Delay Outbound through SIP Trunk

Hi there,
When calling Outbound through a SIP trunk takes about 20 seconds. Inbound calls are going fine. I tried the following scenario's:
IP Phone > CUCM > SIP Trunk > CUBE > SIP Provider
IP Phone > CUCM > H323 Gateway > CUBE > SIP Provider
I'm attachting CCSIP logs and if you look at the timestamps, you can see there is a delay of around 10 seconds.
Any suggestions will be highly appreciated.
thanks.

Hi Brian,
A few weeks back I did same kind of configuration with another customer (with the same SIP Provider) and I don't have this probleem there. I did the same on CUCM and also on the CUBE (same version of IOS and almost same configuration).
!dial-peer voice 1010 voip
destination-pattern T
progress_ind alert enable 8
session protocol sipv2
session target dns:pbx.signet.nl
incoming called-number T
dtmf-relay rtp-nte cisco-rtp
codec g711ulaw
no vad
dial-peer voice 1000 voip
destination-pattern 717470101
session target ipv4:192.168.1.250
incoming called-number 717470101
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 1020 voip
destination-pattern 8886401..
progress_ind alert enable 8
session target ipv4:192.168.1.250
incoming called-number 8886401..
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad

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    Remote-Party-ID: "" <sip:[email protected]>;party=calling;screen=yes;privacy=off
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    Date: Wed, 18 Dec 2013 21:40:10 GMT
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    Timestamp: 1387402810
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    From: "" <sip:[email protected]>;tag=4037968~a812c08a-f7f0-43a7-a92c-e5ac2a38867c-65745615
    To: <sip:[email protected]>
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    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:
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    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]>
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    CSeq: 101 INVITE
    Timestamp: 1387402810
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    Via: SIP/2.0/UDP 10.139.64.52:5060;branch=z9hG4bK636B313D7
    From: "" <sip:[email protected]>;tag=78FC5414-198D
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    +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
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    Via: SIP/2.0/TCP 192.168.106.11:5060;branch=z9hG4bK6c4bb65f140c8
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    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

  • SIP Trunk Question - Outbound Calls Fail

    Hi Folks,
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    *Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo: 
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    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 1528xxxx
    Called Number            : 909
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    Destination Name         : 10.200.7.157
    *Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPIMediaCallInfo: 
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    Media Stream             : 1
    Negotiated Codec         : No Codec   
    Negotiated Codec Bytes   : 0
    Nego. Codec payload      : 255 (tx), 255 (rx)
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    Dtmf-relay Payload       : 0 (tx), 0 (rx)
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  • SIP Trunk - No voice with Single Number Reach

    Hi Community.
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     vlan 1
     authentication open
    dot11 ssid cisco-voice
     vlan 100
     authentication open
    ip source-route
    ip cef
    ip dhcp relay information trust-all
    ip dhcp excluded-address 10.1.1.1 10.1.1.9
    ip dhcp excluded-address 10.1.1.241 10.1.1.255
    ip dhcp pool phone
     network 10.1.1.0 255.255.255.0
     default-router 10.1.1.1
     option 150 ip 10.1.1.1
    ip domain name site1.365873.trk.ipvoip.ch
    ip name-server 8.8.8.8
    ip inspect WAAS flush-timeout 10
    ip inspect name SDM_LOW dns
    ip inspect name SDM_LOW ftp
    ip inspect name SDM_LOW h323
    ip inspect name SDM_LOW https
    ip inspect name SDM_LOW icmp
    ip inspect name SDM_LOW imap
    ip inspect name SDM_LOW pop3
    ip inspect name SDM_LOW netshow
    ip inspect name SDM_LOW rcmd
    ip inspect name SDM_LOW realaudio
    ip inspect name SDM_LOW rtsp
    ip inspect name SDM_LOW esmtp
    ip inspect name SDM_LOW sqlnet
    ip inspect name SDM_LOW streamworks
    ip inspect name SDM_LOW tftp
    ip inspect name SDM_LOW tcp router-traffic
    ip inspect name SDM_LOW udp router-traffic
    ip inspect name SDM_LOW vdolive
    no ipv6 cef
    multilink bundle-name authenticated
    stcapp ccm-group 1
    stcapp
    isdn switch-type basic-net3
    voice call send-alert
    voice rtp send-recv
    voice service voip
     ip address trusted list
      ipv4 0.0.0.0 0.0.0.0
     allow-connections h323 to h323
     allow-connections h323 to sip
     allow-connections sip to h323
     allow-connections sip to sip
     supplementary-service h450.12
     no supplementary-service sip refer
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
     sip
      registrar server expires max 3600 min 3600
      localhost dns:site1.365873.trk.ipvoip.ch
      no update-callerid
    voice class codec 1
     codec preference 1 g711alaw
    voice register global
     mode cme
     source-address 10.1.1.1 port 5060
     load 9971 sip9971.9-2-2
     load 9951 sip9951.9-2-2
     load 8961 sip8961.9-2-2
     timezone 23
    voice source-group CCA_SIP_SOURCE_GROUP_CUE_CME
     access-list 2
     translation-profile incoming SIP_Incoming
    voice source-group CCA_SIP_SOURCE_GROUP_EXTERNAL
     access-list 3
    voice translation-rule 9
     rule 1 /0041449475090/ /90/
     rule 2 /0041449475091/ /91/
     rule 3 /0041449475092/ /92/
     rule 4 /0041449475093/ /93/
     rule 5 /0041449475094/ /94/
     rule 6 /0041449475095/ /95/
     rule 7 /0041449475096/ /96/
     rule 8 /0041449475097/ /97/
     rule 9 /0041449475098/ /98/
     rule 10 /0041449475099/ /99/
    voice translation-rule 410
     rule 1 /^0\(.*\)/ /\1/
     rule 15 /^..$/ /0041449475090/
    voice translation-rule 411
     rule 1 /^0\(.*\)/ /ABCD0\1/
    voice translation-rule 412
     rule 1 /^ABCD\(.*\)/ /\1/
    voice translation-rule 422
     rule 15 /^ABCD\(.*\)/ /\1/
    voice translation-rule 1000
     rule 1 /.*/ //
    voice translation-rule 1111
     rule 1 /^9\([1-9]\)$/ /004144947509\1/
     rule 15 /^..$/ /0041449475090/
    voice translation-rule 1112
     rule 1 /^0/ //
    voice translation-rule 2000
     rule 1 /0041449475098/ /98/
    voice translation-rule 2001
     rule 1 /0041449475097/ /97/
    voice translation-rule 2002
     rule 1 /^6/ //
    voice translation-rule 2222
    voice translation-profile AA_Profile
     translate called 2001
    voice translation-profile CALLER_ID_TRANSLATION_PROFILE
     translate calling 1111
    voice translation-profile CallBlocking
     translate called 2222
    voice translation-profile OUTGOING_TRANSLATION_PROFILE
     translate called 1112
    voice translation-profile PSTN_CallForwarding
     translate redirect-target 410
     translate redirect-called 410
    voice translation-profile PSTN_Outgoing
     translate calling 1111
     translate called 1112
     translate redirect-target 410
     translate redirect-called 410
    voice translation-profile SIP_Called_9
     translate calling 3265
     translate called 9
    voice translation-profile SIP_Incoming
     translate called 411
    voice translation-profile SIP_Passthrough
     translate called 412
    voice translation-profile SIP_Passthrough_CallBlocking
     translate called 422
    voice translation-profile VM_Profile
     translate called 2000
    voice translation-profile XFER_TO_VM_PROFILE
     translate redirect-called 2002
    voice translation-profile nondialable
     translate called 1000
    voice-card 0
     dspfarm
     dsp services dspfarm
    fax interface-type fax-mail
    license udi pid UC540W-BRI-K9 sn FGL163220SL
    archive
     log config
      logging enable
      logging size 600
      hidekeys
    username admin privilege 15 secret xxx
    username xxx password 0 ""
    username xxx password 0 ""
    ip tftp source-interface Loopback0
    bridge irb
    interface Loopback0
     description $FW_INSIDE$
     ip address 10.1.10.2 255.255.255.252
     ip access-group 101 in
     ip nat inside
     ip virtual-reassembly in
    interface FastEthernet0/0
     description $FW_OUTSIDE$
     no ip address
     ip inspect SDM_LOW out
     ip virtual-reassembly in
     ip verify unicast reverse-path
     load-interval 30
     shutdown
     duplex auto
     speed auto
    interface Integrated-Service-Engine0/0
     description cue is initialized with default IMAP group
     ip unnumbered Loopback0
     ip nat inside
     ip virtual-reassembly in
     service-module ip address 10.1.10.1 255.255.255.252
     service-module ip default-gateway 10.1.10.2
    interface FastEthernet0/1/0
     no ip address
     macro description cisco-desktop
     spanning-tree portfast
    interface FastEthernet0/1/1
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/2
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/3
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/4
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/5
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/6
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/7
     switchport voice vlan 100
     no ip address
     macro description cisco-phone
     spanning-tree portfast
    interface FastEthernet0/1/8
     no ip address
     macro description cisco-desktop
     spanning-tree portfast
    interface BRI0/1/0
     no ip address
     isdn switch-type basic-net3
     isdn point-to-point-setup
     isdn incoming-voice voice
     isdn sending-complete
     isdn static-tei 0
    interface BRI0/1/1
     no ip address
     shutdown
     isdn switch-type basic-net3
     isdn point-to-point-setup
     isdn incoming-voice voice
     isdn sending-complete
     isdn static-tei 0
    interface Dot11Radio0/5/0
     no ip address
     ssid cisco-data
     ssid cisco-voice
     speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
     station-role root
     antenna receive right
     antenna transmit right
    interface Dot11Radio0/5/0.1
     encapsulation dot1Q 1 native
     bridge-group 1
     bridge-group 1 subscriber-loop-control
     bridge-group 1 spanning-disabled
     bridge-group 1 block-unknown-source
     no bridge-group 1 source-learning
     no bridge-group 1 unicast-flooding
    interface Dot11Radio0/5/0.100
     encapsulation dot1Q 100
     bridge-group 100
     bridge-group 100 subscriber-loop-control
     bridge-group 100 spanning-disabled
     bridge-group 100 block-unknown-source
     no bridge-group 100 source-learning
     no bridge-group 100 unicast-flooding
    interface Vlan1
     no ip address
     bridge-group 1
     bridge-group 1 spanning-disabled
    interface Vlan100
     no ip address
     bridge-group 100
     bridge-group 100 spanning-disabled
    interface BVI1
     description $FW_INSIDE$
     ip address 192.168.10.2 255.255.255.0
     ip access-group 102 in
     ip nat inside
     ip virtual-reassembly in
    interface BVI100
     description $FW_INSIDE$
     ip address 10.1.1.1 255.255.255.0
     ip access-group 103 in
     ip nat inside
     ip virtual-reassembly in
    ip forward-protocol nd
    ip http server
    ip http authentication local
    ip http secure-server
    ip http path flash:/gui
    ip dns server
    ip nat inside source list 1 interface FastEthernet0/0 overload
    ip route 0.0.0.0 0.0.0.0 192.168.10.1
    ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
    access-list 1 remark SDM_ACL Category=2
    access-list 1 permit 10.1.1.0 0.0.0.255
    access-list 1 permit 192.168.10.0 0.0.0.255
    access-list 1 permit 10.1.10.0 0.0.0.3
    access-list 2 remark CCA_SIP_SOURCE_GROUP_ACL_INTERNAL
    access-list 2 remark SDM_ACL Category=1
    access-list 2 permit 192.168.10.2
    access-list 2 permit 10.1.10.0 0.0.0.3
    access-list 2 permit 192.168.10.0 0.0.0.255
    access-list 2 permit 10.1.1.0 0.0.0.255
    access-list 3 remark CCA_SIP_SOURCE_GROUP_ACL_EXTERNAL
    access-list 3 remark SDM_ACL Category=1
    access-list 3 permit 212.147.47.216
    access-list 3 deny   any
    access-list 100 remark auto generated by SDM firewall configuration
    access-list 100 remark SDM_ACL Category=1
    access-list 100 deny   ip 192.168.10.0 0.0.0.255 any
    access-list 100 deny   ip host 255.255.255.255 any
    access-list 100 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 100 permit ip any any
    access-list 101 remark auto generated by SDM firewall configuration##NO_ACES_8##
    access-list 101 remark SDM_ACL Category=1
    access-list 101 permit tcp 10.1.1.0 0.0.0.255 eq 2000 any
    access-list 101 permit udp 10.1.1.0 0.0.0.255 eq 2000 any
    access-list 101 deny   ip 10.1.1.0 0.0.0.255 any
    access-list 101 deny   ip 192.168.10.0 0.0.0.255 any
    access-list 101 deny   ip 192.168.1.0 0.0.0.255 any
    access-list 101 deny   ip host 255.255.255.255 any
    access-list 101 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 101 permit ip any any
    access-list 102 remark auto generated by SDM firewall configuration##NO_ACES_6##
    access-list 102 remark SDM_ACL Category=1
    access-list 102 deny   ip 10.1.10.0 0.0.0.3 any
    access-list 102 deny   ip 10.1.1.0 0.0.0.255 any
    access-list 102 deny   ip 192.168.1.0 0.0.0.255 any
    access-list 102 deny   ip host 255.255.255.255 any
    access-list 102 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 102 permit ip any any
    access-list 103 remark auto generated by SDM firewall configuration##NO_ACES_8##
    access-list 103 remark SDM_ACL Category=1
    access-list 103 permit tcp 10.1.10.0 0.0.0.3 any eq 2000
    access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
    access-list 103 deny   ip 10.1.10.0 0.0.0.3 any
    access-list 103 deny   ip 192.168.10.0 0.0.0.255 any
    access-list 103 deny   ip 192.168.1.0 0.0.0.255 any
    access-list 103 deny   ip host 255.255.255.255 any
    access-list 103 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 103 permit ip any any
    access-list 104 remark auto generated by SDM firewall configuration##NO_ACES_14##
    access-list 104 remark SDM_ACL Category=1
    access-list 104 deny   ip 10.1.10.0 0.0.0.3 any
    access-list 104 deny   ip 10.1.1.0 0.0.0.255 any
    access-list 104 permit ip any any
    access-list 104 permit udp host 8.8.8.8 eq domain any
    access-list 104 permit icmp any any echo-reply
    access-list 104 permit icmp any any time-exceeded
    access-list 104 permit icmp any any unreachable
    access-list 104 deny   ip 10.0.0.0 0.255.255.255 any
    access-list 104 deny   ip 172.16.0.0 0.15.255.255 any
    access-list 104 deny   ip 192.168.0.0 0.0.255.255 any
    access-list 104 deny   ip 127.0.0.0 0.255.255.255 any
    access-list 104 deny   ip host 255.255.255.255 any
    access-list 104 deny   ip host 0.0.0.0 any
    access-list 104 deny   ip any any
    control-plane
    bridge 1 route ip
    bridge 100 route ip
    voice-port 0/0/0
     cptone CH
     station-id name FAX
     station-id number 99
     caller-id enable
    voice-port 0/0/1
     cptone CH
     shutdown
     caller-id enable
    voice-port 0/0/2
     cptone CH
     shutdown
     caller-id enable
    voice-port 0/0/3
     cptone CH
     shutdown
     caller-id enable
    voice-port 0/1/0
     compand-type a-law
     cptone CH
     bearer-cap Speech
    voice-port 0/1/1
     compand-type a-law
     cptone CH
     bearer-cap Speech
    voice-port 0/4/0
     auto-cut-through
     signal immediate
     input gain auto-control -15
     description Music On Hold Port
    sccp local Loopback0
    sccp ccm 10.1.1.1 identifier 1 version 4.0
    sccp
    sccp ccm group 1
     associate ccm 1 priority 1
     associate profile 2 register mtpa4934c6ee4e0
    dspfarm profile 2 transcode
     description CCA transcoding for SIP Trunk VTX
     codec g711ulaw
     codec g711alaw
     codec g729ar8
     codec g729abr8
     maximum sessions 10
     associate application SCCP
    dial-peer cor custom
     name internal
     name local
     name local-plus
     name international
     name national
     name national-plus
     name emergency
     name toll-free
    dial-peer cor list call-internal
     member internal
    dial-peer cor list call-local
     member local
    dial-peer cor list call-local-plus
     member local-plus
    dial-peer cor list call-national
     member national
    dial-peer cor list call-national-plus
     member national-plus
    dial-peer cor list call-international
     member international
    dial-peer cor list call-emergency
     member emergency
    dial-peer cor list call-toll-free
     member toll-free
    dial-peer cor list user-internal
     member internal
     member emergency
    dial-peer cor list user-local
     member internal
     member local
     member emergency
     member toll-free
    dial-peer cor list user-local-plus
     member internal
     member local
     member local-plus
     member emergency
     member toll-free
    dial-peer cor list user-national
     member internal
     member local
     member local-plus
     member national
     member emergency
     member toll-free
    dial-peer cor list user-national-plus
     member internal
     member local
     member local-plus
     member national
     member national-plus
     member emergency
     member toll-free
    dial-peer cor list user-international
     member internal
     member local
     member local-plus
     member international
     member national
     member national-plus
     member emergency
     member toll-free
    dial-peer voice 1 pots
     destination-pattern 99
     port 0/0/0
     no sip-register
    dial-peer voice 2 pots
     port 0/0/1
     no sip-register
    dial-peer voice 3 pots
     port 0/0/2
     no sip-register
    dial-peer voice 4 pots
     port 0/0/3
     no sip-register
    dial-peer voice 5 pots
     description ** MOH Port **
     destination-pattern ABC
     port 0/4/0
     no sip-register
    dial-peer voice 6 pots
     description tcatch all dial peer for BRI/PRIv
     translation-profile incoming nondialable
     incoming called-number .%
     direct-inward-dial
    dial-peer voice 50 pots
     description ** incoming dial peer **
     incoming called-number ^AAAA$
     direct-inward-dial
     port 0/1/0
    dial-peer voice 51 pots
     description ** incoming dial peer **
     incoming called-number ^AAAA$
     direct-inward-dial
     port 0/1/1
    dial-peer voice 2000 voip
     description ** cue voicemail pilot number **
     translation-profile outgoing XFER_TO_VM_PROFILE
     destination-pattern 98
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 2001 voip
     description ** cue auto attendant number **
     translation-profile outgoing PSTN_CallForwarding
     destination-pattern 97
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 2012 voip
     description ** cue prompt manager number **
     translation-profile outgoing PSTN_CallForwarding
     destination-pattern 96
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1000 voip
     permission term
     description ** Incoming call from SIP trunk (VTX) **
     session protocol sipv2
     session target sip-server
     incoming called-number .%
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     fax rate 14400
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1001 voip
     corlist outgoing call-local
     description ** star code to SIP trunk (VTX) **
     destination-pattern *..
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     fax rate 14400
     fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1003 voip
     description ** Passthrough Inbound Calls for PSTN from CUE **
     translation-profile incoming SIP_Passthrough
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number ABCDT
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1005 voip
     description ** Passthrough Inbound Calls for MWI from CUE **
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number A80T
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1009 voip
     description ** Passthrough Inbound Calls for Internal Extensions from CUE **
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number ^..$
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1033 voip
     corlist outgoing call-local
     description **CCA*Switzerland*Short Code Services**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 0187
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1042 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Ambulance / Poisioning**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 0014[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1041 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 00333333333
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1025 voip
     corlist outgoing call-national
     description **CCA*Switzerland*National Destination Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00[789]1.......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1020 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Regional Announcement VM**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 01600
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1040 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 000333333333
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1043 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Ambulance / Poisioning**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 014[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1035 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Mobile Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 007[46789].......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1024 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Personal Numbering**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00878......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1029 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Voicemail Access**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00860.........
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1036 voip
     corlist outgoing call-national
     description **CCA*Switzerland*VPN Access**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00869.............
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1027 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Premium Rate (Business)**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00900......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1026 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Test Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00868T
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1034 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Shared Cost numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 0084[0248]......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1038 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Emergency**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 0011[278]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1037 voip
     corlist outgoing call-toll-free
     description **CCA*Switzerland*Toll Free Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00800......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1039 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*Emergency**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 011[278]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1032 voip
     corlist outgoing call-national
     description **CCA*Switzerland*National Destination Numbers**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 00[23456]........
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1023 voip
     corlist outgoing call-international
     description **CCA*Switzerland*International Calls**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 000T
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1031 voip
     description **CCA*Switzerland*Premium Rate (Social)**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 0090[16]......
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1030 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Short Code**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 014[0357]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1045 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA/Glaciers Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 0141[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1028 voip
     corlist outgoing call-national-plus
     description **CCA*Switzerland*Directory Enquiries**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 018[15].
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1021 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Short Code**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 011[45].
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1022 voip
     corlist outgoing call-national
     description **CCA*Switzerland*Short Code Services**
     translation-profile outgoing PSTN_Outgoing
     preference 1
     destination-pattern 01[67].
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 1044 voip
     corlist outgoing call-emergency
     description **CCA*Switzerland*REGA/Glaciers Air Rescue**
     translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
     preference 1
     destination-pattern 00141[45]
     session protocol sipv2
     session target sip-server
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    dial-peer voice 2002 voip
     description ** cue voicemail PSTN number **
     translation-profile outgoing VM_Profile
     destination-pattern xxx$
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 2003 voip
     description ** cue auto attendant PSTN number **
     translation-profile outgoing AA_Profile
     destination-pattern xxx$
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     voice-class sip outbound-proxy ipv4:10.1.10.1
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
    dial-peer voice 1110 pots
     preference 9
     destination-pattern xxx
     port 0/0/0
     no sip-register
    dial-peer voice 3006 voip
     description SIP
     translation-profile incoming SIP_Called_9
     session protocol sipv2
     session target sip-server
     incoming called-number xxx.
     voice-class codec 1
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     ip qos dscp cs5 media
     ip qos dscp cs4 signaling
     no vad
    no dial-peer outbound status-check pots
    sip-ua
     keepalive target dns:site1.365873.trk.ipvoip.ch
     authentication username xxx password 7 xxx
     no remote-party-id
     retry invite 2
     retry register 10
     timers connect 100
     timers keepalive active 100
     registrar dns:site1.365873.trk.ipvoip.ch expires 3600
     sip-server dns:site1.365873.trk.ipvoip.ch
     host-registrar
    telephony-service
     sdspfarm units 5
     sdspfarm transcode sessions 10
     sdspfarm tag 2 mtpa4934c6ee4e0
     video
     fxo hook-flash
     max-ephones 40
     max-dn 300
     ip source-address 10.1.1.1 port 2000
     auto assign 1 to 1 type bri
     calling-number initiator
     service phone videoCapability 1
     service phone ehookenable 1
     service phone ehookEnable 1
     service dnis overlay
     service dnis dir-lookup
     service dss
     timeouts interdigit 5
     system message SwissT.Net
     url services http://10.1.10.1/voiceview/common/login.do
     url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
     cnf-file location flash:
     cnf-file perphone
     user-locale U4 load CME-locale-de_DE-German-8.1.2.2.tar
     network-locale U4
     load 521G-524G cp524g-8-1-17
     load 525G spa525g-7-5-4
     load 501G spa50x-30x-7-5-2b
     load 502G spa50x-30x-7-5-2b
     load 504G spa50x-30x-7-5-2b
     load 508G spa50x-30x-7-5-2b
     load 509G spa50x-30x-7-5-2b
     load 525G2 spa525g-7-5-4
     load 301 spa50x-30x-7-5-2b
     load 303 spa50x-30x-7-5-2b
     time-zone 23
     time-format 24
     date-format dd-mm-yy
     keepalive 30 auxiliary 4
     voicemail 98
     max-conferences 8 gain -6
     call-forward pattern .T
     call-forward system redirecting-expanded
     hunt-group logout HLog
     moh flash:/media/music-on-hold.au
     multicast moh 239.10.16.16 port 2000
     web admin system name cisco secret 5 xxx
     dn-webedit
     time-webedit
     transfer-system full-consult dss
     transfer-pattern .T
     transfer-pattern 0.T
     transfer-pattern 6.. blind
     secondary-dialtone 0
     night-service day Sun 17:00 09:00
     night-service day Mon 17:00 09:00
     night-service day Tue 17:00 09:00
     night-service day Wed 17:00 09:00
     night-service day Thu 17:00 09:00
     night-service day Fri 17:00 09:00
     night-service day Sat 17:00 09:00
     fac standard
     create cnf-files version-stamp Jan 01 2002 00:00:00
    ephone-template  1
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     service phone webAccess 0
     softkeys remote-in-use  Newcall
     softkeys idle  Redial Pickup Mobility Newcall Cfwdall Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Trnsfer Mobility TrnsfVM Confrn Acct Park
     button-layout 7931 2
    ephone-template  15
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
     button-layout 7931 2
    ephone-template  16
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
    ephone-template  17
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  CBarge Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
    ephone-template  18
     url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
     softkeys remote-in-use  CBarge Newcall
     softkeys idle  Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
     softkeys seized  Cfwdall Endcall Redial Pickup Gpickup Callback
     softkeys connected  Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
     button-layout 7931 2
    ephone-dn  9
     number BCD no-reg primary
     description MoH
     moh out-call ABC
    ephone-dn  292
     number xxx
     description SIP Main Number registration
     preference 10
    ephone-dn  293  dual-line
     number 90 secondary xxx no-reg both
     label Zentrale
     description 90
     name Zentrale
     call-forward busy 98
     call-forward noan 98 timeout 20
    ephone-dn  294  dual-line
     number 94 secondary xxx no-reg both
     label LL
     description Lehrling Lehrnende
     name Lehrling Lehrnende
     mobility
     snr xxx delay 1 timeout 30 cfwd-noan 98
     snr ring-stop
     call-forward busy 98
     call-forward noan 98 timeout 20
    ephone-dn  295  dual-line
     number 93 secondary xxx no-reg both
     label CM
     description
     name
     snr xxx delay 1 timeout 30 cfwd-noan 98
     snr ring-stop
     call-forward busy 98
     call-forward noan 98 timeout 10
    ephone-dn  296  dual-line
     number 92 secondary xxx no-reg both
     label EE
     description
     name
     mobility
     call-forward busy 98
     call-forward noan 98 timeout 20
    ephone-dn  297  dual-line
     number 91 secondary xxx no-reg both
     label RS
     description
     name
     mobility
     snr xxx delay 1 timeout 30 cfwd-noan 98
     snr ring-stop
     call-forward busy 98
     call-forward noan 98 timeout 10
    ephone-dn  298
     number 6.. no-reg primary
     description ***CCA XFER TO VM EXTENSION***
     call-forward all 98
    ephone-dn  299
     number A801.. no-reg primary
     mwi off
    ephone-dn  300
     number A800.. no-reg primary
     mwi on
    ephone  1
     device-security-mode none
     mac-address A44C.11A0.B648
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:296 2:293 3m297 4m295
     button  5m294
    ephone  2
     device-security-mode none
     mac-address A44C.11A0.B566
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:297 2:293 3m296 4m295
     button  5m294
    ephone  3
     device-security-mode none
     mac-address A44C.11A0.B5C4
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:295 2:293 3m297 4m296
     button  5m294
    ephone  4
     device-security-mode none
     mac-address A44C.11A0.B67A
     ephone-template 1
     max-calls-per-button 2
     username "xxx" password xxx
     type 525G2
     button  1:294 2:293 3m297 4m296
     button  5m295
    alias exec cca_voice_mode PBX
    alias exec cca_vm_notification schedule from_time=00 to_time=24
    alias exec clid-ALL_BRI ;1:0-4;1:0-9;1:0-9;1:1-9
    alias exec clid-SIP ;1:1-9;1:1-9;1:1-9
    banner login ^CCisco Configuration Assistant. Version: 3.2 (3). Fri Jul 04 13:18:33 CEST 2014^C
    line con 0
     no modem enable
    line aux 0
    line 2
     no activation-character
     no exec
     transport preferred none
     transport input all
    line vty 0 4
     transport preferred none
     transport input all
    line vty 5 100
     transport preferred none
     transport input all
    ntp master
    ntp server 91.240.0.5 prefer
    en

    Hi Patrick
    I am working on this one as well. I have a UC560 with SIP Trunk provider Les.NET.
    It was working fine until a few weeks ago when something changed on the provider end and broke it. My hunch it is something to do with the SIP REFER.
    http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html
    Here is an excerpt from the above page:
    Call Transfer
    When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:
    Router(config)#voice service voip
    Router(conf-voi-serv)#no supplementary-service sip refer
    Figure 3 shows the behavior of the CME system with the REFER method disabled.

  • 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"

  • Considerations on migrating from TDM PRI's to SIP Trunking?

    Hello,
    We are planning a migration from traditional TDM PRI's to SIP trunking for inbound toll-free and outbound LD telephone traffic. Currently, we have 7 x PRI's connected to a 3945 router feeding our CUCM 8.6 system. This works fine. We are planning on making use of a 100 Mb data circuit to bring in SIP service from our carrier. This would terminate on our 3945 voice gateway. We would have roughly the same number of SIP concurrent call paths as we have PRI channels (162).
    I know that the 3945 makes heavy use of DSP resources to handle the PRI traffic. Are DSP resources needed for SIP traffic as well?
    Will the SIP usage (162 SIP concurrent call paths) cause similar router utilization as the PRI's (router CPU, memory, etc.)?
    What are other things we should look out for in this migration?
    Thank you!
    Brian

    1. Just to add to the excellent tip provided by George (+5), Here is the capacity matrix for the 3900 gateways. Your 3945 can support 950 concurrent calls. So in terms of capacity, you are well taken care of.
     Number of IP-to-IP Calls per Platform
    Platform
    Maximum Number of Simultaneous Calls (Flow-Through)
    Cisco 3945E
    2500
    Cisco 3925E
    2100
    Cisco 3945
    950
    Cisco 3925
    800
    Cisco 2951
    500
    Cisco 2921
    400
    Cisco 2911
    200
    Cisco 2901
    100
    Cisco ASR 1004; and Cisco ASR 1006 Router Processor 2 (RP2)
    5000; 16000*
    Cisco ASR 1002, ASR 1004, and ASR 1006 RP1
    1750
    Cisco AS5350XM and AS5400XM
    600
    Cisco 3845
    500
    Cisco 3825
    400
    Cisco 2851
    225
    Cisco 2821
    200
    Cisco 2811
    110
    Cisco 2801
    55
    2. You should definitely make provisions for dsps. You may need DSPs for MTP, xcoding, etc. Especially with SIP, MTP may be needed for DTMF mistmatch, supplementary services etc
    3. One of the most important thing to consider is the codec you will use for your calls. Your users are used to PSTN (TDM). Using G711 on your sip calls is not even the same as the traditional PSTN. The quality will be noticeable. Using G729 is going to be distinct. I have seen where users complain of this during a deployment. Using G729 will be a rude shock and they may not like it.
    4. You also need to consider your analogue options etc FAX. What fax protocol does your ITSP support etc...We have been sued before by a customer because their fax machines didnt work. The ITSP said it supported T.38 while in reality it didnt.
    5. In terms of memory utilization and CPU, I will think that it should be less for IP-IP call. This is because in an IP-TDM call, your router is constantly encoding via your dsp the IP payload (codecs) to TDM  for transimissiont ot he PSTN. This wont be happening any more.

  • ATA187, SIP trunks, CUBE

    Hi,
    What dial-peer fax statements would be advised in a case where the ITSP (SIP Trunking) only supports g711a pass though for fax?
    We are having a hard time making outbound faxes work reliably from a ATA187.
    How much 'relay' can/should the gateway do in this case (to my understanding there is no 'relay' in this case)?
    Should we rely solely on the ATA187 configuration in CUCM and the settings on the fax machine ? (Speed, ECM, etc)
    Gateway/CUBE 15.3, CUCM 9.1.1a
    Regards,
    Erik
    Sent from Cisco Technical Support iPhone App

    The following is an example config you can make use of with an SIP enviroment.
    voice service voip
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    fax protocol pass-through g711alaw
    h323
    modem passthrough nse codec g711alaw
    sip
      registrar server expires max 3600 min 3600
      asserted-id pai
      privacy pstn
      options-ping 60
      midcall-signaling passthru
    dial-peer voice 200 voip
    description -----  WYCHODZACE SIP-T ------
    destination-pattern [1-9]........
    modem passthrough nse codec g711alaw
    session protocol sipv2
    session target ipv4:SIP-SRV:5020
    voice-class codec 23 
    voice-class sip dtmf-relay force rtp-nte
    voice-class sip early-offer forced
    voice-class sip pass-thru headers unsupp
    voice-class sip pass-thru content unsupp
    voice-class sip pass-thru content sdp
    dtmf-relay rtp-nte
    req-qos controlled-load audio
    fax-relay ecm disable
    fax protocol pass-through g711alaw
    no vad
    dial-peer voice 101 voip
    description **Outgoing Call to CUCM**
    destination-pattern 895216...
    session target ipv4:CCM1
    voice-class codec 1 
    voice-class h323 1
    voice-class sip dtmf-relay force rtp-nte
    voice-class sip early-offer forced
    dtmf-relay h245-alphanumeric
    fax protocol pass-through g711alaw

  • Unity Connection not passing CallerID to CUCM over SIP Trunk

    I'm trying to get CallerID working for Unity Connection Device Notification (and it seems everything else), however, when I run UC Remote Port Status Monitor and the Call-Out goes to CUCM for the Device Notification, no caller ID is presented to the CUCM SIP trunk.
    06:06:02, New Call, CalledId=,  RedirectingId=,  Origin=16,  Reason=1024,  CallGuid=, 
    CallerName=,  LastRedirectingId=,  LastRedirectingReason=1024,  PortDisplayName=LFC_CUCM-1-134,
    [Origin=Unknown],[Reason=Unknown]
    06:06:02,
    Dialing '99254753'
    06:06:32, Idle
    06:06:33, Idle
    Therefore, the out-going call to the PRI PSTN is:
    10:59:01.005: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8  callref = 0x5B03
            Sending Complete
            Bearer Capability i = 0x8090A2
                    Standard = CCITT
                    Transfer Capability = Speech
                    Transfer Mode = Circuit
                    Transfer Rate = 64 kbit/s
            Channel ID i = 0xA98397
                    Exclusive, Channel 23
            Calling Party Number i = 0x0081, N/A
                    Plan:Unknown, Type:Unknown
            Called Party Number i = 0xC1, '9254753'
                    Plan:ISDN, Type:Subscriber(local)
    *Dec  6 10:59:01.513: ISDN Se0/0/0:23 Q931: RX <- CALL_PR
    I've looked through my SIP trunk on the CUCM side and for Inbound Calls, Connected Line ID and Presentation Name are set to "allowed" or "default" doesn't make a difference. RTMT Port Status also shows no "caller", so I'm thinking there is some way to set or allow the calling number on the Unity Connection (8.5) side.
    Oddly enough, I also noticed that in Unity Connection> Telephony Integrations > Port Group, if I change the Contact Line Name from nothing to "Unity" (or whatever), the Q931 debug outbound doesn't show ANY "Calling Party Numer - = XXXXX" and the carrier throws out the BTN as the ANI.
    10:46:00.837: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8  callref = 0x5AFF
            Sending Complete
            Bearer Capability i = 0x8090A2
                    Standard = CCITT
                    Transfer Capability = Speech
                    Transfer Mode = Circuit
                    Transfer Rate = 64 kbit/s
            Channel ID i = 0xA98397
                    Exclusive, Channel 23
            Called Party Number i = 0xC1, '9254753'
                    Plan:ISDN, Type:Subscriber(local)
    Any ideas on where/how CallerID comes from, on Unity Connection with a SIP integration?
    THANKS!!
    Mike.

    I did not- my work around has been to put in a name for Contact Line Name under Port Group Basics Switch configuration in Unity Connection- this for some reason keeps CUCM from sending ANI TYPE/PLAN information in the Q931 message, and my carrier then sends a default ANI of the circuit's BTN. When I have time, I'll open up a TAC ticket.
    Mike.

  • Calls from Sip Trunk to UC540 and then to CUE returned ** Service Unavailable**

    Hi to all
    i have something strange here and i need your assistance
    Call Flow:
    Sip trunk-->UC540--> CUE
    When calls coming to UC540 from outside and then going to cue then we send back service unavailable.I made a translation and i sent directly the incoming calls to CUE
    The same behavior is also if i send the calls to dummy number and then from there set forward all to voice mail.
    Incoming voicemail is working fine
    Incoming calls to phones also ok
    Uc540: 8.6
    CUE: 8.6.5
    A number: 99999999
    B number: 22777777
    Voice Mail Number:111
    Attached is the trace
    i see that we hit the correct dial peers .
    I have enable only trancoder since MTP is not register ( don't know why , but i don't think also that is necessary..
    voice service voip
     ip address trusted list
      ipv4 172.16.80.0 255.255.255.0
      ipv4 172.16.81.0 255.255.255.0
     allow-connections sip to sip
     supplementary-service h450.12
     no supplementary-service sip moved-temporarily
     no supplementary-service sip refer
     supplementary-service media-renegotiate
     sip
      no update-callerid
    dial-peer voice 1000 voip
     description **SIP TRUNK**
     translation-profile incoming SIP-INCOMING
     translation-profile outgoing SIP-OUTGOING
     destination-pattern 9T
     modem passthrough nse codec g711alaw
     session protocol sipv2
     session target sip-server
     incoming called-number .T
     voice-class codec 2  
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     fax-relay ecm disable
     no fax-relay sg3-to-g3
     fax rate 9600
     fax protocol pass-through g711alaw
     no vad
    dial-peer voice 2001 voip
     description ** cue voicemail pilot number **
     destination-pattern 111
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number 111
     no voice-class sip outbound-proxy   
     dtmf-relay sip-notify
     codec g711ulaw
     no vad
    Regards
    chrysostomos

    Hi
    Interface                  IP-Address      OK? Method Status                Protocol
    FastEthernet0/0            unassigned      YES NVRAM  up                    up
    FastEthernet0/0.10         192.168.0.10    YES DHCP   up                    up   ----> For internet
    FastEthernet0/0.20         10.151.5.130    YES NVRAM  up                    up  ------> For sip trunk
    In0/0                      10.1.10.2       YES unset  up                    up    --------> default gw for cue
    Vlan1                      unassigned      YES unset  up                    up
    Vlan100                    unassigned      YES unset  up                    up
    Vlan200                    unassigned      YES unset  up                    down
    Vlan300                    unassigned      YES unset  up                    down
    NVI0                       10.1.10.2       YES unset  up                    up
    BVI1                       192.168.20.1    YES NVRAM  up                    up
    BVI100                     10.1.1.1        YES NVRAM  up                    up   ---------> ip for cme
    Loopback0                  10.1.10.2       YES NVRAM  up                    up   ------> default gw for cue
    dial-peer voice 2001 voip
     description ** cue voicemail pilot number **
     destination-pattern 111
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number 111
     no voice-class sip outbound-proxy
     voice-class sip bind control source-interface BVI100
     voice-class sip bind media source-interface BVI100
     dtmf-relay sip-notify
     codec g711ulaw
     no vad
    interface FastEthernet0/0.10
     description **FOR INTERNET**
     encapsulation dot1Q 10
     ip address dhcp
     ip access-group 105 in
     ip nat outside
     ip inspect SDM_LOW out
     ip virtual-reassembly in
    interface FastEthernet0/0.20
     description **FOR SIP TRUNK WITH ISP**
     encapsulation dot1Q 20
     ip address 10.151.5.130 255.255.255.240
    ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
    ping 10.1.10.1 source bvi100
    Type escape sequence to abort.
    Sending 5, 100-byte ICMP Echos to 10.1.10.1, timeout is 2 seconds:
    Packet sent with a source address of 10.1.1.1
    Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/1 ms
    I have bind the interface of cme ( 10.1.1.1) but the call fails again
    Attached is the trace
    Anything to advice?

  • 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

  • NexVortex SIP trunk and UC500 default timeout settings?

    Hey guys,
    I'm doing a little SIP trunk testing to determine a good provider for my customer base, and had some general questions as I can't seem to get outgoing or incoming phone calls to work at all.
    To keep things simple, I'm using an 8user UC540W with 3 IP phones - a 525G, a 524G, and a 7937 conference phone.  I have a static IP on the UC540, have run through the telephony wizard and everything seems to be working on the LAN/PBX side of things.  The big difference, and the major variable that we are working with (I believe), is that we're working with Satellite internet connectivity rather than terrestrial Internet connectivity.  This is an Enterprise satellite connection, and we have run voice over the connection without problems, but this is our first attempts at SIP trunking from a UC500.  Due to the latency involved inherent in satellite (ping times around 550-700ms), I believe that either UC540 or NexVortex server/switch is timing out.  Is there any way to determine what the default setting is for a SIP acknowledgement on the UC540 and change this if it is too small?
    Here is what I have found, if it is helpful:
    Outgoing calls:
    1. The SIP provider, NexVortex, says that they are seeing an invite from the UC540, but not on port 5060.  On the two calls that we tested, it first saw an invite on 63452, and then on 51677.  Is there any reason why this would not be sent out on 5060?
    Incomign calls:
    1. On incoming calls, Nexvortex is routing the calls to the proper IP, but is then receiving an "error 500 reason Q850" from the UC540.  What does this error mean?
    I am also attaching my config in the event that it helps.  When I look at the SIP trunk status in CCA, it does not show that registration is working, so I assume that's a good place to start.
    Lastly, the guys over at NexVortex don't seem to run across the UC500 very often.  If anybody has setup their UC500 to work with NexVortex and wouldn't mind posting a screenshot from CCA (feel free to remove usernames and passwords), I'd appreciate it.  I'm not certain that I have all of the information in the right places.
    Thanks,
    Seth

    Hi Steven,
    Thanks for the continued help.
    I was able to make the changes in the config.  Here are snapshots from the current config:
    dial-peer voice 1000 voip
    description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
    voice-class codec 1
    voice-class sip dtmf-relay force rtp-nte
    session protocol sipv2
    session target sip-server
    incoming called-number .%
    dtmf-relay rtp-nte
    ip qos dscp cs5 media
    ip qos dscp cs4 signaling
    no vad
    dial-peer voice 3000 voip
    description IncomingSIP
    translation-profile incoming IncomingSIP_Called_4
    voice-class codec 1
    voice-class sip dtmf-relay force rtp-nte
    session protocol sipv2
    session target sip-server
    incoming called-number 14068906254$
    dtmf-relay rtp-nte
    ip qos dscp cs5 media
    ip qos dscp cs4 signaling
    no vad
    dial-peer voice 3001 voip
    description IncomingSIP2
    translation-profile incoming IncomingSIP2_Called_5
    voice-class codec 1
    voice-class sip dtmf-relay force rtp-nte
    session protocol sipv2
    session target sip-server
    incoming called-number 1406890624[2-3]$
    dtmf-relay rtp-nte
    ip qos dscp cs5 media
    ip qos dscp cs4 signaling
    no vad
    dial-peer voice 3002 voip
    incoming called-number 14068906254$
    no dial-peer outbound status-check pots
    sip-ua
    authentication username nomadgcs password 7 *removed*
    no remote-party-id
    retry invite 2
    retry register 10
    timers connect 100
    registrar ipv4:66.23.129.253:5060 expires 3600
    sip-server ipv4:66.23.129.253:5060
    connection-reuse
    host-registrar
    We are calling from within the 406 area code, so when we dial the number with the leading 406, we get a message saying "You don't need the area code" from the telephone company.  When we dial this from a cell, we get the following:
    1. 4068906254 - "All circuits are busy, please try your call again..."
    2. 8906254 - rings once, then no sound, then disconnects after about 10 seconds.
    I don't know if this would factor in at all, but our NexVortex account is setup to deliver 14068906254 to the UC500, but would NexVortex deliver the entire string of characters if it is only receiving 4068906254 or 8906254?
    Thanks,
    Seth

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