Connecting CME 4.0 to Voice Gateway

I have a 2600XM running CME 4.0 and a 1760 with 2 FX0 and 1 FXS card. Both routers are running IOS 10.4(9)T. I would like the 1760 to act as a gateway using H.323. Can anyone advise me on how to set this up?

on 2600
dial-peer voice 1 voip
dtmf-relay h245-alpha
codec g711ulaw
no vad
destination-pattern 9T
ip qos dscp cs3 signalling
On 1760
dial-peer voice 1 voip
dtmf-relay h245-alpha
codec g711ulaw
no vad
incoming-called-number 9T
ip qos dscp cs3 signalling
dial-peer voice 1 pots
port 1/0 ---- fxs port
destination-pattern 9T
HTH
Sankar
PS: please remember to rate posts!

Similar Messages

CME 4.0 to Voice Gateway

I have a 2600XM running CME 4.0 and a 1760 with 2 FX0 and 1 FXS card. Both routers are running IOS 10.4(9)T. I would like the 1760 to act as a gateway using H.323. Can anyone advise me on how to set this up?

Take a look at this URL:
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guide09186a00800b65dd.html

Site to Site Connectivity Between BE6K and Voice Gateway 2901

Greetings,
Is an Ethernet handoff required for site to site connectivity between BE6K and a voice gateway 2901. My vendor is suggesting that it's required in order for both sites to see the BE6K as one phone system. However, here in lies the problem. I have a point-to-point T-1 between the sites that does not have an Ethernet handoff, just the smartjack to the T-1.
What would I need to get this to work? Have a router at each site? If so, which model? Or is there a component I could add to the BE6K or voice gateway?
Any help would be greatly appreciated.
Thanks in advance.

Ethernet handoff just means that the provider will deliver the circuit using Ethernet. What circuit is your provider delivering for you? Do they manage your WAN? Like I said what you need is ip connectivity between your sites and BE6k. If your T1 connection provides WAN connectivity and you have ip connectivity between the sites, then I don't know what you need any handoff for. The question is do you have ip connectivity between the sites via your T1 connection

Setup of 2 B-channel transfer (TBCT) on Cisco Voice Gateway

Hi all,
In my scenario, I have several remote locations, each with their own T1-PRI connection to the PSTN via Cisco voice gateways.
The call flow has the caller calling a PSTN number at the remote location. This call hits a CUCM translation pattern and is immediately hairpinned/tromboned out to a PSTN number located at the UCCE central location.
The customer would like to perform a 2 b-channel transfer back up to the CO switch, to free up B-channels to the remote location.
My question is this: Assuming my switch type is supported and the PSTN provider supports 2 b-channel transfers, do I simply need to add "isdn supp-service tbct" to my serial interface configuration?
I have seen discussions that it is as simple as this, to needing some tcl/xml code, to also requiring aaa to be setup.
As there are over 100 locations being considered for this enhancement, the customer would like to know if it is 15 minutes or 2 hours per location to implement.
Thanks,
Joe

Thanks Paolo,
That was my expectation.
They have this going on the VG's utilized by CVP, so I will ask them to review the XML in use there for tweaking for the remote locations.
Thanks again!
Joe

**ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) in SIP voice gateway

Hi all,
I am configuring a SIP voice gateway. After finsihing the config, the outbound calls are working properly. But I met some issue for inbound calls.
There is no debug log for show ccsip message and show voice dialpeer command. Only q931 has the debug log.
Can any one have any idea for this issue?
below is the debug shows for show isdn q931.
   Jun 1 05:27:49.435: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x03A3
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xE1858384
Preferred, Interface 5, Channel 4
Calling Party Number i = 0x2181, '18600586101'
Plan:ISDN, Type:National
Called Party Number i = 0xC1, '82197910'
Plan:ISDN, Type:Subscriber(local)
High Layer Compat i = 0x9181
High Layer Compat i = 0x9181
Jun 1 05:27:49.439: ISDN Se0/0/0:15 Q931: Received SETUP callref = 0x83A3 callID = 0x0004 switch = primary-net5 interface = User
d8w-sr-2811f20a-vpn#
Jun 1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting call
Jun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3
Cause i = 0x80E418 - Invalid information element contents
下午01:28:17: Zhuliang: Jun 1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting call
Jun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3
Cause i = 0x80E418 - Invalid information element contents

Channel ID i = 0xE1858384 Preferred, Interface 5, Channel 4 Jun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3 Cause i = 0x80E418 - Invalid information element contents下午01:28:17: Zhuliang: Jun 1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting callJun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3 Cause i = 0x80E418 - Invalid information element contents
To what are you connecting? Call comes with explicit interface indicator IE, that is not normal for ISDN E1.

How to verify voice channels on voice gateway

Hello,
I am in the testing phase of deploying a 2921 voice gateway with 2 PRIs. After connecting the T1s, how do I verify if all the channels are up and ready for use? Also, how do I verify if clocking on the T1 interfaces are configured correctly?
This 2921 is a replacement for a 2811 currently being used. When I login to the 2811 via SSH, I see this:
560240: Aug 13 20:37:24.368: %ISDN-6-CONNECT: Interface Serial0/2/0:3 is now connected to 3854143418 N/A
560241: Aug 13 20:37:24.368: %ISDN-6-CONNECT: Interface Serial0/2/0:3 is now connected to 3854143418 N/A
560242: Aug 13 20:37:24.544: %ISDN-6-CONNECT: Interface Serial0/2/0:5 is now connected to 7753428028 N/A
560243: Aug 13 20:37:24.544: %ISDN-6-CONNECT: Interface Serial0/2/0:5 is now connected to 7753428028 N/A
560244: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:4 is now connected to 7753296300 N/A
560245: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:4 is now connected to 7753296300 N/A
560246: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:22 is now connected to 3296300 N/A
560247: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:22 is now connected to 3296300 N/A
560248: Aug 13 20:37:27.652: %ISDN-6-CONNECT: Interface Serial0/2/0:20 is now connected to 3525101 N/A
560249: Aug 13 20:37:27.652: %ISDN-6-CONNECT: Interface Serial0/2/0:20 is now connected to 3525101 N/A
560250: Aug 13 20:37:30.320: %ISDN-6-CONNECT: Interface Serial0/2/1:22 is now connected to 3525101 N/A
560251: Aug 13 20:37:30.320: %ISDN-6-CONNECT: Interface Serial0/2/1:22 is now connected to 3525101 N/A
How do I do that with the 2921 in an SSH session? I've enabled "logging monitor" and "term mon" but I still can't see messages like that.
I thank you very much for your time.

Hello,
To check clocking you can use the "show network-clock" command. For example:
SiteA-RTR#show network-clock
Network Clock Configuration
Priority Clock Source Clock State Clock Type
1 T1 0/0/0 GOOD T1
10 Backplane GOOD PLL
Current Primary Clock Source
Priority Clock Source Clock State Clock Type
1 T1 0/0/0 GOOD T1
The above is an example of a "good" configuration, assuming that T1 0/0/0 is the controller terminating the line from your carrier. In your case, it would be T1 0/2/0 or 0/2/1. You can also look at "show controller t1 [interface]" (e.g. show controller t1 0/2/0) and pay particular attention to "Line Code Violations", "Slip Secs", and "Errored Secs" counters. Note that the "Clock Source Line" status in the "show controller t1" command is not enough to verify you have proper clocking.
Example of a "bad" clock scenario:
SiteA-RTR(config-voiceport)#do sh controller t1 0/0/0
T1 0/0/0 is up.
Applique type is Channelized T1
Cablelength is long 0db
No alarms detected.
alarm-trigger is not set
Soaking time: 3, Clearance time: 10
AIS State:Clear LOS State:Clear LOF State:Clear
Version info Firmware: 20080918, FPGA: 13, spm_count = 0
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
CRC Threshold is 320. Reported from firmware is 320.
Data in current interval (483 seconds elapsed):
3 Line Code Violations, 4 Path Code Violations
207 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
206 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 109 Unavail Secs
To make sure the clock is set correctly, look at the "network-clock-participate" and the "network-clock-select" commands.
For the ISDN status information, you can use "show isdn status". For example:
SiteA-RTR(config)#do sh isdn stat
Global ISDN Switchtype = primary-ni
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply
ISDN Serial0/0/0:23 interface
dsl 0, interface ISDN Switchtype = primary-ni
L2 Protocol = Q.921 0x0000 L3 Protocol(s) = CCM MANAGER 0x0003
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x800000FF
Number of L2 Discards = 0, L2 Session ID = 15
*Note: The above is output from a gateway provisioned to use MGCP.
For all gateways, just make sure that Layer 1 is "ACTIVE" and that Layer 2 has "MULTIPLE_FRAME_ESTABLISHED".
You can also use the "show isdn service" command to look at status for individual B-channels.
SiteA-Rtr#sh isdn service
PRI Channel Statistics:
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not appl
ISDN Se0/0/0:23, Channel [1-24]
Configured Isdn Interface (dsl) 0
Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend)
Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
State : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 3
Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend)
Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
State : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 2
Use the "Channel State" and "Service State" to verify things are what you expect.
HTH.
-Bill (@ucguerrilla)
http://ucguerrilla.com

Voice gateways, SLT and PGW

Hi everyone,
I dont know if its the right place to start this discussion so forgive me if i am wrong.( I also have opened a discussion in IP telephony portion)
The company in which i am currently wokring is running a call center and IPT as well. i will try to explain the scenario
voicegateways (AS5400 and AS5350 routers with IOS Version 12.4(15)T7)
SLT (2651XM with IOS Version 12.2(8)T10)
PGW (PGW 2200 with SunOS 5.10, MGC)
I was under the impression that if you want to connect PSTN and a voip network you need a PSTN gateway. So why are we using 3 different types of hardware.
Follwing are the explainations in a doucment given to me
the Cisco PGW 2200 provides service providers with the capability to seamlessly route voice and data calls between the PSTN and New World packet networks.
Cisco 2611 Signaling Link Terminals
(E1 terminates, Take Signaling part and send B n D channels to Voice gateway, Signaling part is resolved by it self and PGW)
Voice Gateways
allows terminals of one type, such as H.323, to communicate with terminals of another type, such as a PBX, by converting protocols. Gateways connect an organization’s network to the PSTN
Any information is much appreciated

have you read this
End-of-Sale and End-of-Life Announcement for the Cisco PGW 2200 Softswitch and Software
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2027/end_of_life_notice_c51-676990.html
hope this help

VG202 Voice gateway

I have multiple environments with Avaya Partner phone systems in them. I used touse a multitech VOIP modem for connection back to Headquarters - this works perfectly fine. Situation I am having, when the Multitech units die, I replace them with the Cisco VG202 unit. When placed at the remote locations, the time in the windows of the Avaya phones never corrects it self...it always wrong. It is correct on the Avaya system at the location, but the integration of the Cisco unit seems to make it go awry. All other functionality is there, but the time display is a mess. Any ideas as to where to troubleshoot this? Manually establish the correct time from a console phone at the remote locations, and it goes awry within hours.

Hi Peter,
This forum is for discussions about specific known bugs. You might find better discussion in the IP Telephony forum: https://supportforums.cisco.com/community/netpro/collaboration-voice-video/ip-telephony
Conceptually as an analog voice gateway the VG shouldn't generally provide any time information to an attached phone system. Is it possible the phone system is pointed to the VG as an NTP reference? In that situation the internal platform clock isn't terribly reliable and synch to an more reliable external NTP reference is recommended. The only other possibility that comes to mind is delivery of a timestamp information along with callerid over the analog voice ports. In that situation the router again depends on internal clock. Configuring an NTP reference on the router will help to keep the router clock better synchronized. These are all a bit of a stretch. The most likely, IMHO, is the pone system pointing to VG202 as NTP reference. In that case giving the VG202 a reliable NTP reference so the router can keep more accurate time would be my first approach.
Hope this helps,
Wes

Analog interface between voice gateway and pbx system, please suggest too

Hi
Im a fresh in telephony and voip technology. I didnt find information about design analog interface between voice gateway and pbx system such as fxo-to-fxo, fxo-to-fxs, fxs-to-fxs, fxo-to-e&m, fxs-to-e&m, or e&m-to-e&m. Could you please suggest any resource for me?
Ps. my networks have 1 HQ and 2 branch offices connect to the HQ.
Thanks,
Nitass

Hello,
you can find lots of info about analog voice interfaces here:
http://www.cisco.com/en/US/tech/tk652/tk653/tk754/tech_protocol_home.html
I've found this particular document very useful when trying to understand how telephone signaling works. This applies to any telephone system where interfacing to analog phones or the PSTN is involved.
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800a6210.shtml
Regards.

Voice gateway Config issue

Hello everyone;
I do have an analog line that i want to affect to (04) four IP Phones, so that any one of these four ip phones users can use this same analog line, and when we call that analog line number, any one of these four ip phone users can answer.
Is this possible to do on the Voice gateway?
I think about configuring many connection plar under one voice port, like this:
voice port 1/0/0
connection plar opx 1000
connection plar opx 1001
connection plar opx 1002
connection plar opx 1003
And about, puting many answer adress under dial-peer voice this way:
Dial-peer voice
answer adress 1000
answer adress 1001
answer adress 1002
answer adress 1003
but i don't know if it will work , and don't see how it could be done for the voice translation rule and the voice translation profile.
Ragards.
Camélia

This will be done in only specific Dial-peer and Voice port.
A voice Translation rule is not need because once the voice-port number is dialed for example 011-27272, the plar command forward it 2001 and the dial-peer 10 is matched, you only need to configure a hunt-pilot in CUCM.
Voice-Port 1/0/0
connection plar opx 2001
2.) You will need to match incoming calls from PSTN, you can do this by use the below commands to match all incoming calls from PSTN.
Dial-peer voice 1 Pots
Incoming called-number .
3.) Finally you will need to send calls to 2001 via CUCM to match the Hunt-Pilot.
Dial-peer voice 10 voip
destination-pattern 2001
session-target ipv4:<<CUCM-IP-Address>>
Please Rate.

Cisco 2911 Voice Gateway SIP PSTN Calls Fail

Hello All,
    I am having trouble with outboud SIP PSTN calls through a Cisco 2911 Voice Gateway. 2911 VG terminates PSTN SIP Traffic and connects to Avaya CS1000M via QSIG PRI Trunks. When calls are attempted outbound fron the PBX the caller gets a fast busy. Debug ISDN q931 shows the call hitting the 2911 properly, debug voip ccapi inout shows the call matching the correct dial peers and debug ccsip shows the invite to the PSTN Provider SBC, however within the invite the "from" address incorrectly shows the calling number with the provider SBC address (see below). does anyone have any insight on how to correct this? Attached are VG config and Debug isdn q931, voip ccapi inout, ccsip messages and ccsip call. Thanks in advance for any help!!
From: <sip:[email protected]>:tag=6166CDC4-882
To: <sip:[email protected]>
Shawn C. Smith

i have same problem my cucm ip is 192.168.200.53
my Voice Gateway is SIP by ip 192.168.200.86 for internal
and 172.29.7.94
and my SIP Server is 10.208.9.69
if its oky can yuo take a look at my problem please
this is the syslog from debug
May 30 20:19:34.284: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.200.53:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
Session-Expires: 1800
P-Asserted-Identity: "Aysar Mohamed" <sip:[email protected]>
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
May 30 20:19:34.284: //-1/B5A494800000/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=2217156
   ----- ccCallInfo IE subfields -----
   cisco-ani=2217156
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=90555769123
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
May 30 20:19:34.288: //-1/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x30CF41D4, Call Info(
   Calling Number=2217156,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=90555769123(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=465
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 1
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832953048,feature_id:85
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=90555769123(TON=Unknown, NPI=Unknown))
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
   Event=0x2B82D890
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 90555769123
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetContext:
   Context=0x2ABC2E44
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 465 with tag 0 to app "_ManagedAppProcess_Default"
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=802, Params=0x2ABC19D4, Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
   In: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
   Out: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Destination Pattern=9T, Called Number=0555769123, Digit Strip=FALSE
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0555769123(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=Aysar Mohamed
   Account Number=2217156, Final Destination Flag=TRUE,
   Guid=B5A49480-0001-0000-0000-11C535C8A8C0, Outgoing Dial-peer=802
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=2217156
   ----- ccCallInfo IE subfields -----
   cisco-ani=2217156
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=0555769123
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x30CF41D4, Interface Type=3, Destination=, Mode=0x0,
   Call Params(Calling Number=2217156,(Calling Name=Aysar Mohamed)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0555769123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=802, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 2
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832952824,feature_id:86
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccCallSetContext:
   Context=0x2ABC1984
May 30 20:19:34.292: //465/B5A494800000/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=802
May 30 20:19:34.292: //466/B5A494800000/CCAPI/cc_api_call_proceeding:
   Interface=0x30CF41D4, Progress Indication=NULL(0)
May 30 20:19:34.292: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
May 30 20:19:34.292: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1401481174
Contact: <sip:[email protected]:5060>
Call-Info: <sip:172.29.7.94:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
May 30 20:19:34.300: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
CSeq: 101 INVITE
Content-Length: 0
May 30 20:19:34.612: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Contact: <sip:[email protected]:5060;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Content-Length: 328
Content-Type: application/sdp
v=0
o=- 17192647 17192647 IN IP4 10.208.9.69
s=SBC call
c=IN IP4 10.208.9.69
t=0 0
m=audio 39910 RTP/AVP 8 0 102 102 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 AMR/8000
a=rtpmap:102 AMR/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=fmtp:116 0-15
a=fmtp:18 annexb=yes
May 30 20:19:34.612: %SIP-3-UNSUPPORTED: Unsupported ptime value
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=466,
   Caps(Codec=0x2, Fax Rate=0x2, Vad=0x1,
   Modem=0x0, Codec Bytes=160, Signal Type=2)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event=170, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event=98, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
   Interface=0x30CF41D4, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),
   Cause Value=0
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
   Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
   Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0
   Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
   Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccConferenceCreate:
   (confID=0x30C11410, callID1=0x1D1, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
   (confID=0x30C11410, callID2=0x1D2, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   Conference Id=0x30C11410, Call Id1=465, Call Id2=466, Tag=0x0
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
May 30 20:19:34.616: ccConferenceCreate: ret1=0, codecMask1=2, bytes1=160, negot1=0, dtmf1=0
                    ret2=0, codecMask2=2, bytes2=160, negot2=1, dtmf2=6,
                    tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
                    tx_dynamic_pt2=8, rx_dynamic_pt2=8, codec_mode2=0, params_bitmap2 =0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   delay media to slow start case, codec negotation is not done
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_api_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
   Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/B5A494800000/CCAPI/cc_api_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x16, Destination Call Id=466)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x16, Destination Call Id=465)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_process_notify_bridge_done:
   Conference Id=0x16, Call Id1=465, Call Id2=466
May 30 20:19:34.616: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 233
v=0
o=CiscoSystemsSIP-GW-UserAgent 2639 5276 IN IP4 192.168.200.86
s=SIP Call
c=IN IP4 192.168.200.86
t=0 0
m=audio 18288 RTP/AVP 8 0 19
c=IN IP4 192.168.200.86
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Reason: Q.850;cause=127;text="interworking unspecified"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
   Cause Value=41, Interface=0x30CF41D4, Call Id=466
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccCallReleaseResources:
   release reserved xcoding resource.
May 30 20:19:34.680: //466/B5A494800000/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=466
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccConferenceDestroy:
   Conference Id=0x16, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
   Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=466,
   Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: :cc_free_feature_vsa freeing 31A5D9F0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: vsacount in free is 1
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=41
Content-Length: 0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=465,
   Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: :cc_free_feature_vsa freeing 31A5DAD0
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: vsacount in free is 0
May 30 20:19:36.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.7.94:5060 SIP/2.0
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
May 30 20:19:36.048: //467/8DAABF6C8278/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>;tag=739BBC-1CE2
Date: Fri, 30 May 2014 20:19:36 GMT
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 446
v=0
o=CiscoSystemsSIP-GW-UserAgent 3496 1601 IN IP4 172.29.7.94
s=SIP Call
c=IN IP4 172.29.7.94
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.7.94
m=image 0 udptl t38
c=IN IP4 172.29.7.94
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
My SIP GW internal ip address is 192.168.200.86
and the Public IP is : 172.29.7.94
My CUCM is 192.168.200.53
my GW Config is :
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 t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 3
rule 1 /^9/ /\1/
voice translation-rule 4
rule 4 /^22217/ /7/
rule 5 /^2217/ /7/
rule 6 /^022217/ /7/
rule 7 /^0122217/ /7/
voice translation-rule 5
rule 1 /^5/ /905/
rule 2 /^1/ /901/
rule 3 /^2/ /902/
rule 4 /^3/ /903/
rule 5 /^4/ /904/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 10 /^00/ /900/
rule 11 /'+'/ /900/
voice translation-profile OUT
translate called 3
voice translation-profile REDIAL
translate calling 5
voice translation-profile SIP-NEW
translate called 4
application
service mva http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
service ccm http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
license udi pid CISCO2921/K9 sn FCZ164960G0
hw-module pvdm 0/0
hw-module pvdm 0/1
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.200.86 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
ip address 172.29.7.94 255.255.255.252
duplex auto
speed auto
ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip route 0.0.0.0 0.0.0.0 192.168.200.1
ip route 10.208.9.0 255.255.255.0 172.29.7.93
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 192.168.200.53 identifier 1 priority 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register NAGHI-MTP
dspfarm profile 2 mtp
codec g711alaw
maximum sessions hardware 25
associate application SCCP
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern 9T
session protocol sipv2
session target ipv4:10.208.9.69:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay sip-notify rtp-nte sip-kpml
no vad
dial-peer voice 811 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 812 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 813 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 814 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 815 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 816 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 817 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 818 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
Please i need ur help ASAP

Need help for voice gateway NetWeaver

Hi Experts,
I am working on NetWeaver voice.
I would like to know whether i can download any trail version of voice Gateway so that i can integrate with voice application.
These are the list of gateways provided by SAP which supports connectivity.
[http://help.sap.com/saphelp_nwce10/helpdata/en/45/bf1c27692f52f5e10000000a1553f7/content.htm]
Few voice gateways are not supported in India.
I need info on trail version software which i can use with regular microphone n speaker attached to CPU Or any free trail Gateways which i can use for preparing a demo.
Regards,
Syed

Hi,
refer the following link for installation guide of netweaver voice ce 7.1
in the following link it is for nwce 7. 1 sp 3
http://www.sdn.sap.com/irj/scn/go/portal/prtroot/docs/library/uuid/e0e9cc7b-0469-2a10-d3a8-f4f04b30fb0b?overridelayout=true
Regards
Manohar

Could CUOM back up voice gateway configuration?

Hello all,
I wonder if CUOM could back up voice gateway configuration.
And, I assume Cisco works back up router/VGW config in most of cases, is it correct?
Thank you in advance,
master001

By design, the iphone will sync itunes content with ONE computer at a time. If you attempt to sync such content with a second computer, ALL itunes content will first be erased from your phone & then replaced with the content from the second computer. This is a design feature & cannot be overridden. Because you formatted your computer, the iphone will see your computer as a "new" computer.
1.First, disable auto sync when an ipod/iphone is connected, in preferences, in itunes, which in windows is in the edit menu.
2. Put one contact & one event in whatever programs you use for that purpose on your computer, they can be fake, doesn't matter, the important point is to have one entry in each.
3. Connect your phone, itunes running, DO NOT SYNC.
4. Go up to Store>Authorize this Computer.
5. Go up to File>Transfer Purchases. All of the purchased itunes content on your phone will be transferred, music you ripped on your own will not be transferred. You will have to use third party software to first extract that music from your phone BEFORE YOU DO ANYTHING. Same for photos not in your camera roll, the photo sync is one way...computer to phone. This is one example:
http://www.wideanglesoftware.com/touchcopy/index.php
6. Right click the phone in the device pane & select Reset Warnings.
7. Right click the phone, again, & select Backup.
8. Right click the phone, again, & select Restore from Backup. Select the backup you just made. Voice Memos are included in the iphone's backup.
9. This MUST be followed by a sync to restore your itunes content, which you select as before from the various tabs.
You'll get a popup regarding your contacts & calendars asking to merge or replace, select merge.
This article details what's included in the backup you'll make & restore from:
http://support.apple.com/kb/HT1766

Troubleshooting no ringback voice gateway

Hi all,
I'd like to ask about voice gateway that installed on my customer site. below is the topology:
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PSTN – Voice Gateway – C2960-48PST-L – Callmanager and IP Phone
|
|(FO)
Analog phone – PBX.A – Voice Gateway –––– Voice Gateway – PBX.B – Analog phone
the problem is that there's no ring back tone when calling to another site analog phone from ip phone/analog phone. The PBX (NEC) are using E1 connection to the router. Does the ringback provided by PBX or the Voice gateway?
Thanks in advance
Dias

Yes, we did find a solution. It ended up it was on the carrier's end. I believe that they were generating ring back inbound to calls we originated, but not outbound to calls originated from the carriers. They switched that so that they did not send us ring back, but did send carriers ring back for inbound calls and that appeared to solve the issue.
Here's my carrier's explanation in case that wasn't confusing enough: "We made in-band tone available to the SIP trunk via an egress profile change on the SONUS Gateway. Early-media is determined / initiated by
the called or far-end switch."

Call Manager register fxs port with voice gateway- problem

I have a CUCM 6 and a Voice Gateway V224. I've configured the voice gateway's voice FXS ports as MGCP.
I have a Voip connected and registered to the CUCM and a Pots phone connected to the Voice Gateway.
If i dial from the Voip to the Pots phone it rings. The problem is that i cannot ring from the Pots to the Voip phone.
I have no dial tone.
If i write no shut down on the voice port i have a tone. If i configure mgcp on the voice port i have a busy ringtone.
I've entered no mgcp and mgcp commands and i've reset the voice gateway.
How can i call from the pots to the voip phone?
The ios version on the voice gateway is Version 12.4(22)T4.
Here is an outghtput from the Voice gateway.
ccm-manager mgcp
ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 10.1.1.33
ccm-manager config
mgcp
mgcp call-agent CCMIOSS 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp validate domain-name
mgcp rtp payload-type g726r16 static
mgcp profile default
timeout tone busy 600
timeout tone dial 600
dial-peer voice 999223 pots
service mgcpapp
port 2/23
dial-peer voice 999222 pots
service mgcpapp
port 2/22
dial-peer voice 999888 pots
service mgcpapp
port 2/23
The CUCM 6 is registered with the voice gateway.

Is your campaign using CPA? If so, what's the behavior if CPA is not enabled?
I think the best thing to do is to run a trace...
Call Manager > Cisco Unified Serviceability > Trace > Configurations
Select a CUCM server - any subscriber would work.
Service Group - CM Services
Cisco CallManager (Inactive)
Enable SIP Stack Trace and apply to all nodes. Download and install RTMT
Make a bunch of outbound tests and then open RTMT > Trace & Log Central > Collect Files > Check "All Servers" for Cisco CallManager > Next > Next > Relative Range if you made the test calls within the last X minutes, otherwise you can set a From and To datetime. Click Finish and go through your SDL logs and see what errors you find and post them here.
Also, make sure your phone is in the correct CSS in Call Manager

Connecting CME 4.0 to Voice Gateway

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