PSTN Usage in Trunk Configuration

Hello
What is the usage of PSTN usage in the Trunk configuration tab?
I notice that it is not related to anything except the inter Trunk routing
http://masteringlync.com/2013/06/07/inter-trunk-routing-deep-dive/
I have Lync setup and I can route calls (using voice policy- PSTN usage )from the Global trunk configuration which has no PSTN usage assigned to it
can you please clarify the usage of it ?
Thanks

Hi,
PSTN usages are the link between a voice policy and a route. PSTN usages can contain multiple routes, and can be assigned to multiple voice policies. A PSTN usage record specifies a class of call (such as internal, local, or long distance) that
can be made by various users or groups of users in an organization.
If a voice policy doesn’t share a PSTN usage with a voice route, then phone calls made by any user who has been assigned that voice policy will not be able to traverse that route.
You can refer to the link about Enterprise Voice Best Practices in the link below:
http://ucken.blogspot.in/2011/01/enterprise-voice-best-practices-in-lync_21.html
Note: Microsoft is providing this information as a convenience to you. The sites are not controlled by Microsoft. Microsoft cannot make any representations regarding the quality, safety, or suitability of any software or information found there.
Please make sure that you completely understand the risk before retrieving any suggestions from the above link.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Similar Messages

PSTN Usage in Voice Policy

Hi
I am using only one PSTN gateway in my setup to connect Local mobile numbers/National LAN line numbers or international numbers.
Now i want to define PSTN usage to my 3 types of users
1. Local Mobile call users
2. National Mobile/LAN Line number call users
3. International call users
Now i created a dial plan with set of normalization rules for all calls dialling configuration and 3 voice polices for those users. In this case,
should i use same PSTN gateway as PSTN Usage in voice policy configuration?
if i set how it will restrict users on 3 types of calls for those users?
Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if a marked post does not actually answer your question. This can be beneficial to other community members reading the thread. Thanks Chandru CT-HCL
Thanks Chandru CT. MCITP

Hi Thanks for your repies.
I understand. I have to use same PSTN gateway for all 3 PSTN usage records and associate with Voice policy. I create like this.
A user (Only Local Calls - Voice Policy 1)
B user (Only National Calls and Local Calls - Voice Policy 2)
C user (Only International calls, National calls and Local Calls- Voice Policy 3)
My question is
for each PSTN usage i have to define route. how to define route in this below attached image in each PSTN usage record that makes differnces on calls for those users.
I did configure all ignore numbers or addition numbers to make E.164 in normalization rules.
Ex: A users want to dial only local numbers +919003112485 ok. for this what i want to build pattern in route page?
B users want to dial local numbers and national numbers +919003112485 or +914412345678. for this what i want to build pattern in route page?
give me some examples configure route for local/national/internationa call routes.
Thanks
Chandru. R
Thanks

Confused by basic SIP Trunk configuration.

I've went through a few basic SIP trunk configurations and Youtube videos the last couple days but can't figure out what I'm doing wrong.
I've set up H323 and MGCP no problem, but I can't figure out the SIP trunk set up. I'm guessing there are some concepts I'm not understanding yet.
I've got a CUCM lab set up. A 2851 PSTN Simulator, 2851 H323 Gateway at the Main site with a 9.0 CUCM setup in that site and a Branch site that I'm trying to set up as a SIP trunk to connect two phones.
CUCM is on the 192.168.5.x/24 subnet. 172.16.0.x/24 is the subnet connecting the serial(internet) cable between the two gateways in which I'm trying to establish the trunk between.
The Branch phones are still registering with the CUCM at the main site. The Route Pattern is looking to the Branch Route List which has the SIP Trunk listed. I'm just getting a fast busy when trying to place a call from the branch site to the main site.
The most frustrating thing I'm not understanding, is that the debug ccsip and call debugs on my SIP Branch gateway shows absolutely nothing. I've tried registering the branch phones with the SIP Trunk, but stopped when I figured that shouldn't be necessary.
If someone can make some sense of this, I'd truly appreciate it!

Hello Aditya and thanks for the consideration!
I do have a direct IP connection, but I want to set up a SIP trunk and use it just to know how to do it before I do it in production.
I did end up deleting the phones from CUCM so they can register with the 2851 CME that I'm setting up as a SIP trunk. So it is registering there, and I set the allow connections and bind sip commands.
I am now getting Debugs and calls from the SIP Trunk router going to CUCM, but the error message is No Codec, and I Get the fast busy after the call rings on the CUCM Main Site side. So looks like the negotiation is failing. Here is my CLI for the SIP Trunk now after the changes have been made and phones registered to the SIP Branch site as well as the Debug when I tried to place a call to extension "5000":
Note: I did try to change the codecs in the dial-peers to g729r8 instead of 711 and same fast busy after answering.
==============================================
Branch_SIP#show run
Building configuration...
Current configuration : 3529 bytes
! Last configuration change at 03:15:11 UTC Thu Apr 2 2015
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Branch_SIP
boot-start-marker
boot-end-marker
! card type command needed for slot/vwic-slot 0/2
enable secret 5 $1$hOXF$gvfmWW1ZIQE0mAMVg.u1c/
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 10.0.10.1 10.0.10.10
ip dhcp excluded-address 10.0.30.1 10.0.30.10
ip dhcp pool Data
network 10.0.10.0 255.255.255.0
default-router 10.0.10.254
option 150 ip 192.168.5.250
dns-server 192.168.5.200
ip dhcp pool Voice
network 10.0.30.0 255.255.255.0
default-router 10.0.30.254
dns-server 192.168.5.200
option 150 ip 172.16.0.1
ip dhcp pool data
option 150 ip 172.16.0.2
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
bind media source-interface Loopback1
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2851 sn FTX1031A2FM
redundancy
interface Loopback1
ip address 2.2.2.2 255.255.255.255
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.10
encapsulation dot1Q 10
ip address 10.0.10.254 255.255.255.0
interface GigabitEthernet0/0.30
encapsulation dot1Q 30
ip address 10.0.30.254 255.255.255.0
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/3/0
no ip address
shutdown
clock rate 2000000
interface Serial0/3/1
ip address 172.16.0.1 255.255.255.0
clock rate 250000
interface Internal-Service-Module0/0
no ip address
shutdown
!Application: CUE Running on AIM2
hold-queue 512 out
router eigrp 1
network 0.0.0.0
network 2.2.2.2 0.0.0.0
network 10.0.0.0
network 10.0.10.0 0.0.0.255
network 10.0.30.0 0.0.0.255
network 172.16.0.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 172.16.0.2
tftp-server flash:term45.default.loads
tftp-server flash:jar45sccp.8-5-3TH1-6.sbn
tftp-server flash:cnu45.8-5-3TH1-6.sbn
tftp-server flash:apps45.8-5-3TH1-6.sbn
tftp-server flash:dsp45.8-5-3TH1-6.sbn
tftp-server flash:cvm45sccp.8-5-3TH1-6.sbn
control-plane
voice-port 0/0/0
voice-port 0/0/1
mgcp profile default
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
session protocol sipv2
session target sip-server
codec g711ulaw
dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
destination-pattern 5...
session protocol sipv2
session target sip-server
codec g711ulaw
sip-ua
sip-server ipv4:192.168.5.250
telephony-service
codec g711ulaw
max-ephones 24
max-dn 48
ip source-address 172.16.0.1 port 2000
system message SIP Branch Site
cnf-file location flash:
load 7960-7940 P00308010200.bin
max-conferences 8 gain -6
transfer-system full-consult
ephone-dn 1
number 4008
ephone-dn 2
number 4005
ephone 1
device-security-mode none
mac-address 001D.A21A.2065
button 1:1
line con 0
exec-timeout 0 0
line aux 0
line 194
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
speed 115200
line vty 0 4
password cisco
login
transport input all
line vty 5 15
password cisco
login
transport input all
scheduler allocate 20000 1000
end
Branch_SIP#show debug
TFTP:
TFTP Event debugging is on
CCSIP SPI: SIP Call Statistics tracing is enabled       (filter is OFF)
Branch_SIP#
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4B6C5C28
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 4008
Called Number            : 5005
Source IP Address (Sig ): 172.16.0.1
Destn SIP Req Addr:Port : 192.168.5.250:5060
Destn SIP Resp Addr:Port : 192.168.5.250:5060
Destination Name         : 192.168.5.250
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 2.2.2.2
Source IP Port    (Media): 19472
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 63
Disconnect Cause (SIP)   : 503
Branch_SIP#

SIP Trunk Configuration

Hi
We are migrating from Analogue to IP Telephony. I have recieved the following guidlines to configure the SIP Trunk:
*For signaling: use IP : x.x.211.70   ( SIP ) on PORT 5060
*Regarding Numbering Format, use the following:
•             For outgoing Calls :
                The originating Number (A#), should be 96611510XXXX format.
               The Destination Number should be 0NXXXXXX (N area code) or 00XXXXXXXXX (for international)
•             For incoming Calls:
                The Destination Number (B#), should 011510XXXX Format.
                The originating Number (A#), will be 0NXXXXXXX or 00XXXXXXXXXXX Format
*Use Audio Codec's G711-aLaw ; G711-uLaw & G729
*Use T.38 For FAX
*set DTMF to RFC2833
*Make sure to reply with 200Ok for our OPTIONS messages ( ping messages for the SIP)
* configure the following SIP Timers:    “Min-SE=1800 “’ & “Expires=300”
For connectivity consider the following:
SIP CE: 10.65.13.110 (it might be needed to translate this IP to the PBX local IP).
SIP GW: 10.65.13.109
Subnet mask: /30
SIP VLAN: 1191
Notes:
Kindly make sure to have GO SIP GW (x.x.211.70) routed to SIP GW (10.65.13.109) as next-hop.
Kindly make sure to have SIP CE IP addresses are in VLAN 1191.
Can please anyone explain what have to done?
Regards

Ahmed,
Wao..Where do I start...This information is required for configuration on your CUBE..which will be your 2921 router...
Ahmed, here are some pointers I wrote a while ago..
In addition to these points, you will need to configure your cube to be able to route traffic to your ITSP using all the information given to you
1. Configure CUBE for media flow through. In this Mode CUBE acts as a true B2BUA. Advantages you get include address hiding and security becaue CUBE terminates and re-originate both signalling and Media. In this mode CUBE becomes a point of demarcation from th external world.
2. Configure CUCM to use Delayed Offer and CUBE to convert delayed offer to ealry offer...This prevents the need for you to use MTP to send Early offer on CUCM
voice service voip
early-offer forced
3. Configure DTMF signalling method on sip trunk to "No preference" This setting allows Unified CM to make an optimal decision for DTMF and to minimize MTP allocation.
4. Configure your CUBE to meet the requirements of your ITSP. Ask if they have configuration templates or any specific configuration they like you to use. This will save you time troubleshooting. Most of them dont use the default port 5060 because of security, confirm with your proivider what ports they use.
voice service voip
allow-connections sip to sip
sip
early-offer forced
header-passing
error-passthru
5. Use SIP to SIP...Use end to end sip. CUCM---sip---CUBE--sip----ITSP
6. Create a Trusted list of IP addresses on your CUBE is your CUBE IOS is 15.1 .2(T) and above.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
This is imprtant because sometimes your ITSP will send you a single ip address for signalling and will then send media on a different IP adress. So get all the IP address your ITSP is using and add them to the trust list as shown above
7. Configure your inbound and outbound dial-peer approriately
Inbound Dial-Peer for calls from CUCM to CUBE (CUCM sending 9 +all digits dialled to CUBE)
dial-peer voice 100 voip
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to CUCM (SP will be sending 10 digits inbound)
dial-peer voice 200 voip
description *** Outbound LAN side dial-peer ***
destination-pattern [2-9].........
session protocol sipv2
session target ipv4:
codec g711ulaw
dtmf-relay rtp-nte
Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy/load balancing
Inbound Dial-Peer for calls from SP to CUBE
dial-peer voice 100 voip
description *** Inbound WAN side dial-peer ***------------------(catch-all for all inbound PSTN calls)
incoming called-number [2-9].........
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to SP
dial-peer voice 200 voip
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4::XXXX (where XXXX is the port number your provider is using if different from 5060)
codec g711ulaw
dtmf-relay rtp-nte
8. SIP Normalization:
You may need to configure sip normnalization to modify sip headers, CLI etc. A good example is during call forwarding. You may need to change your diversion headers to match the CLI your provider is expecting.. if your call forwarding is failing during testing this may be the reason..We can help you with this.
9. Media Resources
Plan your solution properly. Consider if you will need Xcoders, MTP, Conference bridge etc. You may avoid the need for xcoders if you confure your regions properly and use voice class codecs on your sip profiles. It is important to know if there are any endpoints in your network that do not support dtmf relay rtp-nte. You can avoid the use of MTP if you configure your dial-peers to have multiple dtmf types for thos phones that do not support rtp-nte
e.g
dial-peer voice 1 voip
session protocol sipv2
dtmf-relay rtp-nte digit-drop sip-kpml (if your phones support kpml..then this will be used)
If in your environment you will need to do xcoding or CFB then ensure you have PVDMS
.10.FAX
If you have FAX in your network, determine what fax protocol your sip provider supports. Dont assume. Ask them and confirm in writing what they support. I have seen legal cases because of fax failures over sip trunks
Configure your FAX devices in seperate device pools and use porefix to route calls using G711 only. Even if you are using T38, ensure your fax use G711 to establish the voice calls
Finally
11. Have a detailed and carefully planned TEST Plan. Test the FF:
Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs (if supported by provider)
Outbound calls to information and emergency services
Caller ID and Calling Name Presentation
Supplementary services like Call Hold, Resume, Call Forward & Transfer
DTMF Tests
Fax calls – T.38, modem pass-through--whichever one you decide to use
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"

No "sip domain name" field in trunk configuration

Hello,
A customer' s UC320 throw a "sip 484" error,
Sep 22 16:10:44 UC320W user.debug voice: SIP/2.0 484 Address Incomplete
I think the problem is the lack of the "sip domain name" field in the trunk configuration as i can see one here;
https://supportforums.cisco.com/servlet/JiveServlet/showImage/102-15041-40-10
https://supportforums.cisco.com/docs/DOC-15041#UC_320_SIP_Trunk_Configuration
See attached the screenshot of the state of the trun, it misses his domain after the @
The UC320 is upgraded to 2.2.2 .

The case number is; 623254153
This is what the provider can see, (they say they cant handle "notify" requests);
U 2012/09/21 14:18:28.183877 IP.IP.IP.IP:5060 -> ip.ip.ip.ip:5060
NOTIFY sip:sip.myprovider.net SIP/2.0.
Via: SIP/2.0/UDP IP.IP.IP.IP:5060;branch=z9hG4bK-d56a8b2a.
From: "NIT" [email protected]>;tag=d78ae5c25878b8eco5.
To: .
Call-ID: [email protected]
CSeq: 211 NOTIFY.
Max-Forwards: 70.
Contact: "NIT" [email protected]:5060;ref=MYSIPID>.
Event: keep-alive.
User-Agent: Cisco/UC320W-2.2.2(1).
Allow-Events: talk, hold, conference, x-spa-cti.
Content-Length: 0.
U 2012/09/21 14:18:28.184243 ip.ip.ip.ip:5060 -> IP.IP.IP.IP:5060
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP IP.IP.IP.IP:5060;branch=z9hG4bK-d56a8b2a.
From: "NIT" [email protected]>;tag=d78ae5c25878b8eco5.
To: ;tag=696c4aa18cbb1138a6b8a116e3582f18.6a97.
Call-ID: [email protected]
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Server: Alphalink SIP Proxy 2.0.
Content-Length: 0.

Item category usage - how to configure

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Also are there any additional updates required to allow the usage code to be used ?
Thank you!

Hi
Usage indicator is something else. Just go through F1 help and see for what purpose we can use this field. What you are looking for is free goods configuration. This has been discussed so many times. Search with thext free goods and see if you find something. If you have some other requirement then come again with more explanation.
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Bridge-domain and trunk configuration on ES20 card.

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Hi Andrei,
Im having the same dilemna and im wondering if you were able to find a solution for this?
I need to support HSRP on a vlan interface with bridging over port channel bundled interface between routers.
im trying this but i cant seem to get this to work.
----R2----
interface Vlan10
ip vrf forwarding BOB
ip address 10.1.1.2 255.255.255.248
standby 1 ip 10.1.1.1
standby 1 priority 110
standby 1 preempt
end
interface Port-channel1
no ip address
service instance 10 ethernet
encapsulation dot1q 10
bridge-domain 10
--- R2----
interface Vlan10
ip vrf forwarding BOB
ip address 10.1.1.3 255.255.255.248
standby 1 ip 10.1.1.1
end
interface Port-channel1
no ip address
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encapsulation dot1q 10
bridge-domain 10
BR//
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Search for configuration usage in other configuration

We need to eliminate material groups (MM) in our environment. In order to check where is used the material group ZEB110000 (example) in ABAP programs, we used the standard transaction EWK1.
Now, we need to search for usage of material groups in other customizing in SAP. For example, material group customizing is used as a parameter to customize Characteristic Deviation from COPA (IMG Transaction KEDR)
The where used icon shows more than 400 tables with relationship to material groups, so, there would exist another alternative to do this.
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Hi,
This forum is specific to SAP Master Data Management. Kindly post your query in the below forum to get it solved quickly.
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Trunk configuration (VTP) between 3750 and 2811

Hi,
I have a stack of 6 Cisco Catalyst 3750G with 6 VLANs acting as a VTP server and want to establish a trunk link to my Cisco 2811 router equiped with a switching module 16 ports. Can I set the 2811 as a VTP client to propagate the VLAN info from the 3750 and configure one of the port on the switching module of the 2811 as a trunk encapsulated DOT1q and will I be able to route to the outside world.
Thank you

I beleive the switching cards support vtp and that should be possible.
Table 2. Features Supported on Cisco EtherSwitch Modules
EtherSwitch Network Modules (NM-16ESW and NMD-36ESW)
EtherSwitch HWICs (HWIC-4ESW and HWICD-9ESW)
EtherSwitch Service Modules (NME-16ES-1G, NME-16ES-1G-P, NME-X-23ES-1G, NME-X-23ES-1G-P, NME-XD-48ES-2S-P, and NME-XD-24ES-1S-P)
VLANs
Multiple VLANs per Switch
Yes (varies by chassis; maximum of 64 on Cisco 3845)
Yes (maximum of 15 on any chassis)
Yes (maximum of 1,024 per switch or stack)
VLANs in 4,000 Range
No
No
Yes
IEEE 802.1Q Tagged and Untagged VLANs
Yes
Yes
Yes (802.1Q and Inter-Switch Link [ISL])
VLAN Trunking Protocol (VTP) Support for Client, Server, and Transparent Modes
Yes
Yes
Yes

Apache Environment variables usage for dispatcher configurations

I have tried using apache environment variables within dispatcher.any files which doesn't seem to work. Is it feasible?

Hi Yogesh,
I want to carry forward certain environment variables which we are using all virtual hosts configured in our apache web server.
e.g., #1: Virtual hosts used in dispatcher will be same as the ServerName used in our Apache Virtualhost configuration
e.g.. #2: Using environment variables to identify custom docroot in our web cluster based on VirtualHost configuration.
Any thoughts?
Thanks
Suresh

MPLS TRUNK CONFIGURATION on TWO EDGE

Hi
Actually we have a network operate VRF on two EDGE (ASR9000) the diagram is this:
we try to configurate a MPLS conection between ASR (PE-1) and ASR (PE-2) try to use MPLS LDP and use a VRF OAM between this devices but the comunication is not possible
MPLS LDP is the option? or L2VPN or EoMPLS for this connection?
the actually configuration is:
ASR-2
mpls ldp
router-id 172.16.14.1
discovery hello holdtime 30
discovery hello interval 10
graceful-restart
explicit-null
interface Bundle-Ether100
ASR-1
mpls ldp
router-id 172.16.14.2
discovery hello holdtime 30
discovery hello interval 10
graceful-restart
explicit-null
interface Bundle-Ether100
but the VRF OAM only configurated between PE-1 and PE-2 is not neighbord
We don´t know if we are using the correct concept to connect the devices, can help us
thanks
Best Regards

Harold, thanks for your comments
we are making change for your comments and the final diagrame is:
on ASR9K - PE-1 we have configurated VRF, IGP and Conectivity for BUNDLE-Ethe 100 conectivity
ASR9K (PE-1):
vrf OAM
address-family ipv4 unicast
import route-policy pass-all
import route-target
   64518:64518
export route-policy pass-all
export route-target
   64518:64518
interface Bundle-Ether100
ipv4 address 172.16.14.1 255.255.255.252
interface Loopback10
vrf OAM
ipv4 address 172.16.162.1 255.255.255.255
router ospf 100
router-id 172.16.14.1
mpls ldp sync
mpls ldp auto-config
area 0
interface Bundle-Ether100
mpls ldp
router-id 172.16.14.1
interface Bundle-Ether100
ASR9K (PE-2):
vrf OAM
address-family ipv4 unicast
import route-policy pass-all
import route-target
   64518:64518
export route-policy pass-all
export route-target
   64518:64518
interface Bundle-Ether100
ipv4 address 172.16.14.2 255.255.255.252
interface Loopback10
vrf OAM
ipv4 address 172.16.162.2 255.255.255.255
router ospf 100
router-id 172.16.14.2
mpls ldp sync
mpls ldp auto-config
area 0
interface Bundle-Ether100
mpls ldp
router-id 172.16.14.2
interface Bundle-Ether100
when we verifying the MPLS neighbor is UP
RP/0/RSP0/CPU0:ED_MEX_1#sho mpls ldp neighbor
Wed May 22 18:29:03.496 UTC
Peer LDP Identifier: 172.16.14.2:0
TCP connection: 172.16.14.2:39527 - 172.16.14.1:646
Graceful Restart: No
Session Holdtime: 180 sec
State: Oper; Msgs sent/rcvd: 25/25; Downstream-Unsolicited
Up time: 00:18:46
LDP Discovery Sources:
    Bundle-Ether100
Addresses bound to this peer:
    172.16.14.2
RP/0/RSP0/CPU0:ED_MEX_2#sho mpls ldp neighbor
Wed May 22 16:24:53.223 UTC
Peer LDP Identifier: 172.16.14.1:0
TCP connection: 172.16.14.1:646 - 172.16.14.2:39527
Graceful Restart: No
Session Holdtime: 180 sec
State: Oper; Msgs sent/rcvd: 26/26; Downstream-Unsolicited
Up time: 00:19:19
LDP Discovery Sources:
    Bundle-Ether100
Addresses bound to this peer:
    172.16.14.1
on OSPF 100 the neighbor is UP
RP/0/RSP0/CPU0:ED_MEX_2#sho ospf neighbor
Wed May 22 16:26:15.169 UTC
* Indicates MADJ interface
Neighbors for OSPF 100
Neighbor ID     Pri   State           Dead Time   Address         Interface
172.16.14.1     1     FULL/BDR        00:00:31    172.16.14.1   Bundle-Ether100
    Neighbor is up for 00:54:34
Total neighbor count: 1
RP/0/RSP0/CPU0:ED_MEX_1#sho ospf neighbor
Wed May 22 18:31:18.614 UTC
* Indicates MADJ interface
Neighbors for OSPF 100
Neighbor ID     Pri   State           Dead Time   Address         Interface
172.16.14.2     1     FULL/DR         00:00:36    172.16.14.2     Bundle-Ether100
    Neighbor is up for 00:54:59
Total neighbor count: 1
but when try to send a PING from Loopback 10 from ASR 1 to ASR 2 ocurre this one and viceverse
RP/0/RSP0/CPU0:ED_MEX_1#ping vrf OAM 172.16.162.1
Wed May 22 18:32:54.046 UTC
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 172.16.162.1, timeout is 2 seconds:
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/1 ms
RP/0/RSP0/CPU0:ED_MEX_1#ping vrf OAM 172.16.162.2
Wed May 22 18:32:57.794 UTC
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 172.16.162.2, timeout is 2 seconds:
UUUUU
Success rate is 0 percent (0/5)
the routing table for OAM on ASR-1 is:
RP/0/RSP0/CPU0:ED_MEX_1#sho route vrf OAM
Wed May 22 18:33:59.485 UTC
Codes: C - connected, S - static, R - RIP, B - BGP
       D - EIGRP, EX - EIGRP external, O - OSPF, IA - OSPF inter area
       N1 - OSPF NSSA external type 1, N2 - OSPF NSSA external type 2
       E1 - OSPF external type 1, E2 - OSPF external type 2, E - EGP
       i - ISIS, L1 - IS-IS level-1, L2 - IS-IS level-2
       ia - IS-IS inter area, su - IS-IS summary null, * - candidate default
       U - per-user static route, o - ODR, L - local, G - DAGR
       A - access/subscriber, - FRR Backup path
Gateway of last resort is not set
L    172.16.162.1/32 is directly connected, 00:34:13, Loopback10
for ASR-2
RP/0/RSP0/CPU0:ED_MEX_2#sho route vrf OAM
Wed May 22 16:30:23.400 UTC
Codes: C - connected, S - static, R - RIP, B - BGP
       D - EIGRP, EX - EIGRP external, O - OSPF, IA - OSPF inter area
       N1 - OSPF NSSA external type 1, N2 - OSPF NSSA external type 2
       E1 - OSPF external type 1, E2 - OSPF external type 2, E - EGP
       i - ISIS, L1 - IS-IS level-1, L2 - IS-IS level-2
       ia - IS-IS inter area, su - IS-IS summary null, * - candidate default
       U - per-user static route, o - ODR, L - local, G - DAGR
       A - access/subscriber, - FRR Backup path
Gateway of last resort is not set
L    172.16.162.2/32 is directly connected, 00:34:47, Loopback10
i don´t know if need something on OSPF
Best Regards

Configuring Level3 SIP trunk with Lync 2013

Hi, I ran into some issues trying to configure SIP trunk from Level 3 and I was hoping someone here can help. We have our mediation server collocated with FE and SIP traffic goes from public IP, port 5060 via NAT, to local IP on FE, port 5060.
Level 3 provided us with one signaling IP and two RTP IPs.
I tried multiple trunk configuration settings and I can see that when I'm placing a call from Lync to an outside number I'm getting INVITE from Level 3 signaling IP, the session is established, phone rings, but there is no audio on either side. There's also
a METHOD NOT ALLOWED message coming from them, which doesn't tell me much about what's happening.
If I call to a Level 3 DID (assigned to my Lync user account) there's also INVITE from their side, but later I receive a CANCEL from them due to idle session. The phone never rings.
Questions:
1) Does anyone have Level 3 SIP trunks configured and can share their Get-TrunkConfiguration settings? What settings should I have for encryption, refer, sessionTimer / RTCP, and others? Level 3 refuses to provide any additional information besides IPs.
2) Do I understand this correctly that when configuring PSTN gateways in topology, one of the RTP IPs should be entered in the "alternate media IP" field? We have SIP trunks from another provider (which work fine), and they only use one IP
for everything, so I don't have any experience configuring separate SIP and media IPs with Lync.
Thanks, and let me know if I should provide additional info.

Hi,
On Lync topology PSTN gateways interface, please check if you enter gateway listening port 5060 and enable TCP option.
Please also check if you enable refer support on Lync Server Control Panel, if you enable it please uncheck it.
You can compare the trunk configuration for Level 3 in the part “Sample Trunk Configuration for Level 3” in the link below with yours’, it is for Lync server 2010 but similar for Lync server 2013:
http://blogs.technet.com/b/nexthop/archive/2013/04/10/configuring-lync-2010-server-to-work-with-level-3-sip-trunking-services.aspx
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

How to Integrate Microsoft Lync 2010, Asterisk, and a sip trunk.

Dear Friends.
i need you to assist me to step my new project
Objective:
Setup Asterisk
to Configure a SIP trunk between Asterisk and the SIP provider of my choice
Integrate Lync Server 2010 with Asterisk
Configure a dial plan
Configuring Voice Polices, PSTN Usage Records, and Voice Routes.
To be able to make international
local call to any mobile extension or same number range
This is a new project to me can anyone please simply assist me step by step ?
Thanks
Greenman

Hi GreeMann, Which Flavor of Asterisk you are using ex: FreePBX, Elastix, AsteriskNow.
You can use any of them most of the configuration will be similar.
To configure the SIP Trunk of service provider in asterisk check this
http://wiki.freepbx.org/display/ST/Setting+up+SIPStation+manually+in+FreePBX http://wiki.freepbx.org/display/F2/Trunk+Sample+Configurations
Here is my blog Step by step guide to Integrate asterisk ( Elastix) with Lync
http://mslyncforall.blogspot.in/2014/12/lync-2013-asterisk-pbx-integration.html
http://blogs.technet.com/b/rischwen/archive/2013/08/21/series-exchange-2013-and-lync-2013-integration-with-asterisknow-pbx-pt-1.aspx
Please let me know if you encounter any issues i am happy to help you.
Whenever you see a helpful reply, click on Vote As Helpful & click on Mark As Answer if a post answers your question.

Inter-Trunk not route incoming calls from out

Hi,
I setup one extra gateway where I try to route part of our calls. So far I have success to route internal calls into there, but when I'm making a test call from outside that ends into "number is not used" problem.
I have:
- Route ready, elsewere the internal calls are not working.
- PSTN usage, linked to the Route
- Trunk configuration where I have selected the PSTN usage
- Incoming numbers are coming in E164 format
I have also tested the "Test-CsInterTrunkRouting" and that gives "pass":
FirstMatchingRoute : Description=;NumberPattern=^\+358123654789;Name=Test
                     Gateway;SuppressCallerId=False;AlternateCallerId=
MatchingUsage      : Test PSTN Usage
MatchingRoutes     : {Description=;NumberPattern=^\+358123654789;Name=Test
                     Gateway;SuppressCallerId=False;AlternateCallerId=}
But still, when I made a call from outsited the OCSLogger shows that mediation server try to offer call to Front-End which says only: "SIP/2.0 404 Not Found" and then bye-bye.
What is the missing magic, which made the mediation server to see alternative route? I hope it is not required that mediation server must be collocated on the Front Ends, as that one I do not have.
Any good ideas?
ps.
I'm not sure does it matter, but my Lync gives "SIP/2.0 403 Forbidden" when there is coming call from extra gateway. But as the calls into there works, then I don't see why external calls should not also work.
Petri

Could it be even so, that intra-trunk routing requires consolidated mediation server? As the call is owned by the Mediation server (stanalone), and it is trying to offer that to FE. FE reply "does not exist". Because of the standalone Mediation
server does not have the call routing engine like FE have, the call is lost.
I started to think above as Lync users are able to call to that number. So FE is able to do the routing and get calls into the correct place.
I have to say also, I have read
Ken's blog about inter-trunk routing, I have to say that I'm not so sure what he means by this: "Fortunately, in most cases, adding PSTN usages to the trunk has no effect, since there is almost always a Lync user assigned to the incoming phone
numbers". Why to add additional routing for the numbers which are already inuse? I hope it is not required, that you need to have a users ID for each number you do the inter-trunk routing?
Petri

Lync 2013 as PSTN gateway for CUCM

Hello,
this is an oddity and not really an advertised feature with Lync, but:
we have a really small footprint of CUCM devices, mostly VTC units that would like to be able to call out to PSTN using Lync infrastracture. We have a fully featured deployment of Lync 2013 used company wide 2000+ users.
We created a SIP trunk between CUCM & Lync mediation server, we defined New IP/PSTN Gateway in topology and published it, configured SIP trunk, profiles, etc in CUCM. The purpose of this is to only dial outbound from the CUCM devices, the CUCM devices
do not have to be reached from outside or by Lync users.
We are able to place calls from a VTC endpoint registered to CUCM via Lync SIP trunk to all Lync users, but we are unable to dial out to PSTN.
Most configuration refer to configuring Lync to use CUCM for PSTN connectivity not viceversa. Is there any documentation on achieving this?
Call flow would be like this.
VTC Endpoint > CUCM ----SIP Trunk----> Lync mediation > PSTN
We already have PSTN access built in Lync, we just need to bridge between the CUCM direct SIP trunk and PSTN.
Thank you.

Hi,
To enable inter-trunk routing, associate and configure PSTN usage records to this trunk configuration. The PSTN usages associated to this trunk configuration will be applied for all incoming calls through the trunk that is not originating from a Lync endpoint.
More details:
https://technet.microsoft.com/en-us/library/gg425831.aspx
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

PSTN Usage in Trunk Configuration

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