Voice Gateway & Gatekeeper On a single Router?

Hello All,
Has anyone had experience with putting voice gateway and voice gatekeeper functionality into one router? We will have three clusters initially (with about 1,000 users worldwide though) and was considering one gatekeeper per cluster for CAC and tail end hop off routing for LD/International calls.
Any guidance on what sort of platform these should be running on if kept separate? DSP resources in them, etc.?
The voice gateways will be a mixture of 2851 and 3800 series routers. Thanks in advance for the help.
Thanks,
Dave

Hi,
I have done this many times in lab, dont know if that will be ok in production if you have a good powerfull problem i dont think that it will be a problem.
About the DSP's maybe you can use the DSP calculator - anyway this refers to many parameters on the network.
BR,
Teo

Similar Messages

Video Calls on Voice gateway enabled router

Hi All,
I am having cisco 3925 router which is acting as MGCP voice gateway for PSTN calls,
Is it possible to use same MGCP voice gateway router to do the video calls .
thanx in advance
Regards
Deepak

No, we can only use H.323 / SIP for video calls across ISDN.
H.323 - H.320
http://www.cisco.com/en/US/partner/docs/video/milticomm/h320Bonding.html
SIP - H.320
http://www.cisco.com/en/US/partner/docs/ios/voice/sip/configuration/guide/sip_cg-rsvp_ps10592_TSD_Products_Configuration_Guide_Chapter.html
Arun

LAB OF GATEWAY, GATEKEEPER & CUCM

Hi
I want to do practice on cucm, gateway & gatekeeper through lab enviroment. Please give me some kind of scenario to create lab . I want practice on data & voice enviroment kind of thing through data vlan & voice vlan. I have following device for practice
1 cisco router 3600
1 cisco router 2501
1 L3 switch HP procurve 24 port
3 L2 switch hp procurve 24 port
1 broadband connetion
2 laptop with good configuration
cucm 7 iso
ip communicator
softphone (sip based)
please suggest me any kind of network design (voice & data) according to available metiral with me so that I can start practice.

Wrong forum, try "ip telephony". You can move the post using the control panel on the right.
Anyway it seems to me that you don't have enough equipment.

How to install permenant UC License on 2951 voice gateway

Hi All,
I have C2951 Voice gateway and I want to install Permenant UC license and I don't know how to get it and how to install on it and how to activate it,so I would like to request all of you that if you have any idea please explain me in details with step by step, i will be appreciate all of you and hop I will receive response form you very soon.
Thanks
Rizwan

Hi Rizwan,
a. u need to send the mail to [email protected] along with PAK code and serial number of 2951 router.this u can get by running command show version or show license udi
b. after u recive the license file[uck9_license.lic], upload this file in router flash using command copy tftp flash:
c. u can check the same by running command show flash:
d. now u need to install the license file . run the command license install flash:uck9_license.lic
e. now run the command license boot module c2900 technology-package uck9 and accept :yes
f. reboot the router
g. after the router reboots, run command show license feature /sh license all.
regds,
aman

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

Changing CAS e&m-wink-start to a PRI on voice gateway

Cisco IOS Software, C3900 Software (C3900-UNIVERSALK9-M), Version 15.1(4)M4, RELEASE SOFTWARE (fc1)
I want to change a CAS e&m-wink-start to a PRI on controller T1 0/1/0 using mgcp below. I am posting the new configuration below the current e&m-wink-start configuration on the voice gateway router.
Current
controller T1 0/1/0
cablelength long 0db
ds0-group 1 timeslots 1-24 type e&m-wink-start
description
New configuration
Router(config)# no contoller T1 0/1/0
Router(config)# no interface Serial0/1/0
Router(config)#controller t1 0/1/0
Router(config-controller)#cablelength long 0db
Router(config-controller)#framing esf
Router(config-controller)#linecode b8zs
Router(config-controller)#clock source line
Router(config-controller)#pri-group timeslots 1-24 service mgcp
Router(config-controller)#description circuit ID
Router(config-if)# interface serial0/1/0
Router(config-if)# no ip address
Router(config-if)# encapsulation hdlc
Router(config-if)# isdn switch-type primary-4ess
Router(config-if)# isdn incoming-voice voice
Router(config-if)# isdn bind-l3 ccm-manager

Hi,
You need additional configs for this to work..You also need to configure the isdn bind on the D-channel not as you have it..
Router(config-if)# interface serial0/1/0:23
Router(config-if)# no ip address
Router(config-if)# encapsulation hdlc
Router(config-if)# isdn switch-type primary-4ess
Router(config-if)# isdn incoming-voice voice
Router(config-if)# isdn bind-l3 ccm-manager
Additional configs....change ip to suit your needs
ccm-manager redundant-host 192.168.103.114
ccm-manager mgcp
ccm-manager music-on-hold bind xxx--put relevant interface
mgcp
mgcp call-agent 192.168.103.115 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp modem passthrough voip codec g711alaw
mgcp modem passthrough voip redundancy
mgcp ip qos dscp af31 media
mgcp ip qos dscp cs3 signaling
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
mgcp default-package dtmf-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
mgcp tse payload 100
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interfacexx -----------------------put relevant interfcae here
mgcp bind media source-interface xxx--------------------same
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"

Can't establish a Voice gateway (cisco 2911) using SIP with CUCM 9.1

I have configured a Cisco 2911 as a Voice Gateway using SIP (the configuration is attached), but unfortunately can't establish a test call to a phone (CUPC 8.6 SCCP) using csim start. I have done logging the ccsip debug and ccapi debug and attached them. Could anyone help me to solve this problem?

I just did some research on my end and csim is not supported for SIP. The Invite will never be created and sent to the CUCM to initate the call. It disconnects in the router itself with normal cause.
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/sipSPIOutgoingCallSDP:
Could not create source SDP for Outgoing Call
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/sipSPICreateOutboundSDP:
Error in creating an SDP for the outbound call - Check for supported codecs
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/preprocessSetup:
Error during outbound SDP creation
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(16) for outgoing call
Please use an actual call to test your dial-peer and integration with call manager. csim will not work.
Hantale
Sree

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

Voice Gateway DFM Notifications

We have a 2800 route which is configured as a Voice Gateway.
For some reason, yesterday we started getting DFM notifications like the one below.
Any idea's what the means? We seems to get an active messages every few 10mins or so, then a few minutes later the message will clear before becoming active again...
EVENT ID                = 0009NK2
ALERT ID                = 0000QTM
TIME                    = Tue 21-Jun-2011 16:06:50 BST
STATUS                  = Active
SEVERITY                = Critical
MANAGED OBJECT          = XXXXVG1
MANAGED OBJECT TYPE     = Routers
EVENT DESCRIPTION       = OperationallyDown::Component=IF-XXXXVG1/39 [FXO NULL 0/0/0:15];ComponentClass=Interface;ComponentEventCode=1075;Type=VOICEFXO;IsFlapping=false;AdminStatus=UP;MaxSpeed=0;OperStatus=DOWN;InterfaceCode=CODEUNKNOWN;Mode=NORMAL;DuplexMode=FULLDUPLEX;La

Hi,
We started getting the exact same thing yesterday as well. Can you confirm what IOS version you're running. We're on c2800nm-ipvoicek9-mz.124-15.T5.bin. We've got 2 x E1 cards installed and are getting on both:
EVENT DESCRIPTION       = OperationallyDown::Component=IF-XXXXXGW1/102 [FXO NULL 0/1/0:15];ComponentClass=Interface;ComponentEventCode=1075;Type=VOICEFXO;IsFlapping=false;AdminStatus=UP;MaxSpeed=0;OperStatus=DOWN;InterfaceCode=CODEUNKNOWN;Mode=NORMAL;DuplexMode=FULLDUPLEX;L
EVENT DESCRIPTION       = OperationallyDown::Component=IF-XXXXXGW1/39 [FXO NULL 0/0/0:15];ComponentClass=Interface;ComponentEventCode=1075;Type=VOICEFXO;IsFlapping=false;AdminStatus=UP;MaxSpeed=0;OperStatus=DOWN;InterfaceCode=CODEUNKNOWN;Mode=NORMAL;DuplexMode=FULLDUPLEX;La
EDIT:
These alerts suddenly stopped at approx 3AM GMT today. Out of interest which Telco do you use? We're using Cable & Wireless.

Voice gateway PRI channels

Hello,
Does anyone know which command on a voice gateway will make it use all 24 channels on the controller without having channel D for signaling?

Hi,
In case you have multiple PRI's you can use NFAS which allows to control multiple PRI's with a single d-channel
http://www.cisco.com/c/en/us/support/docs/dial-access/integrated-services-digital-networks-isdn-channel-associated-signaling-cas/9584-quadt1-nfas.html
HTH
Manish

Etherchannel on a Voice Gateway

We're considering etherchannel on a 2851 voice gateway. The goal is to use both of the dual Gig LAN interfaces for redundancy. Anyone with recommendations please let me know.
Thanks

Hi,
Instead of Etherchannel, how about using the backup interface command on the second NIC of the router.
The benefit is we can use the same IP on the second NIC as the first/primary NIC.
The backup interface remains in standby mode and becomes alive only when the primary NIC is down.When the primary NIC comes up the secondary automatically goes in standby mode.
Advantages : "Backup interface" command is independent of routing protocols. That is, it does not depend on routing protocol convergence, route stability and so on.
Each of the voice gateway NICs can be conencted physically to LAN swiches.
The Voice gateway protocol which can be easily used with this setup is H323 or SIP, not sure on MGCP though.
Only thing is the calls active calls will drop when the primary interface goes down, or comes back again(the secondary NIC goes to standby mode) which is the normal behaviour.
Can you please feedback on the usage of "backup interface" command for configuring NIC resilience on the voice gateway?

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

How does a voice gateway handle a call received from CUCM?

For example, I have a voice gateway configured via a SIP trunk as a device in CUCM. When our users dial the international pattern (8011!) and CUCM forwards that to the device(gateway) over the SIP trunk, how does the gateway handle that request in terms of matching dial-peers? Is it handled the same way a call incoming from the PSTN would be? By matching the destination pattern?
Ultimately what I'm trying to do is figure out whether our international calling is configured properly but that's difficult to do If I don't know what the gateway is doing with the call once it receives it from the CUCM.
Thanks in advance!

Thanks, we have a conference call Monday and I'll try some debugging to see what's going on. I am a little more confused now if that's possible because my understanding is that they can make INTL calls but based off of the configs I've seen and what you've told me they should be able to. Below are my configs. There is a translation profile applied but it points to a blank rule. I also didn't see any transformations in the route pattern.
mgcp profile default
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
port 0/0/0:23
dial-peer voice 100 voip
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad
dial-peer voice 101 voip
destination-pattern 5...
session protocol sipv2
session target ipv4:x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad
dial-peer voice 2 pots
destination-pattern 911
port 0/0/0:23
forward-digits all
dial-peer voice 3 pots
destination-pattern 9[94]11
port 0/0/0:23
forward-digits 3
dial-peer voice 5 pots
translation-profile outgoing NATL
destination-pattern 9[2-9]......
port 0/0/0:23
forward-digits 7
dial-peer voice 6 pots
translation-profile outgoing NATL
destination-pattern 9707[2-9]......
port 0/0/0:23
dial-peer voice 7 pots
translation-profile outgoing NATL
destination-pattern 91[2-9]..[2-9]......
port 0/0/0:23
forward-digits 11
dial-peer voice 8 pots
translation-profile outgoing INTL
destination-pattern 9011T
port 0/0/0:23
prefix 011
voice translation-rule 10
rule 1 // // type any international plan any unknown

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

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

Voice Gateway & Gatekeeper On a single Router?

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