Why we dont' see H323 gateway and SIP trunk "registered" in CCM?

in CCM admin, we see the status of H323 gateways and SIP trunks have no "registered to xxxx" status.

Unlike devices with other protocols, there is no registration mechanism for these.

Similar Messages

Why we dont need sender agreement and comm channel in IDOC to file scenario

Hi,
there are 2 queries:
first query :
Scenario is "Idoc to file transfer"
why we dont need sender agreement and communication channel for IDOC to file ,while we do need
both of these for "File to RFC "
Answer : which i got is because IDOC sits at ABAP stack but same is valid for RFC also.
Then why For "IDOC to file" we dont need that.
Second query:
Also when i did scenario for File1-RFC-File2 scenario using BPM i have created a interface
between XI-RFC synchronous to send data which we got from file1 in XI to RFC importing parameter
and getting data from RFC back to File2.
Here in IDOC to file scenario why we dont need any interfface for IDOC-XI Asynchronous.
Regards,
Mayank
Edited by: mayank verdia on Jan 13, 2009 11:49 AM
Edited by: mayank verdia on Jan 13, 2009 11:50 AM

Dear All,
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Adding ICT trunk and SIP trunk into Route group

Hi ,
We need to map ICT and SIP trunk into the same route group ,but the problem here is already same ICT is mapped to another route pattern.
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Thanks Suresh ...
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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.
Jeff

Please configure "dtmf-relay rtp-nte" command under SIP dial-peers.
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers.

Cucm , Cube via Sip and Sip Trunk to ISP , Outgoing calls not working

Hi
We have issue with the outgoing calls to sip trunk
Below is the config and the debugs
It will be great if you give your thoughts since we have stuck here
My thoughts are:
i see that for unknown reason the called number is going with 4 digits instead of 8 digits
i dont see any sip message comming from ISP
Maybe the call not going there ? to isp trunk? From the trace the call hit the correct dialpeer 888 but i see 4 digits as a called number , but i dodnt understant the reason to translated in 4 digits the called number.Not apply a translation rule for that
confused!!!
Calling Numbner:22324086
Called Number: 23823690
CUCM:192.168.1.241 and 242
CUBE:192.168.1.10
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*$....$/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^$9$$.......$$/ /99\2/
rule 3 /^$2$$.......$$/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*$....$/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^$9$$.......$$/ /99\2/
rule 3 /^$2$$.......$$/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad

Hi Aok
I change the default value for IPVMS from g711ulaw to g711alaw but the results remained the same
Also i have restarted the IPVMS
SIP-GW#
SIP-GW#
*Mar 5 14:19:57.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24784 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:57.878: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1362493197
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 101
c=IN IP4 10.249.13.130
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.878: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:57.926: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 102 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 213
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 2 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=inactive
*Mar 5 14:19:57.942: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 101
c=IN IP4 192.168.1.10
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
*Mar 5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Length: 0
*Mar 5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1362493198
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 103 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 216
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 3 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 18 96 99
a=rtpmap:96 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
a=sendrecv
*Mar 5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 283
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 18 101
c=IN IP4 192.168.1.10
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 192
v=0
o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 192.168.1.241
t=0 0
m=audio 4000 RTP/AVP 8
a=X-cisco-media:umoh
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly
*Mar 5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 99
c=IN IP4 10.249.13.130
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
SIP-GW#
SIP-GW#sh voip rtp connections
VoIP RTP active connections :
No. CallId     dstCallId LocalRTP RmtRTP     LocalIP                                RemoteIP
1     716        717        19314    4000     192.168.1.10                           192.168.1.241
2     717        716        19234    54932    10.249.13.130                          10.224.42.72
Found 2 active RTP connections

Why do we need MTP in the SIP trunk for CVP warm transfers

Hi All,
Why do we need to enable MTP in SIP trunk between CUCM and CVP for CVP based trasnfers???
Thanks in advance!!
Regards,
Thammaya Gupta K.

I saw also in the CDR logs that the IP Phone media transport going to CUBE is in G711.And as well in the wireshark capture of the IP communicator that the CUCM invoke to use the g711 codec but as per ITSP logs they are now in the g729.
@ Jamie If I un-tick the MTP point required in SIP trunk will make the call leg from IP Phone to CUBE g729 (w/o hw resource), I have also tried to use g729 preferred originating codec, but still the IP Phone is using g711.
I have seen a documentation states:
" To configure G.729 codecs for use with a SIP trunk, you must use a hardware MTP or transcoder that supports the G.729 codec." - I read this on the CUCM help page under configuring SIP trunk setting.
Our ultimate goal is to use g729 without using HW MTP/ transcoder.
IP Phone ->CUCM SIP Trunk ->CUBE-> ITSP

BE6000 and SIP trunk

Hello!
Now we use Cisco IP telephony based on Cisco CME. Cisco CME is installed on 2821 router with IOS C2800NM-ADVENTERPRISEK9-M, Version 12.4(24)T7. A SIP trunk is configured between CME router and telephony provider.
We plan to upgrade our telephone system and begin using Cisco CBE6000. I study the documentation and in "Cisco Business Edition Unified Communications and Collaboration Solutions for Small and Midsized Companies", I find that for SIP trunk to works in CBE6000 a Cisco Integrated Services Router with Cisco Unified Border Element is required. Or other quote from this document "Public switched telephone network (PSTN) connections using SIP is available through any Cisco Integrated Services Router voice gateway".
In this regard there is a question: what is the functions of this router when I make a trunk between CBE6000 and telephony provider SIP gateway and whether it is possible to use existing Cisco 2821 router as such router.
Thanks.

Hi Ayodeji,
I too have a somewhat similar query.
We have UC560 running currently at our company, the UC560 is connected to an ISDN modem from the network provider. We are now upgrading to BE6000. To connect to the ISDN E1/T1 line, we've ordered Cisco 2921 with a E1/T1 VWIC card (VWIC3-2MFT-T1/E1) which will connect to BE6000 and internal LAN. I just wanted to know what would I need to configure on 2921?
Do I need to install CUBE or just configure it to connect to ISDN line?
Many thanks for any help.
regards,

+ Globalization and SIP Trunks

I'm trying to add the +E.164 number to Missed Call Field. On a SIP Trunk, I know I don't have support for ISDN TON (Subscriber, National, International), so when I push the Calling Party Number to a Transformation how do I ensure I tag Local, National, and International Calling Party Number correctly? Configuration examples for those who have done this would be appreciated.

Hello Sam,
After a lot of research on my personal Lab, I was able to achieve Globalization in SIP Trunks. This is how I did:
I created two partitions, CALLING-XFORM-DP-PT, and CALLING-XFORM-TRK-PT.
One is for the Device Pool, and the other one is for the SIP Trunk. Create the CSSs and add your partitions > CALLING-XFORM-DP-CSS with CALLING-XFORM-DP-PT inside and CALLING-XFORM-DP-CSS with CALLING-XFORM-TRK-PT inside.
Now create two Device Pools, one for the Trunk and one for your Phones.
PHONES-DP and TRUNK-DP were the Device Pools I created. I am trying to be as specific as possible so everybody understands, so bear with me. :)
Now, put the CSSs accordingly. Go to PHONES-DP "Device Mobility Related Information" field and put CALLING-XFORM-DP-CSS there. Save and reset.
On TRUNK-DP you are not require to do anything, of course there is an "unless", but let's stick with your main concern, otherwise this post will be bigger than it will be.
Go to the Trunk itself and add the CALLING-XFORM-TRK-CSS to the "Incoming Calling Party Settings" field. Uncheck the "Use Device Pool CSS" and leave "Strip Digits" and "Prefix" fields in its defaults. Save and Reset.
Now, create your Calling Party Transformation Patterns. I will explain to you using the Brazilian national standard.
For Local Calls, here in Sao Paulo, we receive 11 which is the area code and 8 or 9 digits after that for local number. So, I created the Calling Party Xform mask as follows:
1 - 11.9XXXXXXXX > discard digit instruction: Predot > Prefix +5511
2 - 11.[2-9]XXXXXXX > discard digit instruction: Predot > Prefix +5511
I didn't change the TON or the Plan because it doesn't matter. SIP doesn't recognize these fields.
Now, same thing for national calls. We receive the area code and 8 or 9 digits. Since we already have specific Patterns for Local calls, it is ok to have a more generic pattern for other area codes
1 - [1-9][1-9]9XXXXXXXX > No Discard Digit Instruction > Prefix +55
For international calls, our SP always sends 00 in front of the number. Knowing that, it was easy to simply add the following pattern.
1 - 00.! > discard digit instruction: Predot > Prefix +
Now, you create the Calling Party Transformation Patterns for Phone Device Pools. Achieving Localization as well.
Local Numbers
1 - \+5511.9XXXXXXXX > discard digit instruction: Predot > No need to prefix anything
2 - \+5511.[2-9]XXXXXXX > discard digit instruction: Predot > No need to prefix anything
National Numbers
3 - \+55.[1-9][1-9]9XXXXXXXX > discard digit instruction: Predot > Prefix 0
4 - \+55.[1-9][1-9][2-9]XXXXXXX > discard digit instruction: Predot > Prefix 0
(in Brazil we used to have the numbers delivered to us as 0 + area code + number. That Prefix is just to achieve localization for us)
There is no need to do anything for International Numbers at this level, because if the number is different from +55, there is nothing to be done, just receive the +E164 format.
After all that you will be able to even do Plus Dialing. If you need any help setting it up, please let me know.
Don't forget that if your provider sends you the digits in a way that you can't do much on CUCM, try to identify the numbers by adding a translation pattern in your Trunk (if it is Cisco, of course) before the numbers enter on CUCM. Then use the Calling Party Transformation Patterns to strip off any identification you gave for the inbound numbers.
I hope this helps.
Please rate useful posts.

What i have to do my itunes dont see mu ipad and ipad doesnt want turn on

hi i dont now what to do. I update ios and after he dont want turn on on the screen is blank with cable usb and itunes but when i conect with itunes his doesnt show

FORCE IPAD INTO RECOVERY MODE
1. Turn off iPad
2. Turn on computer and launch iTunes (make sure you have the latest version of iTune)
3. Plug USB cable into computer's USB port
4. Hold Home button down and plug the other end of cable into docking port.
DO NOT RELEASE BUTTON until you see picture of iTunes and plug
5. Release Home button.
ON COMPUTER
6. iTunes has detected iPad in recovery mode. You must restore this iPad before it can be used with iTunes.
7. Select "Restore iPad"...
Note:
1. Data will be lost if you do not have backup
2. You must follow step 1 to step 4 VERY CLOSELY.

Why i dont see java option in firefox 24?

i want to disable java on firefox. i dont find this option in 24. but it was in the previous version.

The ability to disable individual plugins like [http://kb.mozillazine.org/Java Java] has been available in the Add-ons Manager for some time.
(Just to be clear, the "Enable [http://kb.mozillazine.org/JavaScript JavaScript]" option was removed from the Firefox Options window Content panel in Firefox 23. [http://kb.mozillazine.org/JavaScript_is_not_Java JavaScript is not Java]! See the linked articles for details.)
If you want to disable the Java plugin:
#At the top of the Firefox window, click on the '''Firefox''' button
#Click '''Add-ons'''. The Add-ons Manager tab will open.
#In the Add-ons Manager tab, select the '''Plugins''' panel.
#Click on the '''Java (TM) Platform''' plugin to select it.
#Select '''Never Activate''' in its drop-down menu
Ref:
[[How to turn off Java applets]]

How to see Dispatcher, gateway and Message server port numbers?

Hi All,
Can any body tell me how to know the port numbers of dispatcher, gateway & message server in a particular sap application server.
Thanks & Regards
Srinivas K

Hi,
This is very work around way...
SM51-Choose theserver-there one icon (Magnifying glass) choose detaiols -
There you will get dispatchet port.
SMGW -
dispaly file(icon) -
in that you can find the port on which gate way is working.
SMMS-goto-parametere--display...you can find the port for MS.
SMMS -goto-display details (will give more info) you can find diff ports ...
Apart from these all these services when they start they write log in work dir there we can find which port they are using while starting ..
We can find them in the individual monitoring t-codes like SMMS,SMGW,SM51 etc..or from AL11 go to work dir and look at dev_disp,dev_ms,dev_rd.
This is all very work around...
The best will be services file at os level.
Thanks.

Why i cant see the temperature and GPU voltage

hi ,
I recently purchased MSI GEFORCE FX5200-T128.
when i run 3D!Turbo Experience utility it only shows the clock speeds.
why it does'nt shows the GPU, i/o voltage and Temperature.
I am using the drivers provided by MSI site ver : 56.72
my system is also overheating (specially when i play games like farcry,splinter cell) it also heats up when i am not playing, my vid card came with a copper heat-sink. do i need to use a fan? .
thanks in advance.

Unfortunatly not all graphics cards support hardware monitoring, and it looks like your card does not support it either, so you won't get information regarding your temperature or voltages.
If the heatsink gets really hot, then you most probably need a fan for the card. You could get by the overheating problem (if you have already not done so) by installing adequate cooling in your system, which includes system fans and good airflow. Insert intake fans from the front/side and extract fans at the top/rear.
This way you can ensure that cool air is flowing over your motherboard components and graphics card.
Also try to neaten the cables in your pc to help with better airflow.

Curious - why can some see web page and others cannot?

I've had a couple people tell me that the link that I have provided for my website (a cut and paste of the URL address) doesn't work and the error msg is that the page does not exist.
Others can view the site just fine.
I have, until recently, also been able to view my site without issue but I live in China and it is not unusual for there to be sites temporarily blocked from time to time. I have not been able to bring up my site and thought that this was just due to one of these fun times. But the person not able to access the site is in the US.
So my question is, what other things could bring up an error msg like that? Since it is a cut and past issue, it can't really be a mistyping of the address so looking for any other things that could cause this.
Thanks!

Problem sort of solved.
However, the person who was having difficulty wasn't able to click on the full address URL but when I gave him the shortened version (the version that comes up on iWeb when you are asked if you want to publish it) it was fine. Not really sure why one works and not the other but a solution was at least found.
Posting this in case anyone else runs into a similar situation.

Why am i seeing multiple displays and how do i turn it off?!

I turned my mac on tonight and saw 4 little screens of my desktop instead of just one! How to I get it to be just one big display again?! please help it is very frustrating.

Try modifying your class and your code as follows...
change the class linkage to be Cursor. Not that it is wrong, but classes typically have upper case letters to make them easier to identify in code.
then, do not use the name of the class as the instance name....
var cursor:Cursor;
function initializeGame():void
cursor = new Cursor();
etc...

Why we dont' see H323 gateway and SIP trunk "registered" in CCM?

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