Hi Samiulla, So lets go by requirement by requirement and see how we can achieve. So lets say we have ephone 1 for receptionist and ephone 2 for manager 1) requirement 1 . Receptionist wants us to configure her extenstion in such a way that, if a call hits the primary line of managers phone, her extension should ring as well ephone-dn 1   dual-line             ephone-dn 2  dual-line number 3XXX                           number 4xxx                             I am making both the dns as dual-line (you can make them octo-line as well) as you might need to have conference and transfer facility on both dn , with single channel it wont allow u to have more than one call per dn, so no conference and transfer will be available ephone 1                      Receptions phone button 1:1 2:2 ephone 2                       Manager's Phone button 1:1 2:2 Call Comes in on 1XXX, since it will ring both the phones (lines/dn shared) on button 1 2) If there is an active call on managers primary line, this call should go to his secondary line (4xxx) and the same should be indicated on her phone as well ephone-dn 1 number 3XXX    call-forward busy 4xxx huntstop channel             ----->otherwise the second call will come on second channel and second channel is not busy 3) if none of them answer this call, it should finally go to managers voicemail box. ephone-dn 2  dual-line number 4xxx Call-forward noan Let me know if you have any questions. HTH PS:please rate useful posts. Regardsm sumit ... View more

Hi Abu, Awesome Document I must say .Very Nice work.Really appreciate all your efforts . Many Thanks. Regardsm sumit ... View more

Below is the Criteria to Match Inbound Dial-peer for Voip dial-peers 1) Incoming called-number : It will look for the this command in the dial-peers , it will match the number in tis command against the called number(DNIS) 2) Answered Address command L: If Incoming called numbeer command is  not there then it will look for answered addrss command .it will match the number in this command against the calling number(ANI) 3) Destination Pattern :If Answered address is also not there then it will look for answered addrss command .it will match the number in this command against the calling number(ANI) So If you have only one dial-peers and you want to make it as inbound dial-peers for outbound calls and outgoing dial-peers for inbound calls,you can configure a dial-peer like this Dial-peer voice 1 voip incoming called number .                          < it will match with every called number > destination-patern 2...                               session target ipv4: While making a call , you can run debug voip ccapi inout to see which dial-peers are being matched as incoming dial-peer and outgoing dial-peers HTH PS: Please rate useful posts. Regards, sumit ... View more

Yes Voip Dial-peers are used to communicate on Ip legs (between CUCM and Voice Gateway) and POTS dial-peers used to communicate between VGW and PSTN IP Phone---CUCM---(matches RP--> RL--->RG)---pointing towards the voice gateway but for this call leg it has to match the inbound dial-peer. So the voip dial-peer will match as an inbound dial-peer and the pots dial-peer will be outbound dial-peer FOr both inbond call leg and outbound call leg , it matches inbound dial-peer and outbound dial-peers respectively. So for outbound call from IP phone IP phone---CUCM------>Inbound Voip dial-peer---->VGW---pOTS outbound dial-peer --->PSTN For incming call from PSTN           IP phone------CUCM<-------Outbound Vop dial-peer<--------VGW<-----Inbound Pots Dial-peer-<---PSTN Same Dial-peer can act as Inbound dial-peer and Outbound Dial-peer HTH PS:Rate useful posts. Regards, sumit ... View more

Hi Bell, Yes it does support SRST, you need to configure SIP SRST as suggested by Adrian here is the sample working config from my lab voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip   bind control source-interface GigabitEthernet0/1   bind media source-interface GigabitEthernet0/1   registrar server voice register global------------------------> mode is SRST by default unless you change it to cme max-dn 10 max-pool 5 voice register pool  1 id network 14.160.120.0 mask 255.255.255.0             ----->Subnet of sip  phones you want to register to this router application sip.app dtmf-relay rtp-nte codec g711ulaw HTH PS:Rate useful Posts Regards, sumit ... View more

Please check this bug . CSCtg47522  SCCP CIPC and IP Phone 7970 will not failover to SRST 8.0/8.1 Version 7.0(3)  Original-version 7.0(3) To-be-fixed 9.9(9) 8.6(1)  CIPC 7.0(3) and 7.0(3.3) uses an unsupported phone firmware--version 8.3(5)--that sends malformed LineStatV2Message which results to not register with SRST 8.0 ----------- CIPC uses the wrong format for the LineStatV2Message sent during registration to SRST/CME. This causes it not to register with SRST/CME 8.0. CIPC will display "CM Fallback Service Operating" and will never display the line information. The phone will eventually unregister and get stuck in a loop. Symptom: The phone displays the message "CM Fallback Service Operating" in the status area.  Button Lines do not display an extension number.. Conditions: CIPC 7.x SRST router running IOS 15.0(1)XA or 15.1(1)T Workaround: CIPC uses a phone firmware version that currently does not support SRST 8.0.  Please downgrade IOS/SRST version to 15.0(1)M Description *************************************** ---------------- Per Nancy Chen suggestion, fixed committed into 15.1(1)T2 and 15.1(2)T2. Retest 1) 15.1(1)T2 = CME 8.0. Both CIPC 8.5.0.x and IP Phone 7970 SCCP 9.1.1 do not failover, CUCM version 8.5.1.1000-23 2) 15.1(2)T2 = CME 8.1.CIPC 7.0(3) and 7.0(3.3) uses an unsupported phone firmware--version 8.3(5)--that sends malformed LineStatV2Message which results to not register with SRST 8.0 ----------- CIPC uses the wrong format for the LineStatV2Message sent during registration to SRST/CME. This causes it not to register with SRST/CME 8.0. CIPC will display "CM Fallback Service Operating" and will never display the line information. The phone will eventually unregister and get stuck in a loop. Tin Luong 04/12/2011 ---------------- Per Nancy Chen suggestion, fixed committed into 15.1(1)T2 and 15.1(2)T2. Retest 1) 15.1(1)T2 = CME 8.0. Both CIPC 8.5.0.x and IP Phone 7970 SCCP 9.1.1 do not failover, CUCM version 8.5.1.1000-23 2) 15.1(2)T2 = CME 8.1. It could be this bug HTH PS: Rate useful posts Regards, sumit ... View more

If you look in the debug Cisco Support Community Jun  6 10:03:51.279: //-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:    Interface=0x471599EC, Interface Type=9, Destination=0.0.0.0, Mode=0x0,    Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),    Called Number=(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,    Subscriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=0, Call Count On=FALSE,    Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=) Jun  6 10:03:51.279: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa: Jun  6 10:03:51.279: :cc_get_feature_vsa malloc success Jun  6 10:03:51.279: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa: Jun  6 10:03:51.279:  cc_get_feature_vsa count is 1 Jun  6 10:03:51.279: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa: Jun  6 10:03:51.279: :FEATURE_VSA attributes are: feature_name:0,feature_time:1219807008,feature_id:13882 Jun  6 10:03:51.279: //1163276/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:    SPI Call Setup Request Is Success; Interface Type=9, FlowMode=1 Jun  6 10:03:51.279: //1163276/xxxxxxxxxxxx/CCAPI/ccCallSetContext:    Context=0x47D299A4 Jun  6 10:03:51.283: //1163276/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:    Interface=0x471599EC, Data Bitmask=0x0, Progress Indication=NULL(0),    Connection Handle=0 Jun  6 10:03:51.283: //1163276/xxxxxxxxxxxx/CCAPI/cc_api_call_connected:    Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0) Jun  6 10:03:51.299: //1163277/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind:    Call Entry Is Not Found Jun  6 10:03:51.299: //-1/1D07A340882A/CCAPI/cc_api_display_ie_subfields:    cc_api_call_setup_ind_common:    cisco-username=0669    ----- ccCallInfo IE subfields -----    cisco-ani=0669    cisco-anitype=0    cisco-aniplan=0    cisco-anipi=0    cisco-anisi=0    dest=907703722056    cisco-desttype=0    cisco-destplan=0    cisco-rdie=FFFFFFFF    cisco-rdn=    cisco-rdntype=0    cisco-rdnplan=0    cisco-rdnpi=-1    cisco-rdnsi=-1    cisco-redirectreason=-1   fwd_final_type =0    final_redirectNumber =    hunt_group_timeout =0 Jun  6 10:03:51.299: //-1/1D07A340882A/CCAPI/cc_api_call_setup_ind_common:    Interface=0x47283FE0, Call Info(    Calling Number=0669,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),    Called Number=907703722056(TON=Unknown, NPI=Unknown),    Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,   Incoming Dial-peer=1000, Progress Indication=NULL(0), Calling IE Present=TRUE,    Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=1163277 Jun  6 10:03:51.299: //-1/1D07A340882A/CCAPI/ccCheckClipClir:    In: Calling Number=0669(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed) Jun  6 10:03:51.299: //-1/1D07A340882A/CCAPI/ccCheckClipClir:    Out: Calling Number=0669(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed) Jun  6 10:03:51.299: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa: Apply that on the dial-peer voice 1000 voip the translation profile. now apply that voice translation profile on dial-peer vocie 1000 Q-RTR(config)#voice translation-rule 17 HQ-RTR(cfg-translation-rule)#rule 1 /^0669$/ /08708630002/ HQ-RTR(cfg-translation-rule)#exit voice-translation profile change_ani translate calling 17 dial-peer voice1000 voip translation-profile outgoing change_ani. If this does not work and if you can not send me the running config. then Please open a TAC case. Regards, sumit ... View more

ok create th voice translation-rule for this Q-RTR(config)#voice translation-rule 17 HQ-RTR(cfg-translation-rule)#rule 1 /^0669$/ /08708630002/ HQ-RTR(cfg-translation-rule)#exit voice-translation profile change_ani translate calling 17 dial-peer voice 0669 voip translation-profile outgoing change_ani please send me the running config and debug voip ccapi inout. Thanks. Regards, sumit ... View more

Hi Nathan, Can you please answer the questions below 1 ) What is your extesion to which your Ip phone is regsitered with call manager 2)  what is the number you want to be displayed on your mobile 3)  Have you chcked if its matching the correct incoming dial-peer where voice translation profile is bing appplied 4)  Please check debug voip ccai inout to see whats the number it is translating to I think there is some misconfiguration . Regards, sumit ... View more

it needs to be HQ-RTR(config)#voice translation-rule 17 HQ-RTR(cfg-translation-rule)#rule 1 /^0669$/ /0870863\0/ HQ-RTR(cfg-translation-rule)#exit you can check which number is being translated to what. HQ-RTR#test voice tran HQ-RTR#test voice translation-rule 1 0669 Error: Rule set 1 not found HQ-RTR#test voice translation-rule 17 0669 Matched with rule 1 Original number: 0669   Translated number: 08708630669 Original number type: none      Translated number type: none Original number plan: none      Translated number plan: none it will change the number 0669 to 08708630669.I guess thats what your requirement is right ???? ... View more

oice translation-rule 17 rule 1 /^0669$/ /0870863XXXX/---->will never work you cannot have a wild card on the second part of it Regards, sumit ... View more

looks to be a codec issue, software conference bridge only supports G711ulaw  Now if the region settings between CFB  and Br1 is G729 , make sure to check the device pool of CFB and check in their device pool whether it has the mrgl , mrg which contains the transcoder , so the transcder needs to be present in the mrgl of the conference bridge, since Higher Bandwidth codec(G711) always transcodes to lower bandwidth codec(g729) and will invoke the transcoder.So check if the mrgl of the cfb (device pool) contains the transcoder , if not add it . HTH Rate if you find this info useful. Regards, sumit ... View more

Hi So If you want to change only for 2 numbers then What you can do is you can create translation patterns which will be matched before matching route patterns Say your first phone is 123456 and assign a CSS A and create a translation pattern say 9.! and put that in partition A and create calling party transformation in Translation pattern.This has changed your calling party for this phone before matching Route pattern Second phone is 456890 and assign a CSS B and create a translation pattern say 9.! and put that in partition B and create calling party transformation in this Translation pattern .This has changed your calling party for Second phone before matching route pattern. CSSA contains PT A          and          CSS B contain PT B Now assign the CSS of both translation pattern the same CSS which you assign for all the other phones apart from 2 phones which inturn will match the Route pattern Since Its a long and a bit complicated procedure thats why i advised you to create a voice translation rule on the gateway which simplifes the job . Rest is up to you. Regards, sumit ... View more

Hi Nathan, you can do either on CUCM or CUBE On CUbe you can do a voice-translation rule 1 rule 1 /^123456789$/ /987654321/                       ---------------------->whenever it will see number123456789 it will translate to 987654321 then create a voice-translation profile change_ani translate calling 1--------------------------------------------------->this will translate the caling number according to the rule 1 and apply that voice-translation profile on say inbound dial-peer dial-peer dial-peer voice 1 voip translation-profile outgoing change_ani                      --------------> this will change the calling number for the outbound                                                                                call accoridng to the voice translation profile HTH PS: Rate if you find the information useful. Regards, sumit ... View more

enable dtmf relay h245 alpha number numric on the dial-peers and enable debug h245asn1 and debug cch323 h245 and see if you can see the dtmf in h245 debugs. Also check if each digit is sent once and not twice. HTH PS:Please Rate useful posts. Regards, sumit ... View more