2811 H.323 Voice Gateway with CM6 stripping digit

[jjcollaguazo]

Hi,

I'm trying to migrate a CME router to CM and the main number for the site is 66280200 which gest translated to ext 9201 which is AA. The 4 digit extension range for this site is 28.. as per my translation rules.

The issue is for some reason the leading 0 from the incoming call gets stripped.

DEBUG:

*May 15 06:51:31.151: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref = 0x003C
        Sending Complete
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transfer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18381
                Preferred, Channel 1
        Progress Ind i = 0x8283 - Origination address is non-ISDN
        Calling Party Number i = 0x1183, '61415447240'
                Plan:ISDN, Type:International
        Called Party Number i = 0x81, '0201'
                Plan:ISDN, Type:Unknown
*May 15 06:51:31.155: ISDN Se0/0/0:15 Q931: Received SETUP  callref = 0x803C callID = 0x000B switch = primary-net5 interface = User
*May 15 06:51:31.175: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x1 0x1, Calling num 61415447240
*May 15 06:51:31.179: ISDN Se0/0/0:15 Q931: Sending SETUP  callref = 0x0089 callID = 0x800A switch = primary-net5 interface = User
*May 15 06:51:31.179: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x803C
        Channel ID i = 0xA98381
                Exclusive, Channel 1
*May 15 06:51:31.179: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref = 0x0089
        Sending Complete
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transfer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA9839F
                Exclusive, Channel 31
        Progress Ind i = 0x8283 - Origination address is non-ISDN
        Calling Party Number i = 0x1183, '61415447240'
                Plan:ISDN, Type:International
        Called Party Number i = 0x81, '201'
                Plan:ISDN, Type:Unknown
*May 15 06:51:31.259: ISDN Se0/0/0:15 Q931: RX <- STATUS pd = 8  callref = 0x8089
        Cause i = 0x82E4 - Invalid information element contents
        Call State i = 0x01
*May 15 06:51:31.259: ISDN Se0/0/0:15 Q931: RX <- STATUS pd = 8  callref = 0x8089
        Cause i = 0x82E4 - Invalid information element contents
        Call State i = 0x01
*May 15 06:51:31.263: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8  callref = 0x8089
        Channel ID i = 0xA9839F
                Exclusive, Channel 31
*May 15 06:51:31.267: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8  callref = 0x8089
        Cause i = 0x829C - Invalid number format (incomplete number)
        Progress Ind i = 0x8288 - In-band info or appropriate now available
*May 15 06:51:31.271: ISDN Se0/0/0:15 Q931: call_disc: PI received in disconnect; Postpone sending RELEASE for callid 0x800A
*May 15 06:51:31.287: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8  callref = 0x0089
*May 15 06:51:31.291: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  callref = 0x803C
        Cause i = 0x809C - Invalid number format (incomplete number)
*May 15 06:51:31.311: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x8089
        Cause i = 0x829C - Invalid number format (incomplete number)
*May 15 06:51:31.327: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref = 0x003C
        Cause i = 0x809C - Invalid number format (incomplete number)
*May 15 06:51:31.331: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x803C^Z


Here is the relvant config. please let me knwo if more information is required.

interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
no cdp enable
!
voice translation-rule 1
rule 1 /0200/ /9201/
rule 2 /^02/ /28/
!
voice translation-rule 2
rule 1 /^662802/ /28/
!
voice translation-rule 66
rule 1 /^\([^0].*\)/ /00\1/
rule 2 /^\(0[^0].*\)/ /0\1/
!
!
!
!
!
voice translation-profile SRST
translate calling 66
translate called 1
!
voice translation-profile SRST-VM
translate calling 2
!
voice translation-profile calling
translate calling 66
translate called 1
!
dial-peer voice 1 pots
voice cut-through alert
description Inbound DID via PSTN
incoming called-number 662802..
direct-inward-dial
port 0/0/0:15
forward-digits all
!
dial-peer voice 3 pots
description Outbound to PSTN
destination-pattern 0T
port 0/0/0:15
!
dial-peer voice 10 voip
description Inbound CM-Subscriber Peer
translation-profile outgoing calling
destination-pattern 662802..
session target ipv4:10.1.201.11
voice-class codec 1
voice-class h323 1
max-redirects 10
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 11 voip
description Inbound CM-Publisher Peer
translation-profile outgoing calling
preference 2
destination-pattern 662802..
session target ipv4:10.1.201.10
voice-class codec 1
voice-class h323 1
max-redirects 10
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 66 voip
description Inbound SRST Peer
translation-profile outgoing SRST
preference 3
destination-pattern 662802..
session target ipv4:10.1.245.1
voice-class codec 1
voice-class h323 1
max-redirects 10
dtmf-relay h245-alphanumeric
no vad
!
!
dial-peer voice 67 voip
description SRST CUE Voicemail
translation-profile outgoing SRST-VM
destination-pattern 9200
session protocol sipv2
session target ipv4:10.1.245.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 68 voip
description SRST AA
destination-pattern 9201
session protocol sipv2
session target ipv4:10.1.245.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 69 voip
description SRST GMS
destination-pattern 9202
session protocol sipv2
session target ipv4:10.1.245.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 70 voip
description SRST MainNumber
destination-pattern 2800
session protocol sipv2
session target ipv4:10.1.245.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
!
sip-ua
mwi-server ipv4:10.1.245.2 expires 3600 port 5060 transport udp unsolicited
!
!
no telephony-service
!
!
!
call-manager-fallback
secondary-dialtone 0
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 10.1.245.1 port 2000
max-ephones 10
max-dn 20 preference 3
system message primary SRST Mode: Phones in Fallback
dialplan-pattern 1 662802.. extension-length 4 no-reg
voicemail 9200
no huntstop
call-forward busy 9200
call-forward noan 9200 timeout 10
time-zone 39
time-format 24
date-format dd-mm-yy

[paolo bevilacqua]

Your DP 3 is causing that.

Recommend you reconfigure everythin from scratch, as there are many mistakes in current config.

[jjcollaguazo]

Thanks for your reply Paolo, but can you please explain? I've got the exact same config for other VGs with CM and DP 3 works fine, the same with the other DPs and TPs.

The translation rule 1 worked perfectly when applied to dial-peer 1 when this gateway was configured for CME. To me it seems that there's and issue with the CM dial peers and CM, but the gateway has registered succesfully as an H.323 gateway and its definitely configured for 4 significant digits.

The other thing i might mention is that this VG has CUE which will be reconfigured for local VM and AA. CTI route points have been configured to match DP, but wont start CUE config until this incoming call issue is fixed

Hope you can help.

Look forward to your reply.

Cheers,

James

[paolo bevilacqua]

Telco is sending 4 digits, not the full number.

Most translations are wrong.

Think in logical sequence and you will see what needs to be done.

[jjcollaguazo]

i don't know its because i've been starting at the problem for so long but your response itsn't very clear to me.

I'd really appreciate if you could show me exactly what needs to be fixed, if i can't fix this issue soon i have to roll back to CME.

[ADAM CRISP · Accepted solution]

As Paolo says, the telco is sending you the last four digits only.

You haven't got a POTS incoming dial-peer that is configured for this, so as a result the ISR is falling back to using the destination pattern as a matching rule, and stripping  the leading zero. the router is then sending the calls straight back to the telco.

You can manipulate  the dialled digits and cli as the calls passes through the system.

As it stands, right now, you need to re-write both your incoming pots dial-peers and your outbound sip dial-peers, which is why it's been suggested to start again.

Start with something like this:

dial-peer voice 1 pots
voice cut-through alert
description Inbound DID via PSTN
incoming called-number ^02..
direct-inward-dial
port 0/0/0:15
forward-digits all
!

This is really clear that you're accepting incoming POTS calls starting 02 and with length 4 digits.

I would be tempted at this point to sort out the number translations straight away on the incoming pots call get, so your could...

dial-peer voice 1 pots
voice cut-through alert
description Inbound DID via PSTN
translation-profile incoming IncomingPOTS
incoming called-number ^02..
direct-inward-dial
port 0/0/0:15
forward-digits all
!

voice translation-profile IncomingPOTS
translate called 1
translate calling 66

They it's a simple matter or producing some nice and easy outgoing SIP dial-peers to your CUCM

dial-peer voice 10 voip
description MY calls to AA
destination-pattern 9201
session target ipv4:10.1.201.11
voice-class codec 1
voice-class h323 1
max-redirects 10
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 11 voip
description my calls to DID's
destination-pattern ^28..
session target ipv4:10.1.201.10
voice-class codec 1
voice-class h323 1
max-redirects 10
dtmf-relay h245-alphanumeric
no vad
!

don't just cut and paste the above in please...

thanks

[paolo bevilacqua]

Exactly.

I've rated the post above, and the OP should do as well.

[jjcollaguazo]

Hi Adam,

Thanks for your suggestion, but i'm still having no luck.

The call seems to come throuhg correctly now (no digit stripping) but i get a number disconnected message.

Debug isdn Q931 and debug voice ccapi inout

*May 15 23:50:00.688: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref = 0x0056
        Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18394
                Preferred, Channel 20
        Calling Party Number i = 0x1183, '61294610120'
                Plan:ISDN, Type:International
        Called Party Number i = 0x81, '0203'
                Plan:ISDN, Type:Unknown
*May 15 23:50:00.692: ISDN Se0/0/0:15 Q931: Received SETUP  callref = 0x8056 callID = 0x000E switch = primary-net5 interface = User
*May 15 23:50:00.700: //-1/E19AE4A2800F/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=61294610120
   cisco-anitype=1
   cisco-aniplan=1
   cisco-anipi=0
   cisco-anisi=3
   dest=0203
   cisco-desttype=0
   cisco-destplan=1
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-lastrdn=
   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

*May 15 23:50:00.700: //-1/E19AE4A2800F/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x48595570, Call Info(
   Calling Number=61294610120,(Calling Name=)(TON=International, NPI=ISDN, Screening=Network, Presentation=Allowed),
   Called Number=0203(TON=Unknown, NPI=ISDN),
   Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
*May 15 23:50:00.700: //-1/E19AE4A2800F/CCAPI/ccCheckClipClir:
   In: Calling Number=61294610120(TON=International, NPI=ISDN, Screening=Network, Presentation=Allowed)
*May 15 23:50:00.700: //-1/E19AE4A2800F/CCAPI/ccCheckClipClir:
   Out: Calling Number=61294610120(TON=International, NPI=ISDN, Screening=Network, Presentation=Allowed)
*May 15 23:50:00.700: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*May 15 23:50:00.700: :cc_get_feature_vsa malloc success
*May 15 23:50:00.700: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*May 15 23:50:00.700:  cc_get_feature_vsa count is 1
*May 15 23:50:00.700: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*May 15 23:50:00.700: :FEATURE_VSA attributes are: feature_name:0,feature_time:1233036600,feature_id:25
*May 15 23:50:00.704: //21/E19AE4A2800F/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=61294610120(TON=International, NPI=ISDN, Screening=Network, Presentation=Allowed),
   Called Number=0203(TON=Unknown, NPI=ISDN))
*May 15 23:50:00.704: //21/E19AE4A2800F/CCAPI/cc_process_call_setup_ind:
   Event=0x4854EBA8
*May 15 23:50:00.708: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 0203
*May 15 23:50:00.708: //21/E19AE4A2800F/CCAPI/ccCallSetContext:
   Context=0x46B1DED4
*May 15 23:50:00.712: //21/E19AE4A2800F/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 21 with tag 1 to app "_ManagedAppProcess_Default"
*May 15 23:50:00.716: //21/E19AE4A2800F/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
*May 15 23:50:00.720: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x8056
        Channel ID i = 0xA98394
                Exclusive, Channel 20
*May 15 23:50:00.720: //21/E19AE4A2800F/CCAPI/ccCallDisconnect:
   Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*May 15 23:50:00.720: //21/E19AE4A2800F/CCAPI/ccCallDisconnect:
   Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
*May 15 23:50:00.724: //21/E19AE4A2800F/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
*May 15 23:50:00.728: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  callref = 0x8056
        Cause i = 0x8081 - Unallocated/unassigned number
*May 15 23:50:00.876: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref = 0x0056
        Cause i = 0x8081 - Unallocated/unassigned number
*May 15 23:50:00.880: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x8056
*May 15 23:50:00.888: //21/E19AE4A2800F/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x48595570, Tag=0x0, Call Id=21,
   Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
*May 15 23:50:00.888: //21/E19AE4A2800F/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*May 15 23:50:00.888: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*May 15 23:50:00.888: :cc_free_feature_vsa freeing 497EA530
*May 15 23:50:00.888: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*May 15 23:50:00.888:  vsacount in free is 0

Tanslation Rule definitely works:

test voice translation-rule 1 0203
Matched with rule 2
Original number: 0203   Translated number: 2803
Original number type: none      Translated number type: none
Original number plan: none      Translated number plan: none

Config:

voice translation-rule 1
rule 1 /0200/ /9201/
rule 2 /^02/ /28/
!
voice translation-rule 2
rule 1 /^662802/ /28/
!
voice translation-rule 66
rule 1 /^\([^0].*\)/ /00\1/
rule 2 /^\(0[^0].*\)/ /0\1/
!
!
voice translation-profile IncomingPOTS
translate calling 66
translate called 1
!
voice translation-profile SRST
translate calling 66
translate called 1
!
voice translation-profile SRST-VM
translate calling 2
!
voice translation-profile calling
translate calling 66
translate called 1

!
!

dial-peer voice 1 pots
voice cut-through alert
description Inbound DID via PSTN
translation-profile incoming IncomingPOTS
incoming called-number ^02..
direct-inward-dial
port 0/0/0:15
forward-digits all
!
dial-peer voice 3 pots
description Outbound to PSTN
destination-pattern 0T
port 0/0/0:15
!
dial-peer voice 10 voip
description Inbound CM-Subscriber Peer
destination-pattern ^02..
voice-class codec 1
voice-class h323 1
max-redirects 10
session target ipv4:10.1.201.11
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 11 voip
description Inbound CM-Publisher Peer
preference 2
destination-pattern ^02..
voice-class codec 1
voice-class h323 1
max-redirects 10
session target ipv4:10.1.201.10
dtmf-relay h245-alphanumeric
no vad

Just wanted to add that VG has registered to CM successfully and handsets. i can actullay make outgoing calls (from IP communicator) and internel calls with no issue.

I'm an idiot, just realised i had the wrong destination on my voip dail-peers

[jjcollaguazo]

Now its seems i'm getting one way voice on the E1. Not sure if its because i'm testing via an IP Communicator resgistered to

Mumbai. voice from IP Communicator to my desk phone is ok but vice versa doesn't work.

Any ideas? The E1 was working fine with CME. Hopefully someone should be in the Mumbai office in a an hour or so to confrim if E1 calls work or not.

interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
no cdp enable
!

voice-port 0/0/0:15
cptone IN
bearer-cap Speech
!

[jjcollaguazo]

The E1 issue was my IP Communicator being connected via PPTP. Phew!!!!

Adam you've been great help, hope you can help again. Now i seem to have intersite call issues. Mumbai users can hear Sydney fine but Sydney can't hear Mumbai users very well. Alot of distoration and they can only make out some words. Customer has confirmed with the carrier that QoS has been configured but i've asked them to confirm this is active because I've checked my regions in Cm and all looks good. I've also gotton users to confirm that the call is receiveing and transmitting in G729, but as suspected there alot of jitter being seenin the stats.

Here's the relvant config, my conferening and trancoding resources have been configured to use the the Sydney VG because there aren't enough DSP resources locally.

voice call disc-pi-off
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol cisco
h323
  call preserve limit-media-detection
modem passthrough nse codec g711ulaw
!
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711alaw
codec preference 3 g711ulaw
!
!
!
!
voice class h323 1
  h225 timeout tcp establish 3
  call preserve limit-media-detection

Please let me know if you need any other information.

Cheers,

James

[ADAM CRISP]

Yeah, I would say this is probably a little beyond the scope of a forum post, because there's too little information and its a little too close to working for you. I suggest you you sit back and draw out how everything works. Draw all the endpoints all the codec regions, all the trunks and what region they sit in, and confirm you understand how your media flow works in both directions. You can then find out if you have any bottle necks and look to see if you can make things work in a better way. Buy some PVDM's if you need some. If you're running  a global voice network I'm sure you can find somebody with some money.

good luck,

Adam

[paolo bevilacqua]

adamcrisp wrote:
Yeah, I would say this is probably a little beyond the scope of a forum post, because there's too little information and its a little too close to working for you. I suggest you you sit back and draw out how everything works. Draw all the endpoints all the codec regions, all the trunks and what region they sit in, and confirm you understand how your media flow works in both directions. You can then find out if you have any bottle necks and look to see if you can make things work in a better way. Buy some PVDM's if you need some. If you're running  a global voice network I'm sure you can find somebody with some money.
good luck,
Adam

There are pearls of wisdom in the post above, so I've rated it.

[jjcollaguazo]

hahahaha, fair call Adam. Thanks heaps for you help, its been a steep learning curve in regards to Voice and thankfully there's helpful people like you on this forum.