H323 Redundancy
Hello,
I have on my network 2 different h323 (non cisco) gateway in two different route groups, also I have a route list with the two route groups, when I make calls that point to that route list the callmanager does not search in the second route group if the first one returns busy, the only way to make the call go to the second gateway is disconnect the first one. From the cco I have read than if i put the Stop Routing on User Busy Flag -> False in service parameters, it should resolve the problem, but did not. This parameter is a inter-cluster-trunk parameter.
Can any one provide me some help on this issue.
Thanks
Hello,
The first gateway is returning release_comp with a busy cause.
I will provide the ccm detail trace for h225
Similar Messages
-
H323 redundancy between routers
I have 2 2800 series routers with E1 circuits on both, i can place calls on both gateways without any issues. If i pull the E1 circuit on one router, but still have IP connectivity to it from the callmanager, the route list route group will not forward outing calls to the second gateway. I'm wondering if this level of redundancy is going to work with H323. By the way if i loose ip conectivity with the first gateway the route list route group does force the outbond calls to the next H323 gateway and all works fine. Any help would be good.
dave whiteHi, why don't you use MGCP instead H.323? it's the best way to have redundancy.
http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a00801ad22f.shtml -
MGCP and H323 redundancy calling issue......
I have call Manager 7.1 and there is 2 MGCP gateways registered on the CUCM. Each gateway has 1 PRI line and this setup is working fine. Now I am adding new PRI line for redundancy prospect. So I had added the new E1 card for each gateway and then I have created the H323 trunk between the Voice gateway and CUCM. I have configured the Route Group and Route list for MGCP and H323.If primary MGCP is down call auto routed to H323.
Now when MGCP is down, call is auto routed to H323 and its hitting on the proper PRI port but call is not getting established and incoming is working fine.
Kindly find the isdn debug for your reference:
DEL-2921-ROUTER(config)#
DEL-2921-ROUTER(config)# debug isdn q931
Jan 31 16:52:34.655: ISDN Se0/0/0:15 Q931: Ux_DLRelInd: DL_REL_IND received from L2
Jan 31 16:52:44.655: ISDN Se0/0/0:15 Q931: Ux_DLRelInd: DL_REL_IND received from L2
Jan 31 16:52:47.267: ISDN Se0/2/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x0, Calling num 6272
Jan 31 16:52:47.267: ISDN Se0/2/0:15 Q931: Sending SETUP callref = 0x00AC callID = 0x802D switch = primary-net5 interface = User
Jan 31 16:52:47.267: ISDN Se0/2/0:15 Q931: TX -> SETUP pd = 8 callref = 0x00AC
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x0081, '6272'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '09821444335'
Plan:Unknown, Type:Unknown
Jan 31 16:52:47.295: ISDN Se0/2/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x80AC
Cause i = 0x82D2 - Identified channel does not exist
Jan 31 16:52:54.675: ISDN Se0/0/0:15 Q931: Ux_DLRelInd: DL_REL_IND received from L2
DEL-2921-ROUTER(config)#
THANKS IN ADVANCE.....Hi Rupesh,
The cause code "Idenfied channel does not exist" means:- This code indicates a call attempted on a channel that is not configured on the far end. This could happen if you are using a fractional PRI
Please ask to remote end for the number of channels configured and you can configure that number of channels accordingly at your end.
In CUCM 7.1 there is a service parameter which will help you to use the number of channel as per your requirement and rest of the channels you can mark it as busy so that CUCM won't select that channel.
Service Parameters > Call Manager > Advanced > CTRL-F > "maintenance"
In that you will find "Change B-Channel Maintenance Status" and mark channel as 1 which you don't want you to use.
For further information regarding this parameter you can click on that parameter and you will get more information.
And to enable above mentioned parameter, go to MGCP Gateway configuration page and check the box "Enable Status Poll"
Regards,
Nishant Savalia -
H323 Gateway Redundancy In CCM
HI
I would like to configure 2 h323 gateway on the cisco call manager,Each gateway's have got 2 E1's.
I would like to know is it possible to configure redundancy on the ccm ? if we can how the ccm will come to know the first gateway is down or fully utilized..?
Please advise me...
Thanks.
Nazeeryes, it's possible, this is from a similar post gogasca replied:
The default behavior in IOS, when there are no valid POTS dial-peers for a call to go out of, is to return âunallocated numberâ as a cause code to CCM. This can happen even if a t1 goes down. When the T1 goes down, the POTS peer is marked as down, and when CCM sends a call to the gateway, it'll return a UAN cause code to the ccm, causing CCM (by default) to stop hunting for other available gateways. There are ways to change the behavior using service parameters in CCM, but this behavior didn't make sense to me, so we looked for other ways to do it.
It turns out that you can issue the global command "no dial-peer outbound status-check pots" on the IOS GW, if you're opposed to changing the CCM behavior. What this command will do is cause the dial-peer to stay up. IOS will try and route the call, and when the T1 is down, it returns "No circuit available" to CCM. When CCM receives this cause code, it knows there's been a non-user error, and continues hunting, achieving the desired behavior.
In your case since it is H323 it will be retruning only H225 release comp message with proper code to continue re-routing.
The IOS code will translate any code we get from pots side to IP side
also, if possible, please try searching within netpro first as it's very probable someone already asked the same before
HTH
javalenc
if this helps, please rate -
Hello All,
I have an issue with fax, it is connected to an fxs port on my h323 gateway. When a call comes from PRI and get connected to fax after I hear the fax tone the fax drops. IOS is Version 15.0(1)M4, and DSP is PVDM3.
My analysis:
The external calls hit gateway and it matches the pots dialpeer where my fax connected to. From the "debug voip vtsp all", I see Primary Fax Protocol=CISCO_FAX_RELAY, which is not enabled and we are using T38 relay and modem passthrough method for faxing. I believe PVDM3 does not work with Cisco Relay and couldn't allocate any dsp resources for the call and it drops. But still I wonder how gateway takes cisco relay.
Config:
voice service voip
allow-connections h323 to h323
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
h323
emptycapability
modem passthrough nse codec g711ulaw
dial-peer voice 1001 pots
tone ringback alert-no-PI
description PSTN Incoming, Pattern:.
translation-profile incoming PSTN-to-PhoneDN
incoming called-number .
direct-inward-dial
dial-peer voice 999030 pots
tone ringback alert-no-PI
destination-pattern 3027898
progress_ind setup enable 3
port 0/3/0
Output of vtsp debug
Oct 4 20:30:07: //6824/97F0D7BD8350/VTSP:(0/1/0:15):7:1:1/vtsp_dsm_save_fax_config:
Fax Relay=ENABLED
Primary Fax Protocol=CISCO_FAX_RELAY, Fallback Fax Protocol=NONE_FAX_RELAY
Fax Relay CM Suppression :=ENABLED, Fax Relay ANS Suppression :=DISABLED
Fax Parameters Set By=Dialpeer, Peer=999030 (this is the peer where my fax connected to)
I have converted the port to MGCP and found its working, while faxing I see vtsp outputs
Oct 4 23:57:29: //7181/8FE17FEF8379/VTSP:(0/3/0):-1:1:2/vtsp_dsm_save_fax_config:
Fax Relay=DISABLED - MGCP Application
Primary Fax Protocol=IGNORE_FAX_RELAY, Fallback Fax Protocol=IGNORE_FAX_RELAY
Fax Relay CM Suppression :=ENABLED, Fax Relay ANS Suppression :=DISABLED
Fax Parameters Set By=MGCP Call Type --> I believe MGCP disabled the cisco fax relay and switch over to T38.
Oct 4 23:57:40: //7181/8FE17FEF8379/VTSP:(0/3/0):-1:1:2/vtsp_dsm_save_fax_config:
Fax Relay=ENABLED
Primary Fax Protocol=T38_FAX_RELAY, Fallback Fax Protocol=NONE_FAX_RELAY
Fax Relay CM Suppression :=ENABLED, Fax Relay ANS Suppression :=DISABLED
Fax Parameters Set By=MGCP Call Type
Can some one kindly suggest how to make it work on H323, also how we can force disabling cisco fax relay?As I mentioned earlier, we are using MGCP for that particular fax port and the gateway we use as H323.
Even I'm not pretty sure about using mixed protocol, I believe this gateway gonna used to support call center calls. SInce MGCP is easy for configs they might have confiugred only the port as MGCP endpoint.
Now its not working with both the cases, not in mgcp as well as fallback mode. Can you reveiw the debugs and see why I have to remove "modem passthrough" command from voice service voip section which of our standard config and works well with all other gateway of ours. Please help here. -
VG2921 - H323 - VG224 - analoge Modem works not properly
Hello Community, I try to setup a modem to work properly. First I try it w/ SCCP but fails, so I change to H323 but it is also not successful.
Modem answer to an incoming call, but they not negotiate and handshake fails and hang up.
Setup is following.
Calling Modem Called Party 93276 -- PSTN -- BRI -- VG2921 -- H323 -- VG224 -- Port 2/4 93276 Connected Modem.
The Modem at 93276 answer you hear typical whistler tone fax, but still only noise and after time the modem hans up.
I think they are not negotiate the right baud rate.
Trace and Config are attached.
Any help are appreciate,
Thanks in ahead
Armin
HTH, please rate all useful posts!
Config VG2921:
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
no supplementary-service h225-notify cid-update
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
no call service stop
modem passthrough nse codec g711ulaw
sip
no call service stop
voice translation-rule 1
rule 1 /20938/ /93200/
rule 2 /9800261/ /93276/
voice translation-profile AMT_IN
translate called 1
interface GigabitEthernet0/1.140
encapsulation dot1Q 140
ip address 172.xxx.xxx.xxx 255.xxx.xxx.xxx
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.30.xxx.xxx
interface BRI0/0/0
description Anschluss Amt
no ip address
isdn switch-type basic-net3
isdn overlap-receiving T302 3000
isdn tei-negotiation first-call
isdn tei-negotiation preserve
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
voice-port 0/0/0
translation-profile incoming AMT_IN
translation-profile outgoing AMT_OUT
no vad
compand-type u-law
cptone DE
timeouts interdigit 4
bearer-cap 3100Hz
dial-peer voice 10 pots
description Amt inbound
incoming called-number .
direct-inward-dial
dial-peer voice 202 voip
description VG224
preference 1
destination-pattern 93276
modem passthrough nse codec g711ulaw
session target ipv4:172.xxx.xxx.xxx
codec g711ulaw
Config VG224:
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
h323
modem passthrough nse codec g711ulaw
interface FastEthernet0/0
description to VG2921 GE02
ip address 172.xxx.xxx.xxx 255.xxx.xxx.xxx
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.xxx.xxx.xxx
voice-port 2/4
cptone DE
timeouts initial 60
timeouts interdigit 60
timing hookflash-in 250 50
description Fernstromzaehler, 93276
bearer-cap 3100Hz
dial-peer voice 30 pots
description Fernstromzaehler, 93276
answer-address 93276
destination-pattern 93276
incoming called-number 93276
port 2/4
dial-peer voice 100 voip
description CUCM inbound
modem passthrough nse codec g711ulaw
incoming called-number 93276
codec g711ulawHere you see it runs into modem passthrough but always not successful, hangs up after time.
Telephony call-legs: 1
SIP call-legs: 0
H323 call-legs: 1
Call agent controlled call-legs: 0
Total call-legs: 2
120E : 107 192845570ms.106 (*16:47:44.768 CET Thu Nov 28 2013) +22550 +29970 pid:202 Originate 93276
dur 00:00:07 tx:346/54733 rx:597/93550 10 (normal call clearing (16)) dscp:0 media:0 audio tos:0x0 video tos:0x0
IP 172.30.13.186:16404 SRTP: off rtt:1ms pl:4110/0ms lost:0/0/0 delay:55/55/55ms g711ulaw TextRelay: off Transcoded No
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long dur callduration :n/a timestamp:n/a
MODEMPASS nse buf:0/0 loss 0% 0/0 last 0s dur:0/0s
120E : 106 192845570ms.107 (*16:47:44.768 CET Thu Nov 28 2013) +22550 +30170 pid:10 Answer 6157915399
dur 00:00:07 tx:597/98326 rx:346/54733 10 (normal call clearing (16)) dscp:0 media:0 audio tos:0x0 video tos:0x0
Telephony 0/0/0 (106) [0/0/0.1] tx:4530/4530/0ms g711ulaw noise:-84dBm acom:3dBm
long duration call detected:n long dur callduration :n/a timestamp:n/a
HTH, please rate all useful posts! -
DTMF tones from CUCUM 9 thru H323 GW out SIP trunk not working
This is the setup. Currently in lab environment for a client, but needs to go into production
IP Phone -> CUCM 9 -> H323 GW -> SIP Trunk -> Proprietary device -> Analog phone
Calls complete both ways with no issues. Proprietary devices only uses G711ulaw, so I have configured a xcoder on the H323 GW to transcode to G729 across the WAN link (between the CUCM cluster and the H323 GW).
Pressing keys/sending DTMF tones from the IP phone are not heard in the analog phone
Running a debug voice ccpai inout at the H323 gateway shows me that the DTMF tones are being received the GW and are being sent along. See below:
Seaport#
Seaport#
Seaport#! Pressing digit "9" on VoIP phone
Seaport#
Seaport#
Seaport#
Seaport#
*Nov 5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E
*Nov 5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
*Nov 5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E
*Nov 5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
Seaport#
Seaport#! Pressing digit "9" on VoIP phone " on VoIP phone 5" on VoIP phone
Seaport#
Seaport#
Seaport#
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
Seaport#
Seaport#! Pressing digit " 5" on VoIP phone
Seaport#
Seaport#
Seaport#
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov 5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
Seaport#
Seaport#
However, debug ccsip does not give me any indications that the DTMF tone is being sent out the SIP trunk. Debug ccsip all attached.
Relevant portions of the H323 configuration are below
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g729br8
interface Loopback0
ip address 172.16.88.254 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.88.254
interface GigabitEthernet0/1
ip address 192.168.200.254 255.255.255.0
duplex auto
speed auto
interface Loopback0
ip address 172.16.88.254 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.88.254
interface GigabitEthernet0/1 <- interface to proprietary device
ip address 192.168.200.254 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/2 <-interface to Local LAN supporting IP Phones
ip address 10.10.10.254 255.255.255.0
duplex auto
speed auto
sccp local GigabitEthernet0/2
sccp ccm 10.10.10.254 identifier 1 priority 1 version 3.1
sccp ccm group 1
bind interface GigabitEthernet0/2
associate ccm 1 priority 1
associate profile 10 register xcoder_1
dspfarm profile 10 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 10
associate application SCCP
dial-peer voice 2 voip
description Default Incoming Dial Peer
incoming called-number .
voice-class codec 1
dtmf-relay h245-alphanumeric h245-signal rtp-nte
dial-peer voice 6 voip
destination-pattern 90052.. <- DN of analog phone
session protocol sipv2
session target ipv4:192.168.200.1 <- IP of proprietary device
codec g711ulaw
no vad
sip-ua
registrar ipv4:172.16.88.254 expires 3600
no transport tcp
telephony-service
sdspfarm units 4
sdspfarm transcode sessions 2
sdspfarm tag 1 xcoder_1
I also ran the debug voip rtp session named-event all but nothing was displayed when I pressed the digits on the IP Phone.
JeffPlease configure "dtmf-relay rtp-nte" command under SIP dial-peers.
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers. -
How to configure modem connection with GW (H323) and ATA 187
Hello Community,
i stock in configuration and need assistance.
My callflow: Telco – PRI – GW – H323– CUCM – SIP – ATA187 – Modem
Voicegateway (Version 15.3(2)T) + CUCM (Version= 8.6) + ATA187 (Version= 9.2.3.1)
The modem connection is still not working.
What is still to configure on the voicegateway? modem passthrough?
Regards Michael
ATA 187 Configuration:
Fax Mode= T.38 Fax Relay
Fax Error Correction Mode Override= Off
Maximum Fax Rate= 14000bps
Impedance= 900Ohms complex
Gateway Configuration:
voice service voip
ip address trusted list
ipv4 172.30.50.1
ipv4 172.30.50.2
ipv4 172.30.50.3
ipv4 172.30.50.4
ipv4 172.30.50.5
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h225-notify cid-update
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 600 min 60
dial-peer voice 1 pots
translation-profile incoming INCOMING_PSTN
incoming called-number .
direct-inward-dial
dial-peer voice 30 voip
description OUTGOING_CUCM
destination-pattern [1-9]..
session target ipv4:172.20.60.12
voice-class codec 1
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000
no vadIt is possible T38 isn't playing well with the PRI. You could try modem pass-through on the gateway and ATA187 if T38 isn't necessary.
Also, sometimes these commands are needed, but not always, so I would consider whether these fax commands under the dial-peer are necessary:
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000 -
No Redundancy for Inbound Calls
Dear all,
We have 2 CallManagers at our organization. Incoming calls are routed to Attendant Console to listening Music on Hold. Whenever the Publisher CallManager is down, calls coming into our organization receive a busy signal. I thought we were setup for redundancy, but I found out the hard way that this was not the case.
We have a 2911 router which routes incoming phone traffic into our organization. On it are two dial peers which route our organization?s incoming calls to the CallManagers. The primary dial peer is supposed to route to the publishe and the secondary dial peer to the subscriber. Below are the 2 dial peers which route our traffic.
voice class h323 1
h225 timeout tcp establish 3
dial-peer voice 11 voip
preference 2
destination-pattern 3.......
session target ipv4:10.1.11.30 ( publiser)
dtmf-relay h245-alphanumeric
codec g711alaw
dial-peer voice 10 voip
preference 1
destination-pattern 3.......
session target ipv4:10.1.11.31 ( subcriber)
dtmf-relay h245-alphanumeric
codec g711alaw
If i delete dial peer voice 11, incoming call can route to subcriber successfully
I would appreciate anyone taking the time to look at them and offer any suggestions.
CCM 10.0
I got debug for this as below:
TSI-DC-VGR01-0#end
Translating "end"
% Bad IP address or host name
Translating "end"
% Unknown command or computer name, or unable to find computer address
TSI-DC-VGR01-0#
Jul 30 07:48:09.066: //-1/AEEEB6D6813B/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=0905916072
cisco-anitype=2
cisco-aniplan=1
cisco-anipi=0
cisco-anisi=3
dest=38236688
cisco-desttype=0
cisco-destplan=1
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
Jul 30 07:48:09.066: //-1/AEEEB6D6813B/CCAPI/cc_api_call_setup_ind_common: Interface=0x2BA17414, Call Info( Calling Number=0905916072,(Calling Name=)(TON=National, NPI=ISDN, Screenin
etwork, Presentation=Allowed),
Called Number=38236688(TON=Unknown, NPI=ISDN),
Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinatioag=TRUE,
Incoming Dial-peer=9, Progress Indication=NULL(0), Calling IE Present=TRUE
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=F
E), Call Id=-1
Jul 30 07:48:09.066: //-1/AEEEB6D6813B/CCAPI/ccCheckClipClir:
In: Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Network, P
entation=Allowed)
Jul 30 07:48:09.070: //-1/AEEEB6D6813B/CCAPI/ccCheckClipClir:
Out: Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Network,
sentation=Allowed)
Jul 30 07:48:09.070: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.070: :cc_get_feature_vsa malloc success
Jul 30 07:48:09.070: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.070: cc_get_feature_vsa count is 1
Jul 30 07:48:09.070: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.070: :FEATURE_VSA attributes are: feature_name:0,feature_time
0612504,feature_id:1877
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Netw
, Presentation=Allowed),
Called Number=38236688(TON=Unknown, NPI=ISDN))
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/cc_process_call_setup_ind:
Event=0x2B9DF948
Jul 30 07:48:09.070: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 38236688
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCallSetContext:
Context=0x300C4B5C
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 1877 with tag 9 to app "_ManagedAppProcess_Default"
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=10, Params=0x300C606C, Progress Indication=NULL(0)
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCheckClipClir:
In: Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Network, P
entation=Allowed)
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCheckClipClir:
Out: Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Network,
sentation=Allowed)
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCallSetupRequest:
Destination Pattern=3......., Called Number=38236688, Digit Strip=FALSE
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccCallSetupRequest:
Calling Number=0905916072(TON=National, NPI=ISDN, Screening=Network, Prese
tion=Allowed),
Called Number=38236688(TON=Unknown, NPI=ISDN),
Redirect Number=, Display Info=
Account Number=, Final Destination Flag=TRUE,
Guid=AEEEB6D6-16F4-11E4-813B-A4934C055B80, Outgoing Dial-peer=10
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=0905916072
cisco-anitype=2
cisco-aniplan=1
cisco-anipi=0
cisco-anisi=3
dest=38236688
cisco-desttype=0
cisco-destplan=1
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
Jul 30 07:48:09.070: //1877/AEEEB6D6813B/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x312C8048, Interface Type=1, Destination=, Mode=0x0,
Call Params(Calling Number=0905916072,(Calling Name=)(TON=National, NPI=IS
Screening=Network, Presentation=Allowed),
Called Number=38236688(TON=Unknown, NPI=ISDN), Calling Translated=FALSE,
Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Outgoing Dial-
r=10, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, A
ication Call Id=)
Jul 30 07:48:09.070: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.070: :cc_get_feature_vsa malloc success
Jul 30 07:48:09.074: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.074: cc_get_feature_vsa count is 2
Jul 30 07:48:09.074: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 30 07:48:09.074: :FEATURE_VSA attributes are: feature_name:0,feature_time
0612952,feature_id:1878
Jul 30 07:48:09.074: //1878/AEEEB6D6813B/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=1, FlowMode=1
Jul 30 07:48:09.074: //1878/AEEEB6D6813B/CCAPI/ccCallSetContext:
Context=0x300C601C
Jul 30 07:48:09.074: //1877/AEEEB6D6813B/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=10
Jul 30 07:48:19.386: //1877/AEEEB6D6813B/CCAPI/cc_api_call_disconnected:
Cause Value=102, Interface=0x2BA17414, Call Id=1877
Jul 30 07:48:19.386: //1877/AEEEB6D6813B/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=FALSE, Cause Value=102, Retry Count=0)
Jul 30 07:48:19.386: //1878/AEEEB6D6813B/CCAPI/ccCallDisconnect:
Cause Value=102, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconne
Cause=0)
Jul 30 07:48:19.386: //1878/AEEEB6D6813B/CCAPI/ccCallDisconnect:
Cause Value=102, Call Entry(Responsed=FALSE, Cause Value=102)
Jul 30 07:48:19.386: //1878/AEEEB6D6813B/CCAPI/cc_api_get_transfer_info:
Transfer Number Is Null
Jul 30 07:48:19.386: //1877/AEEEB6D6813B/CCAPI/ccCallDisconnect:
Cause Value=102, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconne
Cause=102)
Jul 30 07:48:19.386: //1877/AEEEB6D6813B/CCAPI/ccCallDisconnect:
Cause Value=102, Call Entry(Responsed=TRUE, Cause Value=102)
Jul 30 07:48:19.386: //1877/AEEEB6D6813B/CCAPI/cc_api_get_transfer_info:
Transfer Number Is Null
Jul 30 07:48:19.386: //1878/AEEEB6D6813B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x312C8048, Tag=0x0, Call Id=1878,
Call Entry(Disconnect Cause=102, Voice Class Cause Code=0, Retry Count=0)
Jul 30 07:48:19.386: //1878/AEEEB6D6813B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
Jul 30 07:48:19.386: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Jul 30 07:48:19.386: :cc_free_feature_vsa freeing 2C24DB50
Jul 30 07:48:19.386: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Jul 30 07:48:19.386: vsacount in free is 1
Jul 30 07:48:19.394: //1877/AEEEB6D6813B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x2BA17414, Tag=0x0, Call Id=1877,
Call Entry(Disconnect Cause=102, Voice Class Cause Code=0, Retry Count=0)
Jul 30 07:48:19.398: //1877/AEEEB6D6813B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
Jul 30 07:48:19.398: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Jul 30 07:48:19.398: :cc_free_feature_vsa freeing 2C24D990
Jul 30 07:48:19.398: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:Dear Okanlawon & islam.kamal,
Both of you are correct. I used your command and it worked now. It also help me solved the problem related to Music On Hold cause i use g711ulaw ( MoH wont work with incoming call).
I used c2900-universalk9-mz.SPA.151-4.M4.bin
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.1(4)M4,
EASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2012 by Cisco Systems, Inc.
Compiled Tue 20-Mar-12 18:57 by prod_rel_team
ROM: System Bootstrap, Version 15.0(1r)M15, RELEASE SOFTWARE (fc1)
Thank you very much ! You are the god !
ThanhNT -
AutoAttendant Redundancy for Unity in SRST
I'm trying to acheive redundancy for AA configured in Unity in a Centralised UCM 7.x environment for a remote site in SRST using TLC script, the solution works fine in MGCP environment but I'm trying to acheive it utilizing H323 as the link is CAS R2 and not PRI, the problem is that the aa number does not hunt for the VoIP dial-peer and always hunts for the pots dial-peer because of the "incoming called number" and "service aa" under the pots, is there away that we can acheive this?
Thank you for your reply, but I already tried this and it still hunts to the pots dial-peer a snapshot of the config is as follows :
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 10 voip
preference 0
incoming called-number [Unity AA Number]
destination-pattern [Unity AA Number]
session target ipv4:[UCM IP Address]
codec g711ulaw
no vad
ip qos dscp cs3 signalling
dial-peer voice 100 pots
preference 1
incoming called-number [TLC AA Number]
service autoatt
destination-pattern [TLC AA Number]
port 1/0/0:15
where [Unity AA Number] = [TLC AA Number]
and still the pots gets invoked prior to the voip if I shut down the pots or take incoming called-number for it it goes to unity -
As5350 VOIP h323 g723 codec problem
I setup the VOIP network with H323. All calls are Gatekeeper routed.
When I make a PSTN call either inbound or outbound, as5350 could not
establish a call other than g729 codec.
Question 1. Why AS5350 uses g729 codec as default ? How you setup a
g729 default codec?
Question 2. Why AS5350 cannot use other codec like g723? AS5350
release a call right after setup because it can't talk with g723
codec. Gateway was setup only g723 codec to see if AS5350 can talk.
When I say PSTN call, I mean the call initiated from our h323 gateway
to PSTN number.
Gateway --> AS5350 PRI --> PSTN number (outbound call)
PSTN number --> AS5350 PRI --> H323 gateway (inbound call)
Codec is established in the Internet portion between gateway and
AS5350.
Gateway is non cisco equipment and setup as g729 and g723
capabilities.
AS5350 Configuration only necessary
portion----------------------------------------
voice call send-alert
voice rtp send-recv
voice service pots
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0
h323
voice class codec 1
codec preference 1 g729r8
codec preference 2 g723r63
codec preference 3 g723r53
voice class codec 3
codec preference 1 g723r63
voice class h323 1
call start fast
controller T1 2/0
framing esf
linecode b8zs
pri-group timeslots 1-24
interface Serial2/0:23
no ip address
isdn switch-type primary-ni
isdn incoming-voice modem
isdn T306 30000
isdn T310 60000
isdn send-alerting
isdn sending-complete
no cdp enable
voice-port 2/0:D
echo-cancel coverage 128
no vad
bearer-cap Speech
dial-peer voice 1 pots
preference 1
destination-pattern 1.T
progress_ind setup enable 3
no digit-strip
port 2/0:D
forward-digits all
dial-peer voice 2132332745 voip
destination-pattern 2132332745
progress_ind setup enable 3
voice-class codec 1
voice-class h323 1
session target ras
interface FastEthernet0/0
ip address x.x.x.x 255.255.255.0
no ip proxy-arp
duplex auto
speed auto
no cdp enable
h323-gateway voip interface
h323-gateway voip id AAAA ipaddr x.x.x.x 1719
h323-gateway voip h323-id as5350What solution did you use, if you don't mind me asking?
As far as CUBE goes, I think I have the answer. It can be done only using 2 transcoders.
CUBE can't add or remove transcoder mid-call, and while CUCM can add a transcoder during transfers it seems it can't remove a transcoder mid-call. So in our case during the transfer the xcoder just signals media error and drops the call.
The solution is to use one transcoder on CUBE (CME registered) - this one will always xcode the incoming call from g729 to g711, then there needs to be another xcoder on CUCM - CUCM will use it during transfer to transcode between the CUBE call leg (g711) and phone C call leg (g729).
BTW, I need this functionality in UCCE environment with Outbound Dialer and VoIP connection to PSTN (g729 only). -
(config-subif)#h323-gateway voip bind srcaddr xxx.xxx.xxx.xxx
Bind IP Address configured. Please remove before reconfiguring
I am getting this message despite no bind address configured. anyone seen this issues before?
RTR#sh run | include bind
RTR#
Any advise will be much appreciated.version 15.0
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime localtime show-timezone
service password-encryption
service compress-config
hostname RTR
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.150-1.M4.bin
boot-end-marker
card type e1 0 1
logging buffered 4096
aaa new-model
aaa session-id common
network-clock-participate wic 1
no ipv6 cef
no ip source-route
ip cef
ip vrf 3RD_PARTY
rd
route-target export
route-target import
ip vrf DATA
rd
route-target export
route-target import
ip vrf forwarding
no ip dhcp use vrf connected
no ip dhcp conflict logging
ip dhcp excluded-address 172.27.165.1 172.27.165.25
ip dhcp excluded-address 172.27.165.129
ip dhcp pool VLAN1
network 172.27.165.0 255.255.255.192
dns-server xxx.xxx.xxx.xxx
default-router 172.27.165.1
option 150 ip XXX.XXX.XXX
lease 3
ip dhcp pool VLAN133
network 172.27.165.128 255.255.255.192
default-router 172.27.165.129
lease 0 3
no ip domain lookup
multilink bundle-name authenticated
isdn switch-type primary-net5
crypto pki token default removal timeout 0
voice-card 0
voice rtp send-recv
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711ulaw
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class h323 1
h225 timeout tcp establish 3
call start fast
voice translation-rule 100
rule 1 /^.*/ /xxxxxxxxx/
voice translation-profile MAPOUTSIDE
translate calling 100
hw-module pvdm 0/0
redundancy
controller E1 0/1/0
line-termination 75-ohm
pri-group timeslots 1-31
description ### PABX ###
controller E1 0/1/1
interface GigabitEthernet0/0
description ### Trunk ###
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.1
description ### Managment, Server and Voice network ###
encapsulation dot1Q 1 native
ip address 192.168.1.1 255.255.255.0 secondary
ip address 172.27.165.1 255.255.255.192
no ip redirects
interface GigabitEthernet0/0.2
description ### Lan ###
encapsulation dot1Q 800
ip vrf forwarding DATA
no ip redirects
ip tcp adjust-mss 1400
bridge-group 2
bridge-group 2 input-address-list 702
interface GigabitEthernet0/0.5
description ### Riverbed AUX Port ###
encapsulation dot1Q 5
no ip redirects
ip nat outside
ip virtual-reassembly
interface GigabitEthernet0/0.133
description ### Internet Wifi ###
encapsulation dot1Q 133
ip vrf forwarding 3RD_PARTY
no ip redirects
ip accounting output-packets
bridge-group 33
bridge-group 33 input-address-list 733
bridge-group 33 output-address-list 733
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
interface GigabitEthernet0/2
no ip address
duplex auto
speed auto
interface GigabitEthernet0/2.950
bandwidth 1024
encapsulation dot1Q 950
ip address 172.27.165.254 255.255.255.252
ip flow ingress
ip flow egress
interface GigabitEthernet0/2.951
bandwidth 768
encapsulation dot1Q 951
ip vrf forwarding DATA
ip address 172.27.165.250 255.255.255.252
ip flow ingress
ip flow egress
rate-limit output 768000 8000 8000 conform-action transmit exceed-action drop
interface GigabitEthernet0/2.952
bandwidth 256
encapsulation dot1Q 952
ip vrf forwarding 3RD_PARTY
ip address 172.27.165.246 255.255.255.252
ip flow ingress
ip flow egress
rate-limit output 256000 8000 8000 conform-action transmit exceed-action drop
interface GigabitEthernet0/2.959
encapsulation dot1Q 959 native
ip address 172.27.165.242 255.255.255.252
ip flow ingress
ip flow egress
interface Serial0/0/0
bandwidth 1024
no ip address
encapsulation frame-relay
load-interval 30
no keepalive
priority-group 1
ignore dcd
no clock rate 2000000
interface Serial0/0/1
no ip address
shutdown
clock rate 2000000
interface Serial0/1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn timer T310 120000
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
interface BVI2
ip vrf forwarding DATA
ip address 192.168.1.1 255.255.255.0 secondary
ip address 172.16.199.1 255.255.255.0
no ip redirects
ip accounting output-packets
ip nbar protocol-discovery
ip tcp adjust-mss 1400
interface BVI33
ip vrf forwarding 3RD_PARTY
ip address 172.27.165.129 255.255.255.192
ip accounting output-packets
ip forward-protocol nd
no ip http server
no ip http secure-server
ip flow-top-talkers
top 20
sort-by bytes
cache-timeout 30000
ip nat inside source list 150 interface GigabitEthernet0/0.5 overload
ip route 0.0.0.0 0.0.0.0 172.27.165.253 name default
ip route vrf 3RD_PARTY 0.0.0.0 0.0.0.0 172.27.165.245 name Default
ip route vrf DATA 0.0.0.0 0.0.0.0 172.27.165.249 name Default
control-plane
bridge 2 protocol ieee
bridge 2 route ip
bridge 33 protocol ieee
bridge 33 route ip
no call rsvp-sync
voice-port 0/1/0:15
voice-port 0/2/0
voice-port 0/2/1
voice-port 0/2/2
voice-port 0/2/3
dial-peer voice 100 pots
description ### PABX ###
destination-pattern XXXXXXXX
port 0/1/0:15
forward-digits all
dial-peer voice 2000 voip
preference 2
destination-pattern .T
progress_ind setup enable 3
session target ipv4:xxx.xxx.xxx.xxx
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
playout-delay nominal 130
playout-delay mode fixed
fax-relay ecm disable
fax rate 9600
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
clid network-number XXXXXXXX
no vad
dial-peer voice 2001 voip
preference 1
destination-pattern .T
progress_ind setup enable 3
session target ipv4:xxx.xxx.xxx.xxx
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
playout-delay nominal 130
playout-delay mode fixed
fax-relay ecm disable
fax rate 9600
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
clid network-number XXXXXXXX
no vad -
Hi,
I am using Windows 8.1
I have an External Hard Disk i have partitioned it to 4 parts.
One drive is now inaccessible due to sudden power failure while listening Music from that drive few days ago.
Now it shows "Data error (Cyclic redundancy check)".
I tried all the procedures provided here like
chkdsk /f, diskpart, rescan etc
but no result :( (i mean all processes failed. They could not detect the drive).
Please help me to get those data, pictures and project files.
thank youThen why aren't you posting this in the Windows 8 forums found @
http://social.technet.microsoft.com/Forums/windows/en-US/home?category=w8itpro
This is a Windows 7 forum for discussion about Windows 7.
Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if a marked post does not actually answer your question. This can be beneficial to other community members reading the thread. ” -
Data error (cyclic redundancy check) when installing windows xp..
hi guys.. i'm new here.. just switched to macdom a few days ago but unfortunately, i have been having problems trying to install windows xp with sp 2 on my system using boot camp. everytime i install it, i get to the setup screen ("39 minutes till setup rah rah rah", "windows xp is awesome because it has this cool interface etc... rah rah rah") and then the error of doom comes out -_-
the error given is;
an eror has been encountered that prevents setup from continuing
one of the components that windows needs to continue setup could not be installed
data error (cyclic redundancy check)
if you are installing from a cd, there might be a problem with the disc; try cleaning the disc or using another disc
if you are installing from the network, it is possible that not all of the files were copied correctly to your disk drive. run the disk checking utility on your installation drive from the recovery console and start setup again
press ok to view the setup log file
i have tried numerous times without fail and it is getting to my head.. gah.. if someone could help me out, it would be massive and i would sell my soul to you! (kidding).. thanks for reading!
p/s: my setup is;
Macbook
2.1ghz
1gb ram
120gb hard disk
dvd/cd-rw combo drive
the basic setup pretty much.. again.. any help would be greatly appreciated. thank you so much guys!I guess there is a problem with your XP CD, probably scratched or did not burn successfully. Have you tried it with another installation cd?
-
Recovery Window-Based Retention VS Redundancy-Based Retention
Hi Experts,
We'd like to know your take on the use of Recovery Window-Based Retention Policy e.g.
RMAN> CONFIGURE RETENTION POLICY TO RECOVERY WINDOW OF 7 DAYS;against the use of Redundancy-Based Retention Policy, e.g.
CONFIGURE RETENTION POLICY TO REDUNDANCY 7;Do you have any recommendations or preferences to which should be used? Is there a preferred method by oracle?
We're currently setting up RMAN for a client that's using Oracle 11.1.0.7 standard edition, so is there a preference to what's better suited for the standard edition? The plan is to back up data to Disk, and this data will be then backed up to tape.
ThanksREDUNDANCY 7 is 7 backups -- irrespective of the number of days.
If you are running only 1 backup a day, you'd assume that it is equivalent to 7 days. However, if one day you run a backup twice, then the 7-day old backup becomes redundant ! If, the next day, you again run the backup twice, the 5-day old backup becomes redundant ! (Conversely, if you don't run a backup for 2 days, then even the 9 day old backup is not redundant !).
So, be aware (or beware) that any adhoc backup runs or changes to the backup frequency would change your retention duration (and if this happens 6 months from now, the IT Manager / DBA onsite may not know that retention has changed !)
Hemant K Chitae
Maybe you are looking for
-
HP Laserjet 8000N not working with Maverick
Hello. I have been trying to set up a nework printer on a new MacBook Pro running Mavericks. Software updates have been run. We have tried to de-install and re-install the printer, an HP Laserjet 8000N. HP doesn't have an updated driver for Maveric
-
Difference between Monitoring Process and Integration Process
What is the difference between the Monitoring Process object for BAM and Integration Process object? Both seem to have exactly the same design environment in the Enterprise Services Builder. Is it so that Monitoring Process alarms only appear in the
-
How do I deauthorize my ibook G4 and wipe the hard drive clean?
How do I deauthorize my ibook G4 and wipe the hard drive clean?
-
Hi. On MIGO, there is a field called GOITEM-LEINDT (Delivery Date) which isn't populated. Any idea why? On MIGO i want to be able to see the delivery date which should come from the PO. Why isn't this shown? Thanks Adeel
-
German language changed into english after upgrade but wont re-change
the option in settings is still on german but the whole iTunes is english after i upgraded to the 7.4.3.1 version. whats that?