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

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Cucm 10.1version - training videos and hand guides on understanding voice gateway h323 and SIP? thanks

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Learncisco gives a very good introduction to CUCM - I recommend you start there.

Where can I learn the structure of Voice gateway ?

I'm making a essay about Structure of Voice gateway: hardware and software construction. I can not find any books or any manual deal with it . So,can someone give me more information about structure of voice gateway or give me some useful resources please?
I'm waiting for your replying.

Cisco doesn't make public the software structure of their products; to a certain extent it can be inferred from documentation and familiarity with the product, but probably not enough for an in-deep analysis.
For your task, I suggest you focus on some open-source development of GW, that has no secrets.
Hope this helps, please rate post if it does!

ISR as CUBE and Voice Gateway

Can I set an ISR 2951 as CUBE to receive SIP trunks and configure the same box as voice gateway to deliver TDM E1 voice channels to an enterprise PBx?
Thanks
Sent from Cisco Technical Support iPhone App

Yes You can...
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

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.

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

Troubleshooting SS7 Interconnect for Voice Gateway

Hi
Recently we have implemeted SS7 Interconnect for Voice Gateway Solution in our network.
We are using devices PGW2200, NAS 5400, SLT 2611, 4000 Series S/w and GK.
Customers compalint that they are facing Echo, Clip in Voice and No voice related issues. Can you tell us the commands on PGW/GW/SLT to pinpoint. PGW we are using in Call Control Mode.
Please help.
Thanks in Advance.

Hi,
You can check in the /opt/CiscoMGC/etc directory properties.dat file which has the parameter *.EchoCanRequired which should be set to 1. This can binded by an individual trunk-group as well. Also make sure the voice-ports have the echo-cancel coverage and disable VAD to see if it makes it better.
Regards
-Kashif

How to create a Global Contacts in our CM or Voice Gateway

Hi
we have a UCM6.1.2 and a H.323 voice gateway
we get many calls from different vendors and so on
i want to some how assign a contact or a name to the calls that come in often
for example
if we get 100 calls from 973-333-3333 i want that number to show up and also the customer's name for example Customer XYZ
can i do this any where?
is it possible?
any help will be appreciated.
Thanks
Regards

You can configure Caller ID which is an analog service offered by a central office, which supplies calling party information to subscribers.
Caller ID Name Delivery Issues on Cisco IOS Gateways:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00800a9a49.shtml
Caller ID:
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vclclid.html

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.

Configure E1 card on voice gateway

Hi
I have CUCM 9 and Cisco 2801 voice gateway
There is 2 card on cisco 2801 : VIC2-4FXO and VWIC3-1MFT-T1/E1
I would like to know how I will configure the gateway in the CUCM (H323, MGCP ...)
Should I add H323 or MGCP gateway or something else ?
I also need to know how I will configure E1 card on the Cisco 2801 to handle CUCM PSTN incoming and outgoing call
I will apreciate if you help with some how to document
Thanks in advance
Regards

Which protocol to use, is up to you, you should know the requirements of your customer to define that.

How to Save Global Contacts in our CM or Voice Gateway

Hi
we have a UCM6.1.2 and a H.323 voice gateway
we get many calls from different vendors and so on
i want to some how assign a contact or a name to the calls that come in often
for example
if we get 100 calls from 973-333-3333 i want that number to show up and also the customer's name for example Customer XYZ
can i do this any where?
is it possible?
any help will be appreciated.
Thanks
Regards

You can configure Caller ID which is an analog service offered by a central office, which supplies calling party information to subscribers.
Caller ID Name Delivery Issues on Cisco IOS Gateways:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00800a9a49.shtml
Caller ID:
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vclclid.html

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

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

How to make CUCM as a TFTP server , then copy files to Voice Gateway ?

how to make CUCM as a TFTP server , then copy files to Voice Gateway ? anyone knows?

Hi,
Please check the following link
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/6_1_1/ccmsys/cucm-accm-611/a02tftp.html#wp1023004
Understanding How Devices Access the TFTP Server
You can enable the IP phones and gateways to discover the TFTP server IP address in one or more of the following ways, depending on the device type:
•Gateways and phones can use DHCP custom option 150.
Cisco recommends this method. With this method, you configure the TFTP server IP address as the option value.
•Gateways and phones can use DHCP option 066.
You may configure either the host name or IP address of the TFTP server as the option value.
•Gateways and phones can query CiscoCM1.
Ensure the Domain Name System (DNS) can resolve this name to the IP address of the TFTP server. Cisco does not recommend this option because it does not scale.
•You can configure phones with the IP address of the TFTP server. If DHCP is enabled on the phone, you can still configure an alternate TFTP server IP address locally on the phone that will override the TFTP address that was obtained through DHCP.
•Gateways and phones also accept the DHCP Optional Server Name (sname) parameter.
•The phone or gateway can use the value of Next-Server in the boot processes (siaddr).
Devices save the TFTP server address in nonvolatile memory. If one of the preceding methods was available at least once, but is not currently available, the device uses the address that is saved in memory.
You can configure the TFTP service on the first node or a subsequent node, but usually you should configure it on the first node. For small systems, the TFTP server can coexist with a Cisco Unified Communications Manager on the same server.
HTH
Manish

Voice gateways, SLT and PGW

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