Translating number
Hi all experts.
In data networks, private IPs are translated to public IP when accessing the internet. In Voip, do we have the same concept ?
If i am using SIP trunk from a local ITSP, when an extension (1001) dials to his home number (521 9632), what number will ITSP get as calling ID ? 1001 ? or what ?
I am in assumption that we need to do some sort of translation here so that call can be reached back to us, just like in data networks. Pls correct me
Actually, it depends on the PSTN connectivity provider. In some cases, no matter what you pass out - the calling number is transformed to a DID associated with the trunk group (or, in some cases - a BTN). In others, you can pass whatever you want and the carrier will pass it along. So, this is how it's done in real environments. You generally set the External Calling Party Mask at the line and then that's what is presented to the PSTN as calling party ID.
I understand why you are using the term "translate" but it doesn't really apply in terms of the functionality in CUCM. When you translate, you are manipulating the called number. What you want to do is manipulate the calling number. At any rate, the external calling party mask is typically what is used by the customers I've encountered. In many cases, the external calling party is set to a mask of sorts. For example:
Your DID is 7035551212.
You set the Calling Party External Mask at the line to 703555XXXX. It's easily BAT'd and generic and would pass the full DID out on outbound calls. In other cases, you may have a private number (internal).
The private number might be 7031091212.
You set the Calling Party External Mask to the DID for a company such as 7035551234. Whenever users call out from a private line, the corporate DID is provided as the callback in the calling party info.
Hailey
Please rate helpful posts!
Message was edited: Updated information on use of "translate" in relevance to topic.
Similar Messages
-
Needed program unit to translate number as currency to string
HI All,
Can you please tell me if there is existing program unit that i can use to translate number as currency to string
Ex:
i have numbeer like 110.5 and currency code EGP
i need the program unit to return
Only One Hundred Ten Egyptian Pound (s) and Fifty Piaster (s)
for ococourse if the EGP is USD it'll return dollar instead of egyptian pound
Thanks youYeah i already did so and i got the function and used it my problem just in the currency translation for decimals
that means when i need to translate 100.5 EGP
one hundred Egyptian Pounds and fifty piasters
i can now translate to one hundred Egyptian Pounds and fifty but what about the piasters
i think i had to create a table that contains a currency code and the corresponding currency translation piasters , cents , ...etc
so i think i'll have to search on google about the currency transaltion for the decimal point currency i hoped there is a tabke in the e-business suite contains it as it exists for the currency code -
CUCM doesn't show the translated number
Hey guys,
I had this working on my Call Manager and not sure what has happened, was hoping someone could help.
I have a number of translation patterns configured so that if a user dials another user by their external DID number, the CM routes the call internally.
This still works - however after dialling the full external number, once the dialling is finished we used to see the translated (shortened internal) number of the phone but now we are only seeing exactly what was dialled.
Example:
Internal user has extension 2000, this extension has a DID of number 073072000
If I am on extension 2200 and I dial 073072000 then on my phones display it would change to 2000 as soon as I finished dialling digits.
Now for some reason it is no longer changing the number to 2000 after I finish dialling - it just displays the actual dialled digits of 073072000.
I know that the translation pattern is working, due to call access restrictions the call would normally be blocked. It just is not showing the translated pattern on the phone display.
The pattern is configured as:
007307.2XXXX
The Called Party Transformation is set to discard PreDot.
Thoughts?Thanks this was the setting I had changed!
I had turned it to True as this meant incoming calls from UCCX to agents then shows the callers number and not the CTI port that sends the call over to CUCM - which is annoying.
Sucks you can only have it one way or the other system wide. -
Re-translate number for calls forwarded from Lync
On a user's desk phone, a call shows up as "315554443333" with 3 being the dial-out code. This way, a user can easily return a missed/received call. In Lync, the call shows up as "+5554443333".
This is fine for Lync users, but if a Lync user were to transfer a call to a desk phone user, the call appears as "+5554443333". The desk phone user can no longer return this call, they would have to dial the digits manually.
How would I go about re-translating this number? Thanks!
It worked for me.That is correct, PSTN to Call Manager with SIP trunk off to Lync. I am thinking the manipulation needs to occur in Lync, not in Call Manager. We are using several number translations already, here is an example:
OutsideCaller at 555-444-3333 calls InsideUser1 at 222-333-4444. The user has a desk phone and is configured for Enterprise Voice in Lync, so both their DeskPhone1 extension (4444) and their LyncPhone extension (3444) ring. 315554443333 appears
as the Calling Number on DeskPhone1, but +5554443333 appears on LyncPhone.
1. If InsideUser1 forwards a call received on DeskPhone1 from OutsideCaller to InsideUser2 who answers it on their DeskPhone2, it appears on DeskPhone2 as 315554443333 (GOOD).
2. If InsideUser1 forwards a call received on DeskPhone1 from OutsideCaller to InsideUser2 who answers it on their LyncPhone2, it appears on LyncPhone2 as 4444 (BAD, should show original calling party 315554443333).
3. If InsideUser1 forwards a call received on LyncPhone1 from OutsideCaller to InsideUser2 who answers it on their DeskPhone2, it shows on DeskPhone2 as +555444333 (BAD, should display number as 315554443333).
4. If InsideUser1 forwards a call received on LyncPhone1 from OutsideCaller to InsideUser2 who answers it on their LyncPhone2, it shows on LyncPhone2 as +555444333 (GOOD).
I hope this helps to clarify.
It worked for me. -
Trouble with international translation
Hello all,
I'm having a problem getting international calls to complete. I have the below in my Cisco 5350 configs with some lines removed to save space. Debug is below that. It appears my translation rule is ok to some extent as the test indicates. What am I overlooking? Any help will be greatly appreciated.
Sprint5350#test voice translation-rule 2 011862196990
Matched with rule 1
Original number: 011862196990 Translated number: 0111862196990
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
voice translation-rule 1
rule 1 /\+1\(.*\)/ /\1/
voice translation-rule 2
rule 1 /^01/ /011/
voice translation-profile Incoming_Calling_Party_Num
translate calling 1
voice translation-profile outgoing_international
translate calling 2
translate called 2
controller T3 3/0
framing m23
clock source line
cablelength 133
t1 1-7 controller
description Sprint DS3 - 75955030
controller T1 3/0:1
framing esf
pri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 1
controller T1 3/0:2
framing esf
pri-group timeslots 1-24 nfas_d none nfas_int 2 nfas_group 1
controller T1 3/0:3
framing esf
pri-group timeslots 1-24 nfas_d backup nfas_int 1 nfas_group 1
controller T1 3/0:4
framing esf
pri-group timeslots 1-24 nfas_d none nfas_int 3 nfas_group 1
controller T1 3/0:5
framing esf
pri-group timeslots 1-24 nfas_d none nfas_int 4 nfas_group 1
controller T1 3/0:6
framing esf
pri-group timeslots 1-24 nfas_d none nfas_int 5 nfas_group 1
controller T1 3/0:7
framing esf
pri-group timeslots 1-24 nfas_d none nfas_int 6 nfas_group 1
interface Serial3/0:1:23
no ip address
encapsulation hdlc
isdn switch-type primary-4ess
no cdp enable
dial-peer voice 150 pots
translation-profile incoming Calling_Party_Num
translation-profile outgoing outgoing_international
destination-pattern 011.T
translate-outgoing called 1
direct-inward-dial
port 3/0:1:D
forward-digits 0
dial-peer voice 100 voip
preference 1
modem passthrough nse codec g711ulaw redundancy
voice-class codec 1
incoming called-number 800.......
dtmf-relay rtp-nte
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
no vad
*Sep 4 17:57:57.433: ISDN Se3/0:1:23 Q931: RX <- SERVICE pd = 3 callref = 0x0000
Change Status i = 0xC0 - in-service
Channel ID i = 0xE9818398
Exclusive, Interface 1, Channel 24
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x6871D9EC; count=1
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_get_profile_from_dialpeer_internal: Error: Invalid input peer_tag=0 direction=incom ing
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x6871D9EC; count=0
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x6871D9EC; count=1
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: number=5024101498 type=unknown plan=unknown numbertype= calling
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_match: No match; number=5024101498 rule precedence=1
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_match_internal: No match found
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: No match: number=5024101498 type=unknown plan=unknown
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: number=011862196990 type=unknown plan=unknown numbertyp e=called
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_match_internal: Matched with rule 1 in ruleset 2
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_match_internal: Matched with rule 1 in ruleset 2
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/sed_subst: Successful substitution; pattern=011862196990 matchPattern=^01 replacePattern=011 replaced pattern=0111862196990
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: xlt_number=0111862196990 xlt_type=unknown xlt_plan=unkn own
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-t arget
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-target number not found
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-c alled
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_dp_translate: calling_number=5024101498 calling_octet=0x0
called_number=0111862196990 called_octet=0x0
redirect_number= redirect_type=0 redirect_plan=0 redirect_PI=-1 redirect_SI=-1
*Sep 4 17:57:59.005: //-1/BC10816380D4/RXRULE/regxrule_vp_translate: No profile found in voice port or trunk group for outgoing direction
*Sep 4 17:57:59.005: //-1/BC10816380D4/RXRULE/regxrule_vp_translate: calling_number=5024101498 calling_octet=0x0
called_number=0111862196990 called_octet=0x0
redirect_number= redirect_type=0 redirect_plan=0
*Sep 4 17:57:59.005: ISDN Se3/0:1:23 Q931: Applying typeplan for sw-type 0x2 is 0x2 0x1, Calling num 5024101498
*Sep 4 17:57:59.005: ISDN Se3/0:1:23 Q931: Applying typeplan for sw-type 0x2 is 0x2 0x1, Called num
*Sep 4 17:57:59.005: ISDN Se3/0:1:23 Q931: TX -> SETUP pd = 8 callref = 0x008E
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xE9868398
Exclusive, Interface 6, Channel 24
Calling Party Number i = 0x2180, '5024101498'
Plan:ISDN, Type:National
Called Party Number i = 0xA1
Plan:ISDN, Type:National
*Sep 4 17:57:59.133: ISDN Se3/0:1:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x808E
Cause i = 0x819C - Invalid number format (incomplete number)
*Sep 4 17:57:59.133: ISDN Se3/0:1:23 **ERROR**: call_cleared: VOICE ERROR: NULL VDEV Common(0xFC): bchan -1, call id 0x8016
*Sep 4 17:57:59.141: //-1/BC10816380D4/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x6871D9EC; count=2
VoiceEncapPeer150
peer type = voice, system default peer = FALSE, information type = voice,
description = `',
tag = 150, destination-pattern = `011.T',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 150, Admin state is up, Operation state is up,
Outbound state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
URI classes:
Destination =
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):Calling_Party_Num
Translation profile (Outgoing):outgoing_international
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
type = pots, prefix = `',
forward-digits 0
session-target = `', voice-port = `3/0:1:D',
direct-inward-dial = enabled,
digit_strip = enabled,
register E.164 number with H323 GK and/or SIP Registrar = TRUE
fax rate = system, payload size = 20 bytes
supported-language = ''
preemption level = `routine'
bandwidth:
maximum = 64 KBits/sec, minimum = 64 KBits/sec
voice class called-number:
inbound = `', outbound = `'
dial tone generation after remote onhook = enabled
Time elapsed since last clearing of voice call statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 15, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "1C ",
Last Disconnect Text is "invalid number (28)",
Last Setup Time = 50369257.
Last Disconnect Time = 0.
T1 3/0:1 is up.
Applique type is Channelized T1
No alarms detected.
alarm-trigger is not set
Soaking time: 3, Clearance time: 10
AIS State:Clear LOS State:Clear LOF State:Clear
Version info of slot 3: HW: 1536, PLD Rev: 6
Framer Version: 0x58
Manufacture Cookie Info:
EEPROM Type 0x0001, EEPROM Version 0x04, Board ID 0x01,
Board Hardware Version 6.0, Item Number 73-4089-07,
Board Revision A0, Serial Number JAE08512FBJ,
PLD/ISP Version 48.53, Manufacture Date 13-Dec-2004.Hello all,
I'm having a problem getting international calls to complete. I have the below in my Cisco 5350 configs with some lines removed to save space. Debug is below that. It appears my translation rule is ok to some extent as the test indicates. What am I overlooking? Any help will be greatly appreciated.
Sprint5350#test voice translation-rule 2 011862196990
Matched with rule 1
Original number: 011862196990 Translated number: 0111862196990
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
voice translation-rule 1
rule 1 /\+1\(.*\)/ /\1/
voice translation-rule 2
rule 1 /^01/ /011/
voice translation-profile Incoming_Calling_Party_Num
translate calling 1
voice translation-profile outgoing_international
translate calling 2
translate called 2
controller T3 3/0
framing m23
clock source line
cablelength 133
t1 1-7 controller
description Sprint DS3 - 75955030
controller T1 3/0:1
framing esf
pri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 1
controller T1 3/0:2
framing esf
pri-group timeslots 1-24 nfas_d none nfas_int 2 nfas_group 1
controller T1 3/0:3
framing esf
pri-group timeslots 1-24 nfas_d backup nfas_int 1 nfas_group 1
controller T1 3/0:4
framing esf
pri-group timeslots 1-24 nfas_d none nfas_int 3 nfas_group 1
controller T1 3/0:5
framing esf
pri-group timeslots 1-24 nfas_d none nfas_int 4 nfas_group 1
controller T1 3/0:6
framing esf
pri-group timeslots 1-24 nfas_d none nfas_int 5 nfas_group 1
controller T1 3/0:7
framing esf
pri-group timeslots 1-24 nfas_d none nfas_int 6 nfas_group 1
interface Serial3/0:1:23
no ip address
encapsulation hdlc
isdn switch-type primary-4ess
no cdp enable
dial-peer voice 150 pots
translation-profile incoming Calling_Party_Num
translation-profile outgoing outgoing_international
destination-pattern 011.T
translate-outgoing called 1
direct-inward-dial
port 3/0:1:D
forward-digits 0
dial-peer voice 100 voip
preference 1
modem passthrough nse codec g711ulaw redundancy
voice-class codec 1
incoming called-number 800.......
dtmf-relay rtp-nte
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
no vad
*Sep 4 17:57:57.433: ISDN Se3/0:1:23 Q931: RX <- SERVICE pd = 3 callref = 0x0000
Change Status i = 0xC0 - in-service
Channel ID i = 0xE9818398
Exclusive, Interface 1, Channel 24
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x6871D9EC; count=1
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_get_profile_from_dialpeer_internal: Error: Invalid input peer_tag=0 direction=incom ing
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x6871D9EC; count=0
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x6871D9EC; count=1
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: number=5024101498 type=unknown plan=unknown numbertype= calling
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_match: No match; number=5024101498 rule precedence=1
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_match_internal: No match found
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: No match: number=5024101498 type=unknown plan=unknown
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: number=011862196990 type=unknown plan=unknown numbertyp e=called
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_match_internal: Matched with rule 1 in ruleset 2
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_match_internal: Matched with rule 1 in ruleset 2
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/sed_subst: Successful substitution; pattern=011862196990 matchPattern=^01 replacePattern=011 replaced pattern=0111862196990
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: xlt_number=0111862196990 xlt_type=unknown xlt_plan=unkn own
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-t arget
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-target number not found
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-c alled
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
*Sep 4 17:57:59.001: //-1/BC10816380D4/RXRULE/regxrule_dp_translate: calling_number=5024101498 calling_octet=0x0
called_number=0111862196990 called_octet=0x0
redirect_number= redirect_type=0 redirect_plan=0 redirect_PI=-1 redirect_SI=-1
*Sep 4 17:57:59.005: //-1/BC10816380D4/RXRULE/regxrule_vp_translate: No profile found in voice port or trunk group for outgoing direction
*Sep 4 17:57:59.005: //-1/BC10816380D4/RXRULE/regxrule_vp_translate: calling_number=5024101498 calling_octet=0x0
called_number=0111862196990 called_octet=0x0
redirect_number= redirect_type=0 redirect_plan=0
*Sep 4 17:57:59.005: ISDN Se3/0:1:23 Q931: Applying typeplan for sw-type 0x2 is 0x2 0x1, Calling num 5024101498
*Sep 4 17:57:59.005: ISDN Se3/0:1:23 Q931: Applying typeplan for sw-type 0x2 is 0x2 0x1, Called num
*Sep 4 17:57:59.005: ISDN Se3/0:1:23 Q931: TX -> SETUP pd = 8 callref = 0x008E
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xE9868398
Exclusive, Interface 6, Channel 24
Calling Party Number i = 0x2180, '5024101498'
Plan:ISDN, Type:National
Called Party Number i = 0xA1
Plan:ISDN, Type:National
*Sep 4 17:57:59.133: ISDN Se3/0:1:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x808E
Cause i = 0x819C - Invalid number format (incomplete number)
*Sep 4 17:57:59.133: ISDN Se3/0:1:23 **ERROR**: call_cleared: VOICE ERROR: NULL VDEV Common(0xFC): bchan -1, call id 0x8016
*Sep 4 17:57:59.141: //-1/BC10816380D4/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x6871D9EC; count=2
VoiceEncapPeer150
peer type = voice, system default peer = FALSE, information type = voice,
description = `',
tag = 150, destination-pattern = `011.T',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 150, Admin state is up, Operation state is up,
Outbound state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
URI classes:
Destination =
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):Calling_Party_Num
Translation profile (Outgoing):outgoing_international
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
type = pots, prefix = `',
forward-digits 0
session-target = `', voice-port = `3/0:1:D',
direct-inward-dial = enabled,
digit_strip = enabled,
register E.164 number with H323 GK and/or SIP Registrar = TRUE
fax rate = system, payload size = 20 bytes
supported-language = ''
preemption level = `routine'
bandwidth:
maximum = 64 KBits/sec, minimum = 64 KBits/sec
voice class called-number:
inbound = `', outbound = `'
dial tone generation after remote onhook = enabled
Time elapsed since last clearing of voice call statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 15, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "1C ",
Last Disconnect Text is "invalid number (28)",
Last Setup Time = 50369257.
Last Disconnect Time = 0.
T1 3/0:1 is up.
Applique type is Channelized T1
No alarms detected.
alarm-trigger is not set
Soaking time: 3, Clearance time: 10
AIS State:Clear LOS State:Clear LOF State:Clear
Version info of slot 3: HW: 1536, PLD Rev: 6
Framer Version: 0x58
Manufacture Cookie Info:
EEPROM Type 0x0001, EEPROM Version 0x04, Board ID 0x01,
Board Hardware Version 6.0, Item Number 73-4089-07,
Board Revision A0, Serial Number JAE08512FBJ,
PLD/ISP Version 48.53, Manufacture Date 13-Dec-2004. -
Need help Creating a translation pattern that adds dial out digits to incoming calls
I came across an article yesterday and it showed the steps how to fix Missed Call/Received Call numbers so that you can dial them from the menu correctly (auto-add a 9, etc.)?
I tried it this morning and came up with this translation pattern:
voice translation-rule 6
rule 1 /^201\(.*\)/ /8\1/
rule 2 /\(..........\)/ /81\1/
voice translation-profile filter_Incoming
translate calling 6
This translation pattern rule 1 adds the dial out character 8 and strips 201 for local calls. Rule 2 adds dial out character 8 and adds 1 for long distance. The purpose of this translation rule is when the ephone receives the phone call the characters 8 and 1 are added so when you quickly need to redial you do not have to edit the number and add 8 for each call.
I tested the translation-rule:
ROUTER-2911#test voice translation-rule 6 9082121231
Matched with rule 2
Original number: 9082121231 Translated number: 819082121231
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
ROUTER-2911#test voice translation-rule 6 2019121231
Matched with rule 1
Original number: 2019121231 Translated number: 89121231
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
ROUTER-2911#
Issue is I am not sure with my inbound call leg if it can even work. We dial out through the SIP Trunk and the incoming is translated to the AutoAttendant on Cisco Unity Express.
voice translation-rule 1
rule 1 /2015552100/ /2003/
voice translation-profile CUE_Voicemail/AutoAttendant
translate called 1
dial-peer voice 9 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming CUE_Voicemail/AutoAttendant
call-block translation-profile incoming BLOCKED-INCOMING
call-block disconnect-cause incoming call-reject
session protocol sipv2
session target dns:nd01-04.fs.SIPPROVIDER.net
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
Can what I am trying to do be done with my current setup?Hi patldmart012,
The dial-peer 9 that you have attached will not be affected by following config
voice translation-rule 6
rule 1 /^201\(.*\)/ /8\1/
rule 2 /\(..........\)/ /81\1/
voice translation-profile filter_Incoming
translate calling 6
Because you have not applied the translation profile "filter_incoming" on the dial-peer.
Could you please provide the exact call flow?
Along with that, If you are facing issue with calls on SIP Trunk, please collect following debugs in a logging buffer and attach the file. I will analyse it and will get back to you.
debug voip ccapi inout
debug ccsip message
debug voice translation
Debug h225 asn1 (If H323 involved)
Debug h245 asn1 (If H323 involved)
Debug MGCP Packets (If MGCP involved)
Also provide the running config of the GW.
These are verbose debugs, so please collect them in the following manner:
Router(config)# service sequence
Router(config)# service timestamps debug datetime msec
Router(config)# logging buffered 30000000 7
Router(config)# no logging con
Router(config)# no logging mon
Router# Clear log
Router# term no mon
<Enable debugs, then wait for issue to occur.>
Router# term len 0
<Enable session capture to txt file in terminal program.>
Router# Undebug all
Router# sh log
Once i have the logs, i will analyse it and will get back to you.
Regards,
Mudit Mathur -
I need config a group in a translation rule and fail, any body can help me
rule 1 ^123(1-9) 3456 any subscriber
Incorrect format for Translation Match Pattern
regular expression must be of the form ^(\^)?(\+)?([0-9,ABCD.*%?#]+)$
Invalid match pattern string input ^123(1-9)If your ver IOS is 12.2(11)T or later this will work
voice translation-rule 5000 /^123[1-9]/ /3456/ type any subscriber
show voice translation-rule 5000
Translation-rule tag: 5000
Rule 1:
Match pattern: ^123[1-9]
Replace pattern: 3456
Match type: any Replace type: subscriber
Match plan: none Replace plan: none
you can do the following to see if your rule works as expected.
test voice translation-rule 5000 1234567890
Matched with rule 1
Original number: 1234567890 Translated number: 3456567890
Original number type: none Translated number type: subscriber
Original number plan: none Translated number plan: none -
Hello,
I have 2 questions below please reply.
I have seen a setup where the directory number are assigned as starting with " * " *5234, *5123,*5432 , why that so ???
CUCM 6.1 is integrated with AD, when i search a user in corporate directory his full number is seen 22221234 but when i press dial button the calls go to last 3 digit extension, how come ????
ThanksDear
" so that users wouldn't be able to accidentally dial it or something like that" accidentally dial it means can you elaborate more.
For your second question you may have a Translation Pattern or Transformation Pattern, changing the dialed number or appearance of the dialed number.
there are no transformation number but there are translation number but i am not sure they are doing these job , i will confirm and i will post again.
Thanks -
CME number stripping for outgoing calls
Hello all, I hafe a problem with CME for outgoing calls to the PSTN. Internalnumbers are 3 digits long. Numbers from PSTN are 7 digits long. Incvommin calls working fine. Outgoing calls are working also, but the first 4 digist from the nummer are twice in the calling number on the reciver side. Is there any workaround to fix this? Any suggestion are wellcom.
PS. Sitting in germany!Hi Mehmet,
I know there are better ways to do this. Thank you for your suggestion. I will change to translation rules.
I have other problems probably you can help me. I have a translation-rule for incoming calls to show the leading 0's.
voice translation-rule 11
rule 1 /\(^.*\)/ /0\1/ type national national
rule 2 /\(^.*\)/ /00\1/ type international international
voice translation-profile 0prefix
translate called 11
dial-peer voice 3 pots
translation-profile incoming 0prefix
If I test the rule all working fine, but in real the rule dos not work.
test voice translation-rule 11 1243 type international
Matched with rule 2
Original number: 1243 Translated number: 001243
Original number type: international Translated number type: international
Original number plan: none Translated number plan: none
test voice translation-rule 11 1243 type nat
Matched with rule 1
Original number: 1243 Translated number: 01243
Original number type: national Translated number type: national
Original number plan: none Translated number plan: none
I am not sure I have a problem with the IOS.
Greetings
Rene -
Translation Pattern Wildcard Match
Our organization uses 5 digit internal extensions throughout. Our CEO would like the ability to dial any 5 digit extension in our organization but wants his caller id to be shown as his name and the extension of his secretary – basically masking his 5 digit extension. I believe the simplest way to achieve this is to create a Translation Pattern, but I’m having an issue trying to match the wildcards in a TP in CUCM7.1.5. At this stage I have set up a new Partition and CSS just for the CEO’s phone and placed a test phone in the new CSS. I then created a TP which is where I run into a problem.
In the TP I have selected the proper partition and in the Calling Party Transformations section I have listed the Calling Party Transform Mask as the secretary extension (we’ll say 55555 for this example). When I use an exact Translation Pattern match (say 12345) the translation works as I would expect (when I dial 12345 from the test phone, the caller ID shows as 55555). However, when I use any wildcards in the Translation Pattern (i.e. XXXXX) the translation does not occur. Now when I dial 12345 the true caller ID number shows instead of the translated number.
I’m basically looking for a catch all rule from the CEO’s phone that will translate to 55555. I’m guessing I’m overlooking something simple here – any assistance? Thanks in advance.I set up a calling party transformation pattern with the same results. The issue seems to be in matching the dialed pattern or Translation Pattern field. In my testing the pattern is matched only when it's exact and not when wildcards are used. See the first attached screen shot where the pattern is '12345'. When this is applied it works as would be expected and the caller ID on the receiving phone shows 55555. But, on the second attached screenshot using wildcards, when 12345 is dialed the caller ID shows as the number on the phone and not the translated value. For some reason the wildcards don't seem to match.
I've tried various wildcard patterns such as XXXXX, 1234X, and [0-8]XXXX - none work. The last one is the one I'd really like to use. Other thoughts or suggestions? -
Read-only radio group losing value on validation error
It took quite a while to track this down, but I think I've got it narrowed down properly. I have an APEX form for updating a record in a databse table with a radio group which is conditionally read-only. If that radio group has a value AND is read-only, when the form is submitted and hits a validation error, the value of the radio group is lost.
For example, the radio group SUBMITTED_FOR_APPROVAL is set to Y and is read-only for a given user. That user then changes something else on the form and submits it. However, the form now hits a validation error for some field. When the form reloads with the validation error displayed, all the fields are restored except the read-only radio group which is now blank. If a select list is used with the identical read-only condition, it works fine. Likewise, if a radio group is used, but the read-only condition is removed, it works fine. It is only when it is a radio group and it is read-only, that the value does not appear to be submitted to the session with all the form values whent he form is submitted.
Is this a bug or am I simply missing something?
Apex 3.1.2
Rgds/Mark M.It's using a shared component LOV cleverly called LOV_YN which consists of a static LOV with
1 Display=Yes, Return=Y
2 Display=No, Return=N
The settings in the LOV section on the item iteself:
Named LOV: LOV_YN
Display Extra values: No Dynamic translation: Not translated
Number of columns: 2 Display null: No
Null display value is blank as is null return value
Item was setup as a radio group and was only converted over to a select list to work around this issue. Let me know what else you need and thanks again.
Rgds/Mark M. -
Conditional read only radiogroup item losing value when saving
Apex 3.2
I have a radiogroup item that is set to read only after it's populated and the user is not in the ADMIN Role. When non ADMIN users edit the page, it shows the value and they cannot change it, but if they save the page, it loses the value. Works fine if the user has ADMIN role.
This is the Pl/SQL function in the "Item is read only when" section
begin
if :p3_event_detail_id is not null and :p3_notification_period_id != -1 and not apex_util.public_check_authorization('ADMIN') then
return true;
else
return false;
end if;
end;
There is also a Not Null validation on this radiogroup item.
I changed the item type to select list and it works fine, but user like the radiogroup button.
Any ideas?It's using a shared component LOV cleverly called LOV_YN which consists of a static LOV with
1 Display=Yes, Return=Y
2 Display=No, Return=N
The settings in the LOV section on the item iteself:
Named LOV: LOV_YN
Display Extra values: No Dynamic translation: Not translated
Number of columns: 2 Display null: No
Null display value is blank as is null return value
Item was setup as a radio group and was only converted over to a select list to work around this issue. Let me know what else you need and thanks again.
Rgds/Mark M. -
Read Only for "Group" and "Others" via SMB
I see when Windows vista users of my OS X 10.5.8 server save a file to the server, the permissions show on the server as Read & Write for the owner and Read Only for 'Group' and 'Others'.
Is there a way I can make the OS X server always make 'Group' and 'Others' always be Read & Write?
Thanks!It's using a shared component LOV cleverly called LOV_YN which consists of a static LOV with
1 Display=Yes, Return=Y
2 Display=No, Return=N
The settings in the LOV section on the item iteself:
Named LOV: LOV_YN
Display Extra values: No Dynamic translation: Not translated
Number of columns: 2 Display null: No
Null display value is blank as is null return value
Item was setup as a radio group and was only converted over to a select list to work around this issue. Let me know what else you need and thanks again.
Rgds/Mark M. -
UC560 T1E1 with Direct TCP SIP
Hi,
Firstly I am new to cisco's voice setup, and have a UC560-T1E1-K9 which I need to configure as just a basic gateway. Essentially i want to setup trunk-to-trunk between ISDN PRI and TCP SIP but do not have the first idea where to start.
The main aim is to receive calls over ISDN PRI (E1 NET5) and pass the call stright through to a direct tcp sip trunk (incoming only).
Then configure FXS for a few numbers for two way calling.
Hope someone can give me a couple of pointers as the CUE is not very helpful
Thanks,
Alex.Hi,
You have two approaches.
Option A
Use CCA to configure the unit from scratch. Configure the unit for your ISDN connection and test using the fxs ports. Manually configure a SIP dial-peer to route inbound PSTN calls to your Lync server. You'll need to tweak the configuration at various places to make it work. You therefore need to be pretty confident in understanding all the work CCA has done for you. The benefit this, is you have a local outbound dial-plan pre-configured on the unit.
Option B
Configure the unit yourself from scratch. Rough outline:
1. Configure E1 controller for voice
2. Setup the DSP modules for your region
3. configure a pots inbound dial peer to accept calls from the PSTN
4. configure an outbound SIP dial-peer to send the calls to Lync.
Lync expects calls to be prefixed with a + & E.164 and bare in mind the telco may only send you the last 6 sigits of the called number
Here's an example for UK ISDN with 30 timeslots: note module slots will be incorrect in this code, example is generic IOS and not specific to UC560.
1. Configure E1 controller for voice
card type e1 0
network-clock-participate wic 0 ! something like this, you may need to work out which slot you have
network-clock-participate slot 1 ! or something like this
network-clock-select 5 E1 1/0 ! you get the idea....
controller E1 1/0
pri-group timeslots 1-31
interface Serial1/0:15 ! do a show run to check the right interface slot number
description "use dial-peer voice 10 to reach me for outbound"
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn timer T309 400000
isdn incoming-voice modem
isdn T309-enable
isdn send-alerting
isdn sending-complete
no cdp enable
Your E1 card should now be up
c2821.1.hex#show controllers e1
E1 1/0 is up.
Applique type is Channelized E1 - balanced
Cablelength is Unknown
No alarms detected.
alarm-trigger is not set
Version info Firmware: 20090408, FPGA: 13, spm_count = 0
Framing is NO-CRC4, Line Code is HDB3, Clock Source is Line.
CRC Threshold is 320. Reported from firmware is 320.
Data in current interval (652 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
Total Data (last 24 hours)
0 Line Code Violations, 0 Path Code Violations,
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
2. Setup the DSP modules for your region
voice-port 1/0:15
cptone GB
bearer-cap Speech
3. configure a pots inbound dial peer to accept calls from the PSTN
I suggest at this point, you debug isdn q931 and dial in. See how many digits the telco sends you.
Configure a number translation to put this number in to E.164
In my example we take the 6 digits the telco sends us and prepend 441234
voice translation-rule 10
rule 1 /^\(.+\)$/ /441234\1/
voice translation-profile incomingisdn
translate called 10
in exec mode:
SOV_TAG1#test voice translation-rule 10 567890
Matched with rule 1
Original number: 567890 Translated number: 441234567890
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
dial-peer voice 1000 pots
description Inbound POTS dial-peer
translation-profile incoming incomingisdn
incoming called-number .+
direct-inward-dial
port 1/0:15
4. configure an outbound SIP dial-peer to send the calls to Lync. Lync expects calls to be prefixed with a + & E.164 so we're going to add a plus
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 20
rule 1 /\(.+\)/ /+\1/
voice translation-profile addaplus
translate called 20
in exec mode
#test voice translation-rule 20 123
Matched with rule 1
Original number: 123 Translated number: +123
dial-peer voice 1010 voip
description SIP Trunk to Lync
translation-profile outgoing addaplus
preference 5
destination-pattern ^441234.+ ! note this is the area code we added to make the number into E164
voice-class codec 1
voice-class sip outbound-proxy ipv4:192.168.10.10 ! ip address of lync
session target dns:lync.mydomain.net ! your lync realm
session protocol sipv2
session transport tcp
dtmf-relay rtp-nte
no vad
and that's about it apart from getting the fx0 ports sorted - one step at a time.
For outbound you added an incoming SIP dial-peer and an outgoing pots dial peer.
Adam
VoIP.co.uk -
Insert no#3 and remove the last 3 digits...
Dear All,
I would like to issue a voice translation-rule that can insert the digit number 3 before/after certain digits and also remove the last 3 digits at the same time.
# Details:
I receive the following Dialed numbers on a voice port for example: 004925147260123 For these numbers I have 3 sets:
- Set 1: 004925147 is always the same; I need to insert the digit number 3 just after 004925, for my example 004925 --> must be 0049253
- Set 2: consists of 3 digits can vary so I want the translation rule to translate these numbers to be the same numbers, for my example 260 --> must be 260
- Set 3: The last 3 digits also they can vary but I would like to remove these last 3 digits from the final translated number
So for my example I would like the voice translation-rule to translate the dialed no# 004925147260123 so that it will be 0049253147260
Appreciate you support,
Thanks..Thanks, the way i want it can be done like this:
rule 1 /^\(004925\)\(......\)...$/ /\13\2/
But you gave me the right way..
Appreciate your support..
Thanks.
Maybe you are looking for
-
Problem when creating a Report with a schduled date
When I tried to create a report with a Schedule (any option except NOW) I get the error of "First Report Occur Date should be after or equal to Current Date" unless I use a date of 2012-10-01 or later in the First report occurs on field. Today's dat
-
How to Create a DVD With No Menu?
hi there. i've had DVD Studio Pro for a while and never used it. would it be to hard to give me a step-by-step guidance on how to create a DVD from a QuickTime file in DVD Studio Pro (initially, in the form of a VIDEO_TS folder or IMG file) with no m
-
Regarding mandatory fields, Context Objects and Fault Message Types
Hi All, 1) I am creating a structure with fields "Name", "Street" and "City". While creation i want to make "Name" fields as mandatory. Is it possible. If so how to achieve this. 2) What is the purpose of Context Object and in which situation we will
-
I have had my 9105 18 months and just love it - had a problem with keypad last year but managed to sort and the numerical key pad with traditional predictive 0-9 keypad text entry is fantastic for me to text email or BBM - or anything requiring text
-
SSL exception: Duplicate extensions not allowed
Hi, I have problem with connecting to exchange mail server with java(java mail) I get this exception. javax.net.ssl.SSLProtocolException: java.io.IOException: Duplicate extensions not allowed. Caused by: java.io.IOException: Duplicate extensions not