 CST: //27508/00E60F465903/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
095709: Sep 30 09:48:53.475 CST: //27508/00E60F465903/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=91, Params=0x4E1276A8, Progress Indication=NULL(0)
095710: Sep 30 09:48:53.475 CST: //27508/00E60F465903/CCAPI/ccCheckClipClir:
   In: Calling Number=214765123451234(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
095711: Sep 30 09:48:53.475 CST: //27508/00E60F465903/CCAPI/ccCheckClipClir:
   Out: Calling Number=214765123451234(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
095712: Sep 30 09:48:53.475 CST: //27508/00E60F465903/CCAPI/ccCallSetupRequest:
   Destination Pattern=91[2-9].........$, Called Number=917139074321, Digit Strip=TRUE
095713: Sep 30 09:48:53.475 CST: //27508/00E60F465903/CCAPI/ccCallSetupRequest:
   Calling Number=214765123451234(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=917139074321(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=Dallas Main Number
   Account Number=Dallas Main Number, Final Destination Flag=TRUE,
   Guid=00E60F46-E5D0-A142-5903-1B030A0A023F, Outgoing Dial-peer=91
095714: Sep 30 09:48:53.475 CST: //27508/00E60F465903/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=Dallas Main Number
   ----- ccCallInfo IE subfields -----
   cisco-ani=214765123451234
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=917139074321
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=-1
   cisco-rdnplan=-1
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
095715: Sep 30 09:48:53.475 CST: //27508/00E60F465903/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x4B7C2F08, Interface Type=6, Destination=, Mode=0x0,
   Call Params(Calling Number=214765123451234,(Calling Name=Dallas Main Number)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=917139074321(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=91, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
095716: Sep 30 09:48:53.475 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
095717: Sep 30 09:48:53.479 CST: :cc_get_feature_vsa malloc success
095718: Sep 30 09:48:53.479 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
095719: Sep 30 09:48:53.479 CST: cc_get_feature_vsa count is 4
095720: Sep 30 09:48:53.479 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
095721: Sep 30 09:48:53.479 CST: :FEATURE_VSA attributes are: feature_name:0,feature_time:1309709240,feature_id:27509
095722: Sep 30 09:48:53.479 CST: //27509/00E60F465903/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=6, FlowMode=1
095723: Sep 30 09:48:53.479 CST: //27509/00E60F465903/CCAPI/ccCallSetContext:
   Context=0x4E127658
095724: Sep 30 09:48:53.479 CST: //27508/00E60F465903/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=91
095725: Sep 30 09:48:53.483 CST: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13
095726: Sep 30 09:48:53.483 CST: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x1 0x1, Calling num 214765123451234
095727: Sep 30 09:48:53.487 CST: ISDN Se0/0/0:23 Q931: Sending SETUP callref = 0x13D3 callID = 0x9354 switch = primary-ni interface = User
095728: Sep 30 09:48:53.487 CST: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x13D3
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98397
                Exclusive, Channel 23
        Display i = 'Dallas Main Number'
        Calling Party Number i = 0x1181, '214765123451234'
                Plan:ISDN, Type:International
        Called Party Number i = 0x80, '17139074321'
                Plan:Unknown, Type:Unknown
095729: Sep 30 09:48:53.551 CST: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x93D3
        Channel ID i = 0xA98397
                Exclusive, Channel 23
095730: Sep 30 09:48:53.579 CST: //27509/00E60F465903/CCAPI/cc_api_call_proceeding:
   Interface=0x4B7C2F08, Progress Indication=NULL(0)
095731: Sep 30 09:48:55.295 CST: ISDN Se0/0/0:23 Q931: RX <- ALERTING pd = 8 callref = 0x93D3
        Progress Ind i = 0x8488 - In-band info or appropriate now available
095732: Sep 30 09:48:55.295 CST: //27509/00E60F465903/CCAPI/cc_api_call_alert:
   Interface=0x4B7C2F08, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1)
095733: Sep 30 09:48:55.295 CST: //27509/00E60F465903/CCAPI/cc_api_call_alert:
   Call Entry(Retry Count=0, Responsed=TRUE)
095734: Sep 30 09:48:55.299 CST: //27508/00E60F465903/CCAPI/ccCallAlert:
   Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1)
095735: Sep 30 09:48:55.299 CST: //27508/00E60F465903/CCAPI/ccCallAlert:
   Call Entry(Responsed=TRUE, Alert Sent=TRUE)
095736: Sep 30 09:48:55.299 CST: //27509/00E60F465903/CCAPI/cc_api_get_called_ccm_detected:
   CallInfo(ccm detected=0)
095737: Sep 30 09:48:55.299 CST: //27508/00E60F465903/CCAPI/ccConferenceCreate:
   (confID=0x4E1EE068, callID1=0x6B74, gcid=0-0-0-0, tag=0x0)
095738: Sep 30 09:48:55.299 CST: //27509/00E60F465903/CCAPI/ccConferenceCreate:
   (confID=0x4E1EE068, callID2=0x6B75, gcid=0-0-0-0, tag=0x0)
095739: Sep 30 09:48:55.299 CST: //27508/00E60F465903/CCAPI/ccConferenceCreate:
   Conference Id=0x4E1EE068, Call Id1=27508, Call Id2=27509, Tag=0x0
095740: Sep 30 09:48:55.299 CST: //27508/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
095741: Sep 30 09:48:55.299 CST: cc_api_get_xcode_stream : 4702
095742: Sep 30 09:48:55.299 CST: //27508/00E60F465903/CCAPI/cc_api_bridge_done:
   Conference Id=0x2A56, Source Interface=0x4A04DDA4, Source Call Id=27508,
   Destination Call Id=27509, Disposition=0x0, Tag=0x0
095743: Sep 30 09:48:55.299 CST: //27509/00E60F465903/CCAPI/cc_api_bridge_done:
   Conference Id=0x2A56, Source Interface=0x4B7C2F08, Source Call Id=27509,
   Destination Call Id=27508, Disposition=0x0, Tag=0xFFFFFFFF
095744: Sep 30 09:48:55.299 CST: //27508/00E60F465903/CCAPI/cc_generic_bridge_done:
   Conference Id=0x2A56, Source Interface=0x4B7C2F08, Source Call Id=27509,
   Destination Call Id=27508, Disposition=0x0, Tag=0xFFFFFFFF
095745: Sep 30 09:48:55.299 CST: //27508/00E60F465903/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x2A56, Destination Call Id=27509)
095746: Sep 30 09:48:55.299 CST: //27509/00E60F465903/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x2A56, Destination Call Id=27508)
095747: Sep 30 09:48:55.299 CST: //27509/00E60F465903/CCAPI/cc_api_caps_ind:
   Destination Interface=0x4A04DDA4, Destination Call Id=27508, Source Call Id=27509,
   Caps(Codec=0x1, Fax Rate=0x1, Vad=0x1,
   Modem=0x2, Codec Bytes=20, Signal Type=3)
095748: Sep 30 09:48:55.303 CST: //27509/00E60F465903/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
095749: Sep 30 09:48:55.303 CST: //27508/00E60F465903/CCAPI/ccCallNotify:
   Data Bitmask=0x7, Call Id=27508
095750: Sep 30 09:48:55.303 CST: //27509/00E60F465903/CCAPI/cc_api_get_called_ccm_detected:
   CallInfo(ccm detected=0)
095751: Sep 30 09:48:55.303 CST: //27508/00E60F465903/CCAPI/cc_api_get_delay_xport:
   CallInfo(delay xport=FALSE)
095752: Sep 30 09:48:55.311 CST: //27508/00E60F465903/CCAPI/cc_process_notify_bridge_done:
   Conference Id=0x2A56, Call Id1=27508, Call Id2=27509
095753: Sep 30 09:48:55.811 CST: //27508/00E60F465903/CCAPI/cc_api_caps_ind:
   Destination Interface=0x4B7C2F08, Destination Call Id=27509, Source Call Id=27508,
   Caps(Codec=0x1, Fax Rate=0x80, Vad=0x1,
   Modem=0x0, Codec Bytes=160, Signal Type=2)
095754: Sep 30 09:48:55.811 CST: //27508/00E60F465903/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
095755: Sep 30 09:48:55.811 CST: //27508/00E60F465903/CCAPI/cc_api_caps_ack:
   Destination Interface=0x4B7C2F08, Destination Call Id=27509, Source Call Id=27508,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_14400(0x80), Vad=OFF(0x1),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=8122)
095756: Sep 30 09:48:55.815 CST: //27509/00E60F465903/CCAPI/cc_api_caps_ack:
   Destination Interface=0x4A04DDA4, Destination Call Id=27508, Source Call Id=27509,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_14400(0x80), Vad=OFF(0x1),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=8122)
095757: Sep 30 09:48:55.819 CST: //27509/00E60F465903/CCAPI/cc_api_voice_mode_event:
   Call Id=27509
095758: Sep 30 09:48:55.819 CST: //27509/00E60F465903/CCAPI/cc_api_voice_mode_event:
   Call Entry(Context=0x4E127658)
095759: Sep 30 09:49:03.691 CST: //27507/E25C9BF99854/CCAPI/cc_api_call_disconnected:
   Cause Value=16, Interface=0x4A04DDA4, Call Id=27507
095760: Sep 30 09:49:03.695 CST: //27507/E25C9BF99854/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
095761: Sep 30 09:49:03.695 CST: //27506/E25C9BF99854/CCAPI/ccConferenceDestroy:
   Conference Id=0x2A55, Tag=0x0
095762: Sep 30 09:49:03.695 CST: //27506/E25C9BF99854/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x2A55, Source Interface=0x4B7C2F08, Source Call Id=27506,
   Destination Call Id=27507, Disposition=0x0, Tag=0x0
095763: Sep 30 09:49:03.695 CST: //27507/E25C9BF99854/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x2A55, Source Interface=0x4A04DDA4, Source Call Id=27507,
   Destination Call Id=27506, Disposition=0x0, Tag=0x0
095764: Sep 30 09:49:03.695 CST: //27506/E25C9BF99854/CCAPI/cc_generic_bridge_done:
   Conference Id=0x2A55, Source Interface=0x4A04DDA4, Source Call Id=27507,
   Destination Call Id=27506, Disposition=0x0, Tag=0x0
095765: Sep 30 09:49:03.695 CST: //27506/E25C9BF99854/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
095766: Sep 30 09:49:03.695 CST: //27506/E25C9BF99854/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
095767: Sep 30 09:49:03.695 CST: //27506/E25C9BF99854/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
095768: Sep 30 09:49:03.695 CST: //27507/E25C9BF99854/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)
095769: Sep 30 09:49:03.695 CST: //27507/E25C9BF99854/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
095770: Sep 30 09:49:03.699 CST: //27507/E25C9BF99854/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
095771: Sep 30 09:49:03.719 CST: //27507/E25C9BF99854/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
095772: Sep 30 09:49:03.723 CST: //27507/E25C9BF99854/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x4A04DDA4, Tag=0x0, Call Id=27507,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
095773: Sep 30 09:49:03.723 CST: //27507/E25C9BF99854/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
095774: Sep 30 09:49:03.723 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
095775: Sep 30 09:49:03.723 CST: :cc_free_feature_vsa freeing 4E108850
095776: Sep 30 09:49:03.723 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
095777: Sep 30 09:49:03.723 CST: vsacount in free is 3
095778: Sep 30 09:49:03.727 CST: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 2147662349 , call lasted 105 seconds
095779: Sep 30 09:49:03.731 CST: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x833F
        Cause i = 0x8090 - Normal call clearing
095780: Sep 30 09:49:03.759 CST: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x033F
095781: Sep 30 09:49:03.759 CST: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x833F
095782: Sep 30 09:49:03.767 CST: //27506/E25C9BF99854/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x4B7C2F08, Tag=0x0, Call Id=27506,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
095783: Sep 30 09:49:03.767 CST: //27506/E25C9BF99854/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
095784: Sep 30 09:49:03.767 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
095785: Sep 30 09:49:03.767 CST: :cc_free_feature_vsa freeing 4E109570
095786: Sep 30 09:49:03.767 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
095787: Sep 30 09:49:03.767 CST: vsacount in free is 2
095788: Sep 30 09:49:13.991 CST: ISDN Se0/0/0:23 Q931: RX <- CONNECT pd = 8 callref = 0x93D3
        Progress Ind i = 0x8482 - Destination address is non-ISDN
095789: Sep 30 09:49:13.995 CST: %ISDN-6-CONNECT: Interface Serial0/0/0:22 is now connected to 17139074321 N/A
095790: Sep 30 09:49:13.995 CST: ISDN Se0/0/0:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x13D3
095791: Sep 30 09:49:13.995 CST: //27509/00E60F465903/CCAPI/cc_api_call_connected:
   Interface=0x4B7C2F08, Data Bitmask=0x1, Progress Indication=DESTINATION IS NON ISDN(2),
   Connection Handle=0
095792: Sep 30 09:49:13.999 CST: //27509/00E60F465903/CCAPI/cc_api_call_connected:
   Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
095793: Sep 30 09:49:13.999 CST: //27508/00E60F465903/CCAPI/ccCallConnect:
   Progress Indication=DESTINATION IS NON ISDN(2), Data Bitmask=0x1
095794: Sep 30 09:49:13.999 CST: //27509/00E60F465903/CCAPI/cc_api_get_called_ccm_detected:
   CallInfo(ccm detected=0)
095795: Sep 30 09:49:13.999 CST: //27508/00E60F465903/CCAPI/ccCallConnect:
   Call Entry(Connected=TRUE, Responsed=TRUE)
095796: Sep 30 09:49:13.999 CST: //27508/00E60F465903/CCAPI/ccCallNotify:
   Data Bitmask=0x7, Call Id=27508
095797: Sep 30 09:49:13.999 CST: //27509/00E60F465903/CCAPI/cc_api_get_called_ccm_detected:
   CallInfo(ccm detected=0)
095798: Sep 30 09:49:33.319 CST: //27508/00E60F465903/CCAPI/cc_api_call_disconnected:
   Cause Value=16, Interface=0x4A04DDA4, Call Id=27508
095799: Sep 30 09:49:33.323 CST: //27508/00E60F465903/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
095800: Sep 30 09:49:33.323 CST: //27508/00E60F465903/CCAPI/ccConferenceDestroy:
   Conference Id=0x2A56, Tag=0x0
095801: Sep 30 09:49:33.323 CST: //27508/00E60F465903/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x2A56, Source Interface=0x4A04DDA4, Source Call Id=27508,
   Destination Call Id=27509, Disposition=0x0, Tag=0x0
095802: Sep 30 09:49:33.323 CST: //27509/00E60F465903/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x2A56, Source Interface=0x4B7C2F08, Source Call Id=27509,
   Destination Call Id=27508, Disposition=0x0, Tag=0x0
095803: Sep 30 09:49:33.323 CST: //27508/00E60F465903/CCAPI/cc_generic_bridge_done:
   Conference Id=0x2A56, Source Interface=0x4B7C2F08, Source Call Id=27509,
   Destination Call Id=27508, Disposition=0x0, Tag=0x0
095804: Sep 30 09:49:33.327 CST: //27508/00E60F465903/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)
095805: Sep 30 09:49:33.327 CST: //27508/00E60F465903/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
095806: Sep 30 09:49:33.327 CST: //27508/00E60F465903/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
095807: Sep 30 09:49:33.327 CST: //27509/00E60F465903/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
095808: Sep 30 09:49:33.327 CST: //27509/00E60F465903/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
095809: Sep 30 09:49:33.327 CST: //27509/00E60F465903/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
095810: Sep 30 09:49:33.351 CST: //27508/00E60F465903/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
095811: Sep 30 09:49:33.355 CST: //27508/00E60F465903/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x4A04DDA4, Tag=0x0, Call Id=27508,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
095812: Sep 30 09:49:33.355 CST: //27508/00E60F465903/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
095813: Sep 30 09:49:33.355 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
095814: Sep 30 09:49:33.355 CST: :cc_free_feature_vsa freeing 4E108CB0
095815: Sep 30 09:49:33.355 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
095816: Sep 30 09:49:33.355 CST: vsacount in free is 1
095817: Sep 30 09:49:33.363 CST: %ISDN-6-DISCONNECT: Interface Serial0/0/0:22 disconnected from 17139074321 , call lasted 19 seconds
095818: Sep 30 09:49:33.363 CST: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x13D3
        Cause i = 0x8090 - Normal call clearing
095819: Sep 30 09:49:33.399 CST: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x93D3
095820: Sep 30 09:49:33.399 CST: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x13D3
095821: Sep 30 09:49:33.403 CST: //27509/00E60F465903/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x4B7C2F08, Tag=0x0, Call Id=27509,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
095822: Sep 30 09:49:33.403 CST: //27509/00E60F465903/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
095823: Sep 30 09:49:33.403 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
095824: Sep 30 09:49:33.403 CST: :cc_free_feature_vsa freeing 4E1093B0
095825: Sep 30 09:49:33.403 CST: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
095826: Sep 30 09:49:33.403 CST: vsacount in free is 0

Calling issue with Cisco 7937 conference station

Hi Friends,
I am facing issue wiht Cisco 7937 conference station, our customer have various branch offices accross the world. All branches are connected over MPLS through service provider( SIP service provider) . there is a centralized CUCM and remote office have SIP Voice gateways .
When making calls from once remote site to another using Cisco 6921 phones calls working fine
When making calls from once remote site to another using Cisco 7937 conference station to make call any phone at remote office, calls are getting disconneted, remote phone rings when calls, but its gets fast busy tone when other party picks up the phone and not able to talk.
I suspect the issue with Codec but we have configured transcoders in VG and registered with CUCM
Please help me if any one experience such issue earlier.
Regards
Siva

hi Basant,
1. Actually tow phones A and B are registerd with centralized CUCM, A and B are located in two different locations, RTP traffic between And B pass through service provider.
Call Flow --> Phone A ---->CUCMRouterpattern--> SIP trunk ----> Voice gateway--->Service provider cloud---> Respective Voice Gateway---> CUCM -- Phone B
Show Run
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.27 15:14:52 =~=~=~=~=~=~=~=~=~=~=~=
sh run
Building configuration...
Current configuration : 12139 bytes
! Last configuration change at 06:35:59 UTC Tue Feb 25 2014
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname eucamvgw01
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.151-4.M5.bin
boot-end-marker
card type e1 0 0
logging buffered 51200 warnings
no logging console
no aaa new-model
no network-clock-participate wic 0
no ipv6 cef
ip source-route
ip traffic-export profile cuecapture mode capture
bidirectional
ip cef
ip multicast-routing
ip domain name drreddys.eu
ip name-server 10.197.20.1
ip name-server 10.197.20.2
multilink bundle-name authenticated
stcapp ccm-group 2
stcapp
stcapp feature access-code
stcapp feature speed-dial
stcapp supplementary-services
port 0/1/0
fallback-dn 5428025
port 0/1/1
fallback-dn 5428008
port 0/1/2
fallback-dn 5421462
port 0/1/3
fallback-dn 5421463
isdn switch-type primary-net5
crypto pki token default removal timeout 0
voice-card 0
dsp services dspfarm
voice call send-alert
voice call disc-pi-off
voice call convert-discpi-to-prog
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 10.198.0.0 255.255.255.0
ipv4 152.63.1.0 255.255.255.0
address-hiding
allow-connections sip to sip
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
fax-relay ans-disable
sip
rel1xx supported "track"
privacy pstn
no update-callerid
early-offer forced
call-route p-called-party-id
voice class uri 100 sip
host 41.206.187.71
voice class codec 10
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 ilbc
codec preference 4 g729r8
codec preference 5 g729br8
voice class codec 20
codec preference 1 g729br8
codec preference 2 g729r8
voice moh-group 1
moh flash:moh/Panjo.alaw.wav
description MOH G711 alaw
multicast moh 239.1.1.2 port 16384 route 10.198.2.9
voice translation-rule 1
rule 1 /^012237280$..$/ /54280\1/
rule 2 /^012236514$..$/ /54214\1/
rule 3 /^01223651081/ /5428010/
rule 4 /^01223506701/ /5428010/
voice translation-rule 2
rule 1 /^00$.+$/ /+\1/
rule 2 /^0$.+$/ /+44\1/
rule 3 /^$[0-9].+$/ /+\1/
voice translation-rule 3
rule 1 /^9$.+$/ /\1/
rule 2 /^\+44$.+$/ /0\1/
rule 3 /^\+$.+$/ /00\1/
voice translation-rule 4
rule 1 /^54280$..$/ /12237280\1/
rule 2 /^54214$..$/ /12236514\1/
rule 3 /^\+44$.+$/ /\1/
rule 4 /^.54280$..$/ /12237280\1/
rule 5 /^.54214$..$/ /12236514\1/
voice translation-rule 9
rule 1 /^$....$/ /542\1/
voice translation-rule 10
voice translation-rule 11
rule 1 /^\+44122372$....$/ /542\1/
rule 2 /^\+44122365$....$/ /542\1/
voice translation-rule 12
voice translation-rule 13
rule 1 /^$[18]...$/ /542\1/
voice translation-rule 14
voice translation-profile MPLS-incoming
translate calling 10
translate called 9
voice translation-profile MPLS-outgoing
translate calling 11
translate called 12
voice translation-profile PSTN-incoming
translate calling 2
translate called 1
voice translation-profile PSTN-outgoing
translate calling 4
translate called 3
voice translation-profile SRST-incoming
translate calling 14
translate called 13
license udi pid CISCO2921/K9 sn FGL145110RE
hw-module ism 0
hw-module pvdm 0/0
username administrator privilege 15 secret 5 $1$syu5$DsxdOgfS7Wltx78o4PV.60
redundancy
controller E1 0/0/0
ip tcp path-mtu-discovery
ip scp server enable
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description internal LAN
ip address 10.198.2.9 255.255.255.0
duplex auto
speed auto
interface ISM0/0
ip unnumbered GigabitEthernet0/0
service-module ip address 10.198.2.8 255.255.255.0
!Application: CUE Running on ISM
service-module ip default-gateway 10.198.2.9
interface GigabitEthernet0/1
description to TATA NGN
ip address 115.114.225.122 255.255.255.252
duplex auto
speed auto
interface GigabitEthernet0/2
description SIP Trunks external
ip address 79.121.254.83 255.255.255.248
ip access-group SIP-InBound in
ip traffic-export apply cuecapture size 8000000
duplex auto
speed auto
interface ISM0/1
description Internal switch interface connected to Internal Service Module
no ip address
shutdown
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 10.198.2.1
ip route 10.198.2.8 255.255.255.255 ISM0/0
ip route 41.206.187.0 255.255.255.0 115.114.225.121
ip route 77.37.25.46 255.255.255.255 79.121.254.81
ip route 83.245.6.81 255.255.255.255 79.121.254.81
ip route 83.245.6.82 255.255.255.255 79.121.254.81
ip route 95.223.1.107 255.255.255.255 79.121.254.81
ip route 192.54.47.0 255.255.255.0 79.121.254.81
ip access-list extended SIP-InBound
permit ip host 77.37.25.46 any
permit ip host 83.245.6.81 any
permit ip host 83.245.6.82 any
permit ip 192.54.47.0 0.0.0.255 any
permit icmp any any
permit ip host 95.223.1.107 any
deny ip any any log
control-plane
voice-port 0/1/0
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/1
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/2
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/3
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
no ccm-manager fax protocol cisco
ccm-manager music-on-hold bind GigabitEthernet0/0
ccm-manager config server 152.63.1.19 152.63.1.100 172.27.210.5
ccm-manager sccp local GigabitEthernet0/0
ccm-manager sccp
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 10.198.2.9 identifier 3 priority 3 version 7.0
sccp ccm 152.63.1.19 identifier 4 version 7.0
sccp ccm 152.63.1.100 identifier 5 version 7.0
sccp ccm 172.27.210.5 identifier 6 version 7.0
sccp
sccp ccm group 2
bind interface GigabitEthernet0/0
associate ccm 4 priority 1
associate ccm 5 priority 2
associate ccm 6 priority 3
associate ccm 3 priority 4
associate profile 1002 register CFB_UK_CAM_02
associate profile 1001 register XCODE_UK_CAM_02
associate profile 1000 register MTP_UK_CAM_02
dspfarm profile 1001 transcode
codec ilbc
codec g722-64
codec g729br8
codec g729r8
codec gsmamr-nb
codec pass-through
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 18
associate application SCCP
dspfarm profile 1002 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 1000 mtp
codec g711alaw
maximum sessions software 200
associate application SCCP
dial-peer cor custom
name SRSTMode
dial-peer cor list SRST
member SRSTMode
dial-peer voice 100 voip
description *** Inbound CUCM ***
translation-profile incoming PSTN-incoming
incoming called-number .
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 500 voip
description *** Inbound TATA MPLS ***
translation-profile incoming MPLS-incoming
session protocol sipv2
session target sip-server
incoming called-number ....
incoming uri from 100
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 510 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 54[013-9]....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 520 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 5[0-35-9].....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 200 voip
description *** Inbound M12 *** 01223651081, 01223651440 - 01223651489
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 0122365....
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 201 voip
description *** Inbound M12 *** 012237280XX
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 012237280..
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 202 voip
description *** Inbound M12 *** 01223506701
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 01223506701
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 210 voip
description *** Outbound M12 ***
translation-profile outgoing PSTN-outgoing
destination-pattern +...T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 211 voip
description *** Outbound ISDN for SRST and emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 9.T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 212 voip
description *** Outbound ISDN for emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 11[02]
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 2000 voip
description *** Outbound to CUCM Primary ***
preference 1
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.19
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2001 voip
description *** Outbound to CUCM Secondary ***
preference 2
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.100
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2002 voip
description *** Outbound to CUCM Teritiary ***
preference 3
destination-pattern 542....
session protocol sipv2
session target ipv4:172.27.210.5
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 999010 pots
service stcapp
port 0/1/0
dial-peer voice 999011 pots
service stcapp
port 0/1/1
dial-peer voice 999012 pots
service stcapp
port 0/1/2
dial-peer voice 999013 pots
service stcapp
port 0/1/3
sip-ua
no remote-party-id
gatekeeper
shutdown
call-manager-fallback
secondary-dialtone 9
max-conferences 4 gain -6
transfer-system full-consult
ip source-address 10.198.2.9 port 2000
max-ephones 110
max-dn 400 dual-line no-reg
translation-profile incoming SRST-incoming
moh flash:/moh/Panjo.ulaw.wav
multicast moh 239.1.1.1 port 16384 route 10.198.2.9
time-zone 22
time-format 24
date-format dd-mm-yy
line con 0
login local
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 131
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
line vty 5 15
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
scheduler allocate 20000 1000
ntp server 10.1.30.1
end
eucamvgw01#
Sh SCCP
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 17:57:44 =~=~=~=~=~=~=~=~=~=~=~=
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: 10.198.2.9
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.198.2.9, Port Number: 2000
Priority: 3, Version: 7.0, Identifier: 3
Call Manager: 152.63.1.19, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 4
Trustpoint: N/A
Call Manager: 152.63.1.100, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 5
Trustpoint: N/A
Call Manager: 172.27.210.5, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 6
Trustpoint: N/A
MTP Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1000
Reported Max Streams: 400, Reported Max OOS Streams: 0
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1001
Reported Max Streams: 36, Reported Max OOS Streams: 0
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: g722r64, Maximum Packetization Period: 30
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
Supported Codec: pass-thru, Maximum Packetization Period: N/A
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1002
Reported Max Streams: 16, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070080
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070081
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070082
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070083
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
eucamvgw01#

How to append calling and called number with translation rules?

Hello,
I have one question about digit manipulations.
How to append calling number and called number with IOS commands?
For example, when 123 dials 45678, translations have to be performed and the new called number to be 12345678.
Thank you,
I will vote this conversation.

It is not possible with translation rules.
However, you can do that with a TCL/IVR script.

Route Pattern CSV File Removes "0" from Called Party Prefix Digits Field

I want to upload over 350 route patterns using BAT Tool in CUCM 9.1. All patterns must have their Called Party Prefix Digits (Outgoing Calls) Field containing 10 numbers with the number "0" at the begining. The Problem is the CSV file removes the leading "0" form the digits. I tried to make the cell type in Excel as "Text" and it worked and the "0" is kept normally, but when I save and close the file then open it again, the cell is defaulted to "General" type and the "0" is disappeared again! Changing the CSV file format to any other one and uploading it to CUCM system generates an error stating that the file format is not supported.
Attached is a sample entry of the CSV file. I want to preserve the whole number "0541234567" in "PREFIX_DIGITS_CALLED_PARTY" Field.
Anyone can help me how to upload this big number of route patterns while preserving the number "0" at the begining?

Make sure you change the csv file when using Excel to "Text" on the cell where the string starts with 0, otherwise Excel assumes this is a number and strips it.
HTH, please rate all useful posts!
Chris

Use External phone Number mask * Calling party Transform Mask @ Route patt

Having an issue with CLID at RP level. Lets say I have two phones. One phone is configured with an External phone number mask of 1112223333 and the other one does not have an external number mask set. When a call is placed to the PSTN, Phone one needs to display its external phone number mask. Phone two since nothing has been added under external phone mask needs to be 1112224444.
I configured the RP to "Use External Phone Number mask" and set the Calling Party Transform mask to 1112224444. I figured that if an external phone number mask was specified under the DN that it will route it to the PSTN and if nothing specified under the DN it will default to 1112224444 according to the setting in the RP.
When I configured this, no matter what I do the calling party transform mask takes priority and the external phone number mask is never used. Is there a way around this? I tried it at the RL level and it does the same thing.
Does anyone have any other ideas other than adding an external phone number mask for every phone.

Thats what I thought. Why can't they add that as a feature in the next release of CM? It seems like it could be simple to do. Check for External Phone number mask, if nothing is specified, then check for calling party transform mask. If true set Calling Party transform Mask as CLID and route to RL or something like that..
thanks

Called Party Answer (CPA) / Line Reversal

Hi - is there any reason why the Called Party Answer (CPA) service can only be applied to Business and not Residential lines. SPM/MPF used to be available on Residential sub's lines - seems weird that its replacement (CPA) can only be activated on Business Lines ?

monolog99 wrote:
Hi - is there any reason why the Called Party Answer (CPA) service can only be applied to Business and not Residential lines. SPM/MPF used to be available on Residential sub's lines - seems weird that its replacement (CPA) can only be activated on Business Lines ?
This is a Customer to Customer forum, so messages do not go to BT.
I may be able to give you some idea why.
CPA was a hangover from the days of Strowger mechanical exchanges and was implemented on subsequent exchange designs to maintain compatibility with some legacy products that relied on a physical line reversal to detect when a call was answered.
Modern equipment, like answering machines, can simply detect the network tones to determine when a call has been answered, so this facility was not implemented on residential lines.
Some business lines still have legacy equipment connected to them, like older switchboards(PABX), that require line reversal. Also they may have payphones which need this facility.
So this facility has been retained for business lines until such time as all the legacy equipment has been replaced. From what I remember, this has to happen before full implementation of the 21CN network, whenever that may be.
I hope that is of some help. Was there any reason why you were asking?
There are some useful help pages here, for BT Broadband customers only, on my personal website.
BT Broadband customers - help with broadband, WiFi, networking, e-mail and phones.

Presenting Called Party with IVR?

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