ILBC calls via SIP Trunk with CUBE and CUCM7

hi there,
our SIP Provider offers the IBLC codec which promises to provide better quality compard to G.729.
I'm using this scenario:
IP-Phone(G711) --- CUCM7 --- (SIP-Trunk1) --- CUBE --- (SIP-Trunk2) --- Provider
Everything workes unless I'm configuring IBLC at the provider and on trunk2.
I have the CUBE router acting as a trancoding device and also specified IBLC as codec to be handled.
SIP trunk 2 was placed in a region with IBLC as codec.
On the trunk configuration in CUCM the media ressource group with XCODE capability is configured
Transcoding workes between two IP Phones in different regions with different codecs within the intranet.
Unfortunately the CUBE router doesn't seem to use the transcoder to change internal G711u calls into IBLC codec
so calls are blocked by the CUBE device:
deb ccsip calls
for incoming call:
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4AE7AC98
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0237892992
Called Number            : 036677725231
Source IP Address (Sig ): 10.100.100.50
Destn SIP Req Addr:Port : <IP SIP Provicer>
Destn SIP Resp Addr:Port : <IP SIP Provicer>:5060
Destination Name         : <IP SIP Provicer>
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : ilbc
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): <IP CUBE>
Source IP Port    (Media): 0
Destn IP Address (Media): <IP SIP Provicer>
Destn IP Port    (Media): 22022
Orig Destn IP Address:Port (Media): [ - ]:0
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 65
Disconnect Cause (SIP)   : 488
(Output lookes similar to outgoing calls)
I set up ccm on cube and assigned dsp ressources without success:
Here are the relevant configuration parts:
voice class codec 1
codec preference 1 iblc
voice service voip
address-hiding
allow-connections sip to sip
allow-connections h323 to sip
allow-connections sip to h323
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
h323
sip
header-passing error-passthru
no update-callerid
midcall-signaling passthru
privacy-policy passthru
voice-card 0
dspfarm
dsp services dspfarm
dial-peer voice 40991 voip
description *** Incoming from SIP-Provider
destination-pattern 03667772523.%
session protocol sipv2
session target ipv4:<IP_of_CUCM>
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
ip qos dscp cs5 media
ip qos dscp cs5 signaling
sccp local GigabitEthernet0/0
sccp ccm 10.100.100.50 identifier 11 version 4.1
sccp
sccp ccm group 11
description *** lokaler CCM fuer Codec-Konvertierung von SIP/DUS.NET
associate ccm 11 priority 1
associate profile 21 register DE_WGT_MTP02
dspfarm profile 21 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec ilbc
maximum sessions 10
associate application SCCP
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 10
sdspfarm tag 1 DE_WGT_MTP02
max-ephones 30
max-dn 30
ip source-address 10.100.100.50 port 2000
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 14 2010 02:10:34
sh sccp
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
        IPv4 Address: 10.100.100.50
        Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.100.100.50, Port Number: 2000
                Priority: N/A, Version: 4.1, Identifier: 11
                Trustpoint: N/A
Call Manager: 10.1.1.55, Port Number: 2000
                Priority: N/A, Version: 7.0, Identifier: 10
                Trustpoint: N/A
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.100.100.50, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 21
Reported Max Streams: 20, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
sh dspfarm dsp all
SLOT DSP VERSION STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED
0    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
0    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
Thanks in advance,
David

Hi there,
Just wondering whether you ever got this resolved? I seem to have a very similiar problem.
Regards
Karen

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Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:48:27 GMT
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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,X-cisco-original-called
Call-Info: <sip:192.168.106.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
Session-Expires: 1800
P-Asserted-Identity: "" <sip:[email protected]>
Remote-Party-ID: "" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 390
v=0
o=CiscoSystemsCCM-SIP 4037968 1 IN IP4 192.168.106.11
s=SIP Call
c=IN IP4 10.139.64.171
b=TIAS:64000
b=AS:64
t=0 0
m=audio 30688 RTP/AVP 0 8 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
INVITE sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
Remote-Party-ID: "" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:40:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0520523008-0000065536-0000067523-0191539392
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1387402810
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 348
v=0
o=CiscoSystemsSIP-GW-UserAgent 4778 3356 IN IP4 10.139.64.52
s=SIP Call
c=IN IP4 10.139.64.52
t=0 0
m=audio 23372 RTP/AVP 0 8 116 18 101
c=IN IP4 10.139.64.52
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079120: Dec 18 2013 16:40:10.008: //314738/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079121: Dec 18 2013 16:40:10.080: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079122: Dec 18 2013 16:40:11.176: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
Supported:
Contact: <sip:[email protected]:5073;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:40:09 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=no;privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079128: Dec 18 2013 16:40:12.384: //314739/1F068D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1387402810
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
SIP/2.0 480 Temporarily Not Available
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:40:09 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=18
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079146: Dec 18 2013 16:40:12.388: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
From: "" <sip:[email protected]>;tag=78FC5414-198D
To: <sip:[email protected]>;tag=182903799-1387403308449
Date: Wed, 18 Dec 2013 21:40:10 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
079147: Dec 18 2013 16:40:12.404: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
To: <sip:[email protected]>;tag=78FC58A8-1B6B
Date: Wed, 18 Dec 2013 21:48:27 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
dial-peer voice 9100 voip
description inboubd dial-peer for outgoing calls from CUCM (11D)
preference 1
session protocol sipv2
incoming called-number ^1..........$
voice-class codec 10
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
outbound DP
dial-peer voice 8100 voip
description outbound dial-peer for outgoing calls to Verizon (11D)
destination-pattern ^1..........$
session protocol sipv2
session target sip-server
voice-class codec 10
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class codec 10
codec preference 1 transparent
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g722-64

I created the new voice class and mapped it to the outgoing dial-peer 8100. The call was then successful.
See new voice class:
#sh run | be voice class codec 11
voice class codec 11
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
See revised dial-peer 8100:
dial-peer voice 8100 voip
description outbound dial-peer for outgoing calls to Verizon (11D)
destination-pattern ^1..........$
session protocol sipv2
session target sip-server
voice-class codec 11
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte digit-drop
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
My only remaining question is why did the CUBE invite NOT include the m line for g729r8?
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See the ccapi inout snippet showing the hit with dial-peer 8100:
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
080927: Dec 19 2013 15:27:57.810: //316459/32C4F8800001/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=8100, Params=0x2B912E08, Progress Indication=NULL(0)
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See the debug ccsip messages output showing original CUCM invite received by CUBE with 5 a line references:
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
080907: Dec 19 2013 15:27:57.806: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6d715c9c6ad1
From: "XXXXXXXXXX" ;tag=4077346~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65761788
To:
Date: Thu, 19 Dec 2013 20:36:14 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,X-cisco-original-called
Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
Session-Expires: 1800
P-Asserted-Identity: "XXXXXXXXXX"
Remote-Party-ID: "XXXXXXX" ;party=calling;screen=yes;privacy=off
Contact:
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 464
v=0
o=CiscoSystemsCCM-SIP 4077346 1 IN IP4 192.168.106.11
s=SIP Call
c=IN IP4 10.139.64.52
b=TIAS:64000
b=AS:64
t=0 0
m=audio 26738 RTP/AVP 0 8 116 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:116 iLBC/8000
a=ptime:30
a=maxptime:60
a=fmtp:116 mode=30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
See ccsip messages output showing CUBE sending invite to Verizon:
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Sent:
INVITE sip:[email protected]:5073 SIP/2.0
Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK63F9C611
Remote-Party-ID: "David Callahan" ;party=calling;screen=yes;privacy=off
From: "David Callahan" ;tag=7DE0957C-1CAB
To:
Date: Thu, 19 Dec 2013 20:27:57 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0851769472-0000065536-0000068412-0191539392
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1387484877
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6966 4178 IN IP4 10.139.64.52
s=SIP Call
c=IN IP4 10.139.64.52
t=0 0
m=audio 32502 RTP/AVP 0 8 101
c=IN IP4 10.139.64.52
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

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

Third Party Phone over SIP Trunk with CUCM 9.x

Hi all,
I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
Cisco Phone: INVITE sip.60xxxx%23@ipadress
Third Party SIP Phone: INVITE sip:[email protected]
It seems the Cisco phones gets some extra configured the Third Party ones dont...
Thanks in advance for any help.
//Per

Thanks for the answer
Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty. The termination Cause Code is that the number requested is Unallocated/Unassigned..
In other words something is missing to get CUCM to strip 60 from the Polycom phones dialed number and send it towards the SIP trunk like it does when the Cisco phones call it.
Unfortunatley i dont have the meens to attach the trace...
Thanks again for any help/advice
With regards, Per.

Lync 2013 with SIP trunk with panasonic kx-tde200

Hi
My company has installed a panasonic ip-pbx kx-tde for multiline with 100 number range for telephone service.
Now my company is going to replace multiline by sip trunk . It will still work with Panasonic pbx box just need to reprogramme to be able to connected to the sip proxy which is managed by internet service provider.
For this scenario , would Lync 2013 voice work if I just add PSTN gateway which is the ip of panasonic pbx address to the frontend in topology ? Or I may need a mediation server as a must requirement to make lync voice work?
Thanks
WenFei

Media bypass allow a call to basically skip the mediation server once it's established and go directly from gateway (in this case the PBX) and the endpoint (the telephone handset or Lync client) More information here: http://technet.microsoft.com/en-us/library/gg398719.aspx
By having this (if your PBX supports it) you reduce the load on the mediation servers. Before you go too far down the road also make sure that your PBX supports SIP trunks that are SIP over TCP (as Lync doesn't work with SIP over UDP)
Sort of, the easiest way is to add the .com as an additional SIP domain in Topology builder, you will need to create DNS records for it (both internal and external) and you will need to reissue the certs with additional SANs to support the second domain.
YOu will also need to update all the users to use the new suffix of xxx.com. So it's not a small task.
If this helped you please click "Vote As Helpful" if it answered your question please click "Mark As Answer" | Blog
www.lynced.com.au | Twitter
@imlynced

PMF to allow outgoing calls through SIP Trunk Without Registering

Hello,
I have an intermitant issue with one of our UC320W's running 2.3.2(6) firmware. The customers VOIP SIP trunk becomes unregistered for periods of time, stopping incoming and outgoing calls. Once unregistered it takes quite a while to rergister. Our service provider has informed us that the re-register period is the cause and we should try and shorten it, so first question is there a way to do this, also what is the re-register retry window in the first place?
I have an analogue line that can receive calls only so I have made this the fallover number with the VOIP provider, that gives a little releife for incoming calls, but not outgoing. I beleive in other phone systems a SIP trunk does not need to be registered to make an outgoing call, and it is usually an option to say only make outgoing calls if the SIP trunk is registered. I cannot find that option anywhere to deselect it, is there a PMF I could apply to allow outgoing calls without registering?
Thank you,
Tony

Hi Tony,
Please install the SIP_Trunk_Register_Timer.pmf at status->Devices->Alter PMFs in configure utility. Please remember to apply the configuration afterwards. This PMF can let user to select the re-register period. You can find the PMF at https://supportforums.cisco.com/docs/DOC-16301
Regards,
Wendy Yang

Not receiving the 486 message from CUCM to Genesys via SIP trunk.

I have setup where Genesys is used along with CUCM 9.1
Below is the snapshot how it will look for call flow.
PRI----V.G----CUCM---SIP trunk (created in CUCM)-----Geneys server.
Query here is for outbound call from SIP softphone to PSTN, where if the PSTN user cancel the call.
the SIP phone is still assuming the call is continuing and after 40 sec its getting disconnected.
after looking in to the sip traces... it looks like that SIP trunk from cucm is not sending the user busy message 486....
(checked in V.G and its giving user busy)...but in the CUCM its not getting sent to the genesys...
After some time in genesys server itself send the 480 Temporarily Not Available message...
I assume I should get the 486 message from CUCM to genesys when the PSTN party disconnect the call without answering.
Please assist.

From logs what i can see is after one min call legs stops transmitting and receiving packets.
You certainly need to check this Genesys support for this behavior , as far as I know there is no problem either with the CUCM or with VG.
1477 : 1515 22820290ms.1 +0 pid:0 Originate connecting
dur 00:01:15 tx:3765/602400 rx:3387/541760
IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off
1477 : 1514 22820290ms.2 +0 pid:0 Originate active
dur 00:01:16 tx:3387/568856 rx:3838/614080
Tele 0/3/0:15 (1514) [0/3/0.31] tx:76760/76760/0ms g711ulaw noise:-68 acom:3 i/0:-64/-62 dBm
1477 : 1515 22820290ms.1 +0 pid:0 Originate connecting
dur 00:01:26 tx:4330/692800 rx:3387/541760
IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off
1477 : 1514 22820290ms.2 +0 pid:0 Originate active
dur 00:01:30 tx:3387/568856 rx:4515/722400
Tele 0/3/0:15 (1514) [0/3/0.31] tx:90290/90290/0ms g711ulaw noise:-68 acom:3 i/0:-67/-61 dBm
Rate all the helpful post.
Thanks
Manish

SIP REFER with UCCE and CUPS SIP Proxy

I am running UCCE 8.0.1 with CVP and CUPS as the SIP proxy. I am looking to transfer calls to PSTN and release from CVP to free CVP ports. I am using the rfxxxxxxx method in the ICM script, which seems to work fine from CVP. However the SIP proxy send an Invite to our SBC instead of a REFER. Is there a way to configure SIP Proxy to send the REFER instead of the invite? I would like to release the call from our SBC as well.
The other idea was to insert a custom header in CVP I could then pull out at the SBC and replace with a REFER. Does anyone have any links to documentation on this?
Thanks
TC

how about check "enable send calls to originator" for the refer label routing in CVP? this would bypass proxy.

Trunk with WLC and 1400BR problem

hi everybody,
i have the next proble, i hope someone can help me
Actually I wrok with a 1522 Mesh Network,1130 LWAPP and Bridge 1400 point to point. 1522 and 1130 are asociated with WLC.
I have a WLC4402 (4.1.192.22M (Mesh)image) this wlc is conected via trunk to Sw3750 ex:
interface GigabitEthernet1/0/1
switchport trunk encapsulation dot1q
switchport mode trunk
RAP1 is connected to the sameSw3750 ex:
interface FastEthernet1/0/23
description RAP1
switchport access vlan 10
**(VLAN 10 is Mgmt)**
AP1(1130) is connected to the same Sw3750 ex:
interface FastEthernet1/0/1
description AP1
switchport access vlan 10
The 1410BR Root is connected via trunk to same Sw3750 ex:
interface FastEthernet1/0/19
description BR-1400R
switchport trunk encapsulation dot1q
switchport trunk native vlan 10
switchport mode trunk
In the other point is the Non-Root connected to a Sw2960 ex:
interface GigabitEthernet1/0/1
switchport trunk native vlan 10
switchport mode trunk
AP2(1130) connected to the same Sw2960 ex:
interface fa0/23
descriptipon AP2
switchport access vlan 10
The network is work fine, Mesh UP (RAP and MAPs), and 1130 too.I connected the 1400 Bridge point after the Mesh is up, and the link between Root and Non Root is UP
Now, when the Sw3750 goes down or reboot,the RAP and AP1(1130) can't associated to WLC. The ports of RAP and 1130 are down and up many times, so can't associated to a WLC. Only the Bridge point 1400 Root and Non-root are UP, and the AP2(1130) in the other side can associated to the WLC.
When shutdown the port of the Root Bridge, Now the RAP1 and AP1(1130) can associated to the WLC and the Mesh Net is UP. Then no shutdown the Root Bridge port and the link between Bridges are UP, AP2(1130) up to the controller too.
But after several minutes the Bridge down, and the event log in the Root is:Interface Dot11Radio0 Radio transmit power out of range.
So i have this problems
1) Trunks between WLC and 1400 BR
2) Bridge conectivity range.
Regards
Antonio

The Outdoor Bridge Range Calculation Utility uses parameters that include regulatory domain, device type, data rate, antenna gain, and a few others as inputs.
You can avoid connectivity problems with the Outdoor Bridge Calculation Utility, as this tool helps you to predict the distance between devices. In a wireless environment without a tool like this, you cannot predict the distance between the bridges, the height at which you must place the antennas for maximum throughput, and other variables. This utility also helps you decide on the type of antenna that you must use in order to cover the distance between the bridges.

ILBC calls via SIP Trunk with CUBE and CUCM7

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