Which UK SIP Trunk Provider?

Hi,
I'm looking for a UK SIP trunk service provider. I dont wish to be tied to a specific ISP. Does a list of confirmed working providers exist?
Best regards,
Paul.

Afternoon Peter,
Of course, it's a pleasure:
Description:  Voiptalk
Proxy:  voiptalk.org
Require Registration: Ticked
Uses SIP Remote-Party-ID:  Unticked
Outbound Proxy:  nat.voiptalk.org
Only Allow Calls To:  (Leave this box blank)
Call Capacity: 2
Prefix dialed numbers with "+44": Unticked
Company Name: (Leave this box blank)
Account ID:  Enter your Account ID in this box, e.g. 8442855xx (where the xx are also numbers)
Authentication ID: 
Account ID:  Enter your Account ID in this box, e.g. 8442855xx (where the xx are also numbers this is the same number as entered in the Account ID box above)
Password:  Enter your voiptalk password for the account number entered above
SRV Record Lookup:  Unticked
SRV Auto Prefix:  Unticked
NAT Mapping: Unticked
Send Nat Keep Alive: Unticked
Local SIP Port: 5060
It's pretty straigh forward but let me know if you have any problems and we'll see if I can get a number to you to give me a call and I'll try talk you through it.
Regards,
Paul.

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    access-list 102 deny   ip 127.0.0.0 0.255.255.255 any
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    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
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    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
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    dial-peer cor list user-national-plus
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    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
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    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:
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    Router(conf-voi-serv)#no supplementary-service sip refer
    Figure 3 shows the behavior of the CME system with the REFER method disabled.

  • 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:
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    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?
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    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,
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    description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
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    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

  • Inbound SIP trunk busy when routed to AA

    Hey guys,
    We have been having some strange things with our UC520 lately, so I built up a UC540 as a backup, and then rebuilt our UC520.  Both of these systems are exhibiting the same behavior on all inbound SIP calls that are routed directly to our AA - a fast busy signal.
    At the end of the day, I want all incoming SIP calls to go to a blast group during the day, and our After Hours AA . . . well, after hours.  The way that I would like to accomplish this is through a combination of Floating Extensions and Night Service, but it doesn't quite work.  Ideally, my floating extension would forward all calls to my initial blast group during the day.  Night Service would forward those same calls to our AA after hours.  Floating extensions works to automatically forward inbound calls to my blast group, but the Night Service rules don't bypass that blast group to send the inbound calls directly to our AA.  You have to wait until the cfna timer completes before you finally get the AA.
    So in the past, I've just set up a CIPC where I do the same thing.  Configure the CIPC to forward all calls to the blast group, then set up night service.  This actually works, but right now, those inbound calls give me a fast busy whenever they are delivered to the AA extension.  All other extensions seem to work okay.
    It seems like this is probably a transcoding issue, but I've not been able to find it.  Preferred codec on the UC is G.711ulaw.
    SIP Trunk provider is NexVortex.
    Any suggestions on where to start looking?  Thanks for your help!
    Seth

    Figured it out.  I had turned off "hairpinning" for the SIP trunk.  It would appear that the system was therefore not forwarding the calls to the AA.

  • SIP Trunk not accepting inbound calls

    I have a CME setup using Engin as a SIP provider
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    Router#sh sip-ua register status
    Line                             peer       expires(sec) registered P-Associ-URI
    ================================ ========== ============ ========== ============
    038682XXXX                       -1         1124         yes       
    101                              20001      45           no        
    102                              20003      18           no        
    103                              20005      45           no        
    104                              20006      45           no        
    I do see the call come in if I debug the dial peer, but it only seems to match an outgoing dp
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    Router#
    Sep  8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
    Sep  8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:4F947560
    Sep  8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:uas, Min-SE Value:1800, flags:2001
    Sep  8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
    Sep  8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:4F947DF8
    Sep  8 18:15:15.025: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
    Sep  8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
    Sep  8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:403, container:4F947B38
    Sep  8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x4C39C570
    State of The Call        : STATE_DEAD
    TCP Sockets Used         : NO
    Calling Number           : 0417XXXXXX
    Called Number            : 038682XXXX
    Source IP Address (Sig  ): 211.30.48.136
    Destn SIP Req Addr:Port  : 203.161.164.69:5060
    Destn SIP Resp Addr:Port : 203.161.164.69:5060
    Destination Name         : 203.161.164.69
    Sep  8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream             : 1
    Negotiated Codec         : g711ulaw
    Negotiated Codec Bytes   : 160
    Nego. Codec payload      : 0 (tx), 0 (rx)
    Negotiated Dtmf-relay    : 0
    Dtmf-relay Payload       : 0 (tx), 0 (rx)
    Source IP Address (Media): 211.30.48.136
    Source IP Port    (Media): 17768
    Destn  IP Address (Media): 203.161.164.69
    Destn  IP Port    (Media): 18314
    Orig Destn IP Address:Port (Media): [ - ]:0
    Sep  8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC)    : 21
    Disconnect Cause (SIP)   : 403
    Any Ideas
    Doug

    Hi Tapan,
    Firstly the topology is as follows
    ISP/VOIP provider - Internet - Cable modem - 2800 CME router - IP Phone
    The VM is provided by the ISP
    debug ccsip messages
    Sep  9 10:21:41.488: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    CSeq: 633854439 INVITE
    Contact:
    Supported: 100rel,timer
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Accept: multipart/mixed,application/media_control+xml,application/sdp
    Min-SE: 60
    Session-Expires: 1800;refresher=uas
    Max-Forwards: 9
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 317
    v=0
    o=BroadWorks 18275729 1 IN IP4 203.161.164.69
    s=-
    c=IN IP4 203.161.164.69
    t=0 0
    m=audio 18128 RTP/AVP 18 8 0 101
    c=IN IP4 203.161.164.69
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=bsoft: 1 image udptl t38
    Sep  9 10:21:41.508: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>
    Date: Fri, 09 Sep 2011 00:21:41 GMT
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    CSeq: 633854439 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    Sep  9 10:21:41.516: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>;tag=39373A4-586
    Date: Fri, 09 Sep 2011 00:21:41 GMT
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    CSeq: 633854439 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=21
    Content-Length: 0
    Sep  9 10:21:41.544: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
    CSeq: 633854439 ACK
    From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
    To: "Doug Goding"[email protected]>;tag=39373A4-586
    Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
    Max-Forwards: 9
    Content-Length: 0
    Voice Config
    Router#
    voice service voip
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    sip
      registrar server expires max 3600 min 3600
      localhost dns:mel.byo.engin.com.au
      no call service stop
    voice class codec 1
    codec preference 1 g711ulaw
    voice translation-rule 10
    rule 1 /^0/ //
    voice translation-rule 11
    rule 1 /^.*/ /0386821234/
    voice translation-profile PSTN_Outgoing
    translate calling 11
    voice-card 0
    dsp services dspfarm
    mgcp profile default
    sccp local Vlan100
    sccp ccm 10.1.100.1 identifier 1 version 7.0
    sccp
    sccp ccm group 1
    bind interface Vlan100
    associate ccm 1 priority 1
    associate profile 1 register confdsp
    dspfarm profile 1 conference 
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    maximum sessions 4
    associate application SCCP
    dial-peer voice 99 voip
    translation-profile outgoing PSTN_Outgoing
    destination-pattern .T
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 100 voip
    session protocol sipv2
    session target dns:mel.byo.engin.com.au
    incoming called-number 0386821234
    dtmf-relay sip-notify
    codec g711ulaw
    no vad
    dial-peer voice 110 voip
    description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
    session protocol sipv2
    session target dns:mel.byo.engin.com.au
    incoming called-number .%
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 90 voip
    description Melbourne 03 Numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern [89].......
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 91 voip
    description National Numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 0[278]........
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 92 voip
    description Vic/Tas 03 numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern [56].......
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 93 voip
    description Mobile numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 04........
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 94 voip
    description 13XXXX numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 13[1-9]...
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 96 voip
    description 1300/1800 numbers
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 1[38]00......
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 98 voip
    description Emergency 000
    translation-profile outgoing PSTN_Outgoing
    destination-pattern 000
    session protocol sipv2
    session target sip-server
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    sip-ua
    credentials username 0386821234 password 7 XXXX realm voice.mibroadband.com.au
    authentication username 0386821234 password 7 XXXX
    nat symmetric role active
    nat symmetric check-media-src
    no remote-party-id
    retry invite 2
    retry register 10
    timers connect 100
    mwi-server dns:mel.byo.engin.com.au expires 3600 port 5060 transport udp unsolicited
    registrar dns:mel.byo.engin.com.au expires 3600
    sip-server dns:mel.byo.engin.com.au
    connection-reuse
    telephony-service
    sdspfarm conference mute-on #1 mute-off #2
    sdspfarm units 2
    sdspfarm tag 1 confdsp
    conference hardware
    max-ephones 42
    max-dn 144
    ip source-address 10.1.100.1 port 2000
    calling-number initiator
    service phone videoCapability 1
    service phone displayOnDuration 00:01
    service phone displayOnTime 08:30
    service phone displayOffTime 17:30
    service phone displayIdleTimeout 00:01
    service phone displayOnWhenIncomingCall 1
    system message Cisco CME
    load 7941 SCCP41.8-4-2S
    load 7942 SCCP42.8-4-2S
    load 7945 SCCP45.8-4-2S
    load 7961 SCCP41.8-4-2S
    load 7962 SCCP42.8-4-2S
    load 7965 SCCP45.8-4-2S
    load ata ATA030204SCCP090202A
    time-zone 48
    date-format dd-mm-yy
    voicemail 90125200
    mwi relay
    max-conferences 8 gain -6
    call-forward pattern .T
    call-forward system redirecting-expanded
    moh music-on-hold.au
    web admin system name cisco secret 5 $1$d8/H$glhLiCCWXmFSUp6BtwGho0
    dn-webedit
    time-webedit
    transfer-system full-consult
    transfer-pattern 0.T
    create cnf-files version-stamp 7960 Jul 06 2011 10:32:45
    ephone-dn  1  dual-line
    number 038682XXXX
    label 101
    name 7965
    mwi sip
    ephone-dn  2  dual-line
    number 102
    label 102
    name 7941
    ephone-dn  3  dual-line
    number 103
    label 103
    name 7920
    ephone  1
    device-security-mode none
    video
    mac-address 0023.5EB8.6E4E
    type 7965
    button  1:2 2:1
    ephone  3
    device-security-mode none
    mac-address 0019.0633.A933
    max-calls-per-button 2
    type 7920
    button  1:3
    ephone  10
    device-security-mode none
    mac-address 0019.E7B7.BAB3
    max-calls-per-button 2
    type ata
    button  1:1

  • MoH is playing from the sip trunk

    Hi currently when first users call to queue pilot number and go to the hold, the MoH is played correctly but MOH of sip trunk provider is played for the second caller to queue.
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    I have the exact same issue ...
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    All the calls in hold are with the moh of sip provider ... and i can't unhold calls .

  • Sip trunk between CUCM7.0 and third party VOIP provider

    Hi all,
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    Got SIP username, password and SIP-proxy IP from the provider.
    I've done such a setup on CUCME a couple of times, but never on the CUCM.
    Who can put me on right way?
    Can it be done on the CUCM, or must an IOS-Device be used (got a PSTN-GW connected through H323 with CUCM)?
    THanks for the hint,
    Greets Norbert

    Here we go.....
    CONFIG (Version=7.1)
    =====================
    Version 7.1
    Cisco Unified Communications Manager Express
    ! Calling nr. incoming
    voice translation-rule 40
    rule 1 /\(.*\)/ /0\1/
    ! Discard prefix (calling nr.)
    voice translation-rule 190
    rule 1 /^0\(.*\)/ /\1/
    rule 2 /^9\(.*\)/ /\1/
    ! Mapping, internat to external nr.
    voice translation-rule 191
    rule 10 /^[1-9].*/ /xxxxEXTERNALxxxx/
    ! for call-forwarding
    rule 15 /^0\(.*\)/ /\1/
    ! Mapping external to internal nr.
    voice translation-rule 192
    rule 2 /^xxxxxEXTERNALxxxx/ /4xx/
    voice translation-profile TP_IN_SIP
    translate calling 40
    translate called 192
    voice translation-profile TP_OUT_SIP
    translate calling 191
    translate called 190
    dial-peer voice 2001 voip
    corlist outgoing dialCORnoFax
    description *** SIP-TRUNK (OUT) ***
    translation-profile incoming TP_IN_SIP
    translation-profile outgoing TP_OUT_SIP
    max-conn 2
    destination-pattern 9.T
    session protocol sipv2
    session target ipv4:2xx.xxx.xxx.xxx
    session transport udp
    ! customer external nr. range (one dot at the and -> 0-9)
    incoming called-number xxxxxxxx.
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    gateway
    timer receive-rtp 1200
    sip-ua
    keepalive target ipv4:2xx.xxx.xxx.xxx
    authentication username xxEXTERNAL NR.xxxxx password 7 111111111111111111111
    calling-info pstn-to-sip from number set xxEXTERNAL NR.xxxxx
    retry invite 2
    retry response 2
    retry bye 2
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  • Controlling which CUCM server communicate over a SIP trunk

    We have 3 CUCM servers, two sub and one pub at two different physical locations.
    There are two SIP trunk servers (non Cisco), we wish to have the CUCM at the same physical location to communicate with the SIP trunk device at its location, instead of going over the WAN to communicate with the other one.
    Communications are initiated from the CUCM side through a route pattern that points to a RL/RG that contains the SIP trunk.
    The SIP trunk uses a DP that uses a CUCM group that only contains the local CUCM server.
    Can this be setup to reliably control which CUCM server talks to the SIP trunk server at its location?
    Do we need to configure something differently?

    Hi,
    If the CUCM's are located in different physical location then the communication should be over the IP-WAN and if it is about different CUCM Cluster then we can use H323 ICT Trunk between CUCM server for communication which is again WANLink.
    Still not clear with your query, However as you want to control the Gwy/Trunks using the Route Pattern. You can create two separate route pattern for internal & external.
    Wherein in Internal Route Pattern you can specify the ICT/SipTrunk/GWY which is connected within the cluster under Route List/RouteGroup to route the call.
    And for External Route Pattern specify the Gwy/SipTrunk which connects you to the outside world.
    Regards,
    Venkatesh

  • Changing external Caller ID over a SIP Trunk to SIP Provider

    I am working with a client and when they place calls out to any external user they have the wrong name showing on the external caller ID. 
    I have spoken with the SIP provider and apparently they want us to pass the CNAM, or rather they have it setup for us to do this.
    I opened a case with Cisco and the TAC engineer said the provider has to do this because it cannot be done from CUCM or the gateway.
    For example, it says right now "location A" for external calls and I want to change this to say "location B" . 
    Is this even possible?

    what is the call flow? did you check the caller name in SIP trunk configuration?

  • Global SIP trunking from single provider

    Our large global organisation is deploying on-premise Lync EV across 200+ sites globally.
    We want to minimise the number of suppliers we use in this project and use SIP trunking where possible.
    Are there any global telcos that would be able to offer me SIP trunking, with global points of presence on their network, and also 'full' SIP services for most countries.
    By full SIP services I mean, competitive pricing (compared to local Telco), emergency number dialling, other special numbers, and full number portability
    I have found some of our existing SIP providers can provide services in other countries, but they are not very good at it, and it is overpriced and they don't really know what they are doing.
    Surely there must be a few big players who can offer me a truly global service with regional trunks and services for all our locations?

    Hi There
    That’s a great question. The answer is, yes there are. There are a number of global carriers or at least carriers who claim to be global (and who
    only serve NA/LA and Europe with bits of the pacific. However, if you use one of these guys, prepare to pay a premium, as their target market is large enterprise business who are prepared to pay for the privilege of a 'one throat too choke' relationship. I
    have worked for these carriers, so know this to be fact. In most countries where this TDM telephony and now Internet, somebody has built an in country SIP business. These in country providers, tend to focus on the small to medium up to mid-market enterprise
    community, for customers who want in country SIP services. A number of regional integrators have gone to these providers in the major geographies and built SIP propositions on top of their own networks and provider these to customers who are prepared to pay
    for the privilege of having one supplier (again I know this to be fact, as I have worked for them).
    What the industry needs is a collective agreement where Tier 1 SIP providers in each of these countries, create an alliance based network for the
    handoff of traffic at International colocation points. For example, my company in the UK perhaps peering at Tele house and enabling the wholesale SIP connectivity to any carrier who likewise wants to be colocating at that point. This would enable each SIP
    carrier to provider their own A-Z IPT tariff and offer in country local break out and emergency services, enabling business A in Country A to call Business B in Country B for the price of a local phone call.
    It's so obvious, but finding someone who does this is a challenge. Probable today each of these in country SIP carriers have enough business by
    focusing on their own country, without the need to expand their geographies and put their hands in their pockets for the required investment. Just as in the TDM industry where carriers built global A-Z traffic exchange points, this organic growth and service
    evolution will take place. When, probably depends on one of two things happening. Either (a) the in country carriers seeing more opportunity from their existing customer base to develop an offer for international service provision or (b) someone visionary
    and with deep pockets grabbing the opportunity.
    The solution for you? Take service from your in country provider and build your own international end points if you can. If I can be of any father
    help do let me know.

  • 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

  • BE6000 and SIP trunk

    Hello!
    Now we use Cisco IP telephony based on Cisco CME. Cisco CME is installed on 2821 router with IOS C2800NM-ADVENTERPRISEK9-M, Version 12.4(24)T7. A SIP trunk is configured between CME router and telephony provider.
    We plan to upgrade our telephone system and begin using Cisco CBE6000. I study the documentation and in "Cisco Business Edition Unified Communications and Collaboration Solutions for Small and Midsized Companies", I find that for SIP trunk to works in CBE6000 a Cisco Integrated Services Router with Cisco Unified Border Element is required. Or other quote from this document "Public switched telephone network (PSTN) connections using SIP is available through any Cisco Integrated Services Router voice gateway".
    In this regard there is a question: what is the functions of this router when I make a trunk between CBE6000 and telephony provider SIP gateway and whether it is possible to use existing Cisco 2821 router as such router.
    Thanks.

    Hi Ayodeji,
    I too have a somewhat similar query.
    We have UC560 running currently at our company, the UC560 is connected to an ISDN modem from the network provider. We are now upgrading to BE6000. To connect to the ISDN E1/T1 line, we've ordered Cisco 2921 with a E1/T1 VWIC card (VWIC3-2MFT-T1/E1) which will connect to BE6000 and internal LAN. I just wanted to know what would I need to configure on 2921?
    Do I need to install CUBE or just configure it to connect to ISDN line?
    Many thanks for any help.
    regards,

  • Calls via Gamme sip trunk

    Has anyone set up Lync server 2010 to use the Gamma SIP trunks, that dont require the use of a gateway?
    No requirement for an additional gateway device, with direct MS Lync connectivity
    The trouble is i cannot get Lync to connect to the trunks. We have purchased the SIP trunks from a gamma supplier(we didnt now they were a supplier, until recently when we asked for support and they went 'duhhhhhh me no know, we just
    sell things dunow how to set things up' what a PAIN IN THE A***), and they say that the SIP trunks are pointed at our EFM IP address. which also has the DDIs assigned to it.
    So, i setup a PSTN gateway on lync topology using IP of EFM, Listening port 5060 using TCP. Are these ports and protocol okay?
    The VoIP phones seem to want to call, they just lack any sound, no ringing tone, no dissconnected tone. Just says calling "+44157322****" So the dial plan is changing 22**** to the correct local code and whatever the +44 thing
    is.
    Any advice on how i can find the problem, or how to setup the trunks up would be hugely appreciated.
    P.S We initially tried to use an audiocodes mediant 1000, which was what we asked our trunk supplier about, and then they informed us about being a gamma supplier, and that the gamma trunks do not require a gateway. Followed setting
    up guide for mediant 1000 with gamma trunks through audiocodes blah, to no success. I think thats because it was changing the coders, which was not needed if the trunks are directly compatable.

    Hi,
    Please review the SIP trunk topology.
    http://technet.microsoft.com/en-us/library/gg398720.aspx
    To
     implement SIP trunking, you must route the connection through a Mediation Server, which proxies communications sessions between Lync Server 2010 clients and the service provider and transcodes media when necessary. Each Mediation Server has
    an internal and an external network interface. The internal interface connects to the Front End Servers. The external interface is commonly called the gateway interface because it has traditionally been used to connect the Mediation Server to a PSTN gateway
    or an IP-PBX. To implement a SIP trunk, you connect the external interface of the Mediation Server to the external edge component of the ITSP. The external edge component of the ITSP could be a Session Border Controller (SBC), a router, or a gateway.
    Generally the gateway is not required in your organization. You need to configure Mediation Server setting. For the details about
    the SIP trunk configuration of ITSP side, you need to contact Gamme Support for further assistance.
    Regards,
    Kent Huang
    TechNet Community Support ************************************************************************************************************************ Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if a
    marked post does not actually answer your question.

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