SIP Trunk - No voice with Single Number Reach
Hi Community.
I setup SIP Trunk with the CCA. Everything is working Call In and Call Out. Call Forward and so on.
But with Single Number reach is something wrong. The mobile phone is ringing and I can get the call, but I hear not any voice.
Can someone please help me out? Below the config.
version 15.1
parser config cache interface
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
service compress-config
service sequence-numbers
dot11 ssid cisco-data
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.
Similar Messages
-
CUCM Mobility/Single Number Reach is not working correctly
Hi,
I'm experiencing difficulties with the CUCM Mobility (Single Number Reach) function.
For a customer of mine, I'm busy setting up the Mobility/Single Number Reach which is designed as follows:
- Users have a 4 digit DN that is attached to their User Device Profile so that they can use Extension Mobility.
- Those same users also have a Remote Destination Profile with a Remote Destination (to a mobile number) attached to their DN.
All has been set but as I was testing a couple of DNs, I noticed that a some numbers could be called to both their DN and Remote Destinations while others could only reach their DN.
As an example I have configured DN 6380 with the correct CSS (which permits to call to mobile phones and national numbers) to a User Device Profile.
That same DN is also connected to a Remote Destination Profile with a configured Remote Destination, which also has the same CSS.
The End User that is needed to login into Extention Mobility is 6380.
The settings on the Remote Destination are to ring always and all the time.
All Remote Destinations have the "Line Association" active.
Each DN has a value of 3 in maximum number of lines field.
Their Remote Destination profile has a value of 2 in maximum number of lines.
With this particular user, I'm sure that I have the right mobile number.
I already found out that the numbers that are having this problem, do not make a call to the PSTN and mobile network when their DN are called.
So I think the problem is within CUCM.
Can somebody help me?
Many thanks in advance!
The version of call manager is 8.6.2.20000-2.I did a debug isdn q931 on the voice router which is connected to the PRI circuit.
A couple of test calls later and this is the output I got:
Jun 28 08:34:15.737: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0E5A
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA18382
Preferred, Channel 2
Calling Party Number i = 0x2183, '51365XXXX'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '88777XXXX'
Plan:ISDN, Type:National
Sending Complete
Jun 28 08:34:15.749: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8E5A
Channel ID i = 0xA98382
Exclusive, Channel 2
This company has 200+ Mobility users so I did a random check.
Strangely enough, the one I described in my first post is now reachable on both his DN and mobile phone.
This is 4 digit number was the only one though.
Jun 28 08:09:20.557: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x00A6
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA18382
Preferred, Channel 2
Calling Party Number i = 0x2181, '51365XXXX'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '88777XXXX'
Plan:ISDN, Type:National
High Layer Compat i = 0x9181
Sending Complete
Jun 28 08:09:20.569: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x80A6
Channel ID i = 0xA98382
Exclusive, Channel 2
Jun 28 08:09:20.573: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x2345
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839C
Exclusive, Channel 28
Display i = 'Testnummer'
Called Party Number i = 0x80, '51365XXXX'
Plan:Unknown, Type:Unknown
Redirecting Number i = 0x00008F, '088777XXXX'
Plan:Unknown, Type:Unknown
I did a Remote Destination Profile export of all records with a Remote Destination attached to it and then re-imported them in CUCM.
The last output of the call to the mobile phone was not appearing last Thursday.
Apparently, the export and import of the profiles and destinations did change something within CUCM.
Could this be a bug in CUCM? -
Recording audio voice with single microphone but getting stereo output
I am trying to record a voice recording to a video using a Scarlett microphone. I can only record in Mono. How do I get the voice out of both speakers when playing back and/or record on both sides using one microphone?
After you add the sound to your timeline search the help file for the "fill left" or "fill right" command
-
SIP trunking between Microsoft OCS server and Cisco Voice GW router.
Hello All,
I have a client with an existing Microsoft OCS (office communications server) environment with the OCS server in their head office. The OCS clients in the remote Office registers with the OCS server in the head office. The WAN connectivity between the remote office and the Head office is MPLS.I would like to facilitate local call (PSTN) features at the remote site through a newly proposed Voice gateway router.
Can I achieve this by doing a SIP trunk between the OCS server in the head office to the newly proposed voice GW router in the remote office through the existing MPLS link. If yes, Could any one please assist me in this regards or suggest any other best solution to achieve the same.
Thank you in advance,
Mohammed Ameen RHi David,
this is a normal behaviour. To CUCM, OCS is a remote destination (just like your mobile phone). When your mobile phone hangs up, the system will put the call on hold for 10 sec.
This is there for the mobile user to go to his desk to pick up the call and continue the conversation (part of single number reach feature)
The best practise will be for the user to ensure that the other party hangs up the call first before he hang up.
Please grade if you think it's useful =) -
Problems between an UC520 and Asterisk with sip trunk
I have an UC520 and Asterisk with a sip trunk created between them, the calls from the UC520 to the Asterisk are ok, but the calls form de Asterisk to the UC520 are always busy.
Logs from the asterisk show that the first part of the call is ok, but the call is not complete, this means that the part where the extensions are with @ipuc520 doesn't appear
I created a sip trunk from de CCA 1.9 and it puts this for incoming calls for the dial peer, if I compare with a CCME, there is no configuration for incoming call there
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Tabla normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:Calibri;
mso-fareast-theme-font:minor-latin;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;
mso-fareast-language:EN-US;}
dial-peer voice 1000 voip
permission term
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 ipv4:x.y.z.w
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
And there is no configurarion at all that could block the calls
The x.y.z.w was the sip server ip (asterisk ip)
The comminication between sip and h323 are allowed in the four ways
The allowed codecs are g711ulaw and g729r8
Asterisk is working now with other CCME and they are ok so I copied the configuration from those CCME to the UC520 and from the other sip trunks in asterisk the new trunk sip for uc520
The sip trunk created from the CCA was replaces for the one from the CCME that is working now
The routes are ok in Asterisk.
There is no translation profile in incoming calls.
There is no ACL applied in all configuration.
There is no log about callres incoming from the asterisk.
Could anyone halp me pls?Hi Rina,
Help me to try and understand what you are trying to do.
In this code snippet i see the following:
001808: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=7129, Called Number=7129, Peer Info Type=DIALPEER_INFO_SPEECH
001809: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=7129
001810: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
001811: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20036
This looks as though you have a call coming in from the Asterisk system to number 7129, which then leads to this according to the config file you provided.
number 7129
label 7129
description7129
name 7129
call-forward busy 6001
call-forward noan 6001 timeout 10
Which at this point I am going to assume this is ephone-dn 10 (Please confirm). If this is the case then the inbound call is being matched correctly to a DN (Which has its own dial-peer tag "Dial-peer Tag=20036".
But then i see this:
001817: 1w3d: //-1/55940098BA19/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1000
001818: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=Unknown, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
001819: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
So the incoming call has been matched to Dial-peer 1000 which is an incoming VoIP dial-peer:
dial-peer voice 1000 voip
permission term
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
But then can see it has no where to go. So either I am reading this all wrong and the 7129 number is a result of another call taking place whilst you were debugging the system, or it is part of the debug and I am missing something here.
Rina, just so I understand this all. Are you trying to do WAN type calling from one system UC-500 (System "A") to the Asterisk system ( System B) and same? And so far calls going from the UC-500 to the Asterisk system are fine, but calls coming in from the Asterisk system to the UC-500 are not?
What happens on the Asterisk side when you try to call an Extension on the UC-500, do you get any ringing? Or is it a fast busy tone?
I am going to look over your configuration and debug a little further when I get home, maybe I am missing something here and can identify it.
Cheers,
David. -
Third Party Phone over SIP Trunk with CUCM 9.x
Hi all,
I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
Cisco Phone: INVITE sip.60xxxx%23@ipadress
Third Party SIP Phone: INVITE sip:[email protected]
It seems the Cisco phones gets some extra configured the Third Party ones dont...
Thanks in advance for any help.
//PerThanks for the answer
Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty. The termination Cause Code is that the number requested is Unallocated/Unassigned..
In other words something is missing to get CUCM to strip 60 from the Polycom phones dialed number and send it towards the SIP trunk like it does when the Cisco phones call it.
Unfortunatley i dont have the meens to attach the trace...
Thanks again for any help/advice
With regards, Per. -
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,
SethHi 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 -
HI, I installed the UC320 for a customer and they have 19 users, we are using sip trunk for voice traffic
it now encountered an annoying problem, The isp is doing the maintenance in recent period and their sip trunk is coming down and up occasionally at night. Whenever the sip trunk broke and come up again, the UC320 seems loss the sync with the wan, and it can work for 1 or 2 days and then the phone can not dial externally and also the incoming call have the problem, Yet the internal call is ok, whenever, this happened, we need to restart the uc320 to resume the service. I configured the auto maintainance happen at Sunday morning 3am , yet, there are times that the sip trunk broke happen on Monday night, then we usually get the complaint from the custom around Wed. or Thurs. and then we had to restart the system to resume the service. It is really troublesome. Do you have any idea how to deal with the problem. Is it a bug of cisco uc320? Is there any software update or any patch for this problem?
We are running 2.3.2(6) now.HI
Thank you for your reply, but the thing seems a bit more complex, our network configured as unregistered by the requirement of isp, and it works nicely. when the sip broke down and come up again, the pbx can work normally for 1 or two days and then it seems drifted away. and the problem at beginning is minor with only a few phones malfunction, and can be retored by restart the phone, but as the time goes by , the problems seemd deteriorated until all phones not working and we have to restart the pbx.
I check the external trunks, the status of sip is unregistered. it is required by isp to be configured so. it works nicely as long as the sip trunk is on.
Regard -
CUCM route calls diferents gateways/sip trunks
Hi at all, I have CUCM 6.1.1 and I want to route calls throughs diferents gateways or sip trunks.
I planned to do with route groups, but I can not add on a route group a H323 gateway and a SIP trunk at the same time.
How can route calls in different ways?
In the CUCM page "Route patterns" I want to make alternative routes, for example, the number 6666 is on route "666X" through a "gateway/route list", but if I can not contact by going this route I need to go through the alternative route "XXXX" through another "gateway/route list".
How can I make by going first to one pattern and then the other pattern?
Thanks!
FranOk thanks but one question more.... if I have a MGCP Gateway? Can I do this from my MGCP Gateway? or I need an H323 Gateway.
And another possibility.... I dont know if it's right....
Can I do this with Partitions and CSS?
This is for example I 'll have a CSS "Global" with Partitions (Primary and Secondary);
It could go the route first to 666x Gateway with CSS "Global" and partitions (Primary and Secondary). This way I do not know if it is routed first through the Gateway of the partition as Primary and Secondary alternative partition that is served by the SIP Trunk.
Using the "Dial Number Analyzer" I get the second path XXXX (SIP Trunk) as an alternative route ... -
Video only enabled when call is initiated from one direction across SIP Trunk
wonder If anyone can shed some light on this.
I have an issue between two cucm clusters, tied together with a SIP trunk.
If we dial from Australia to the US there is two way video and audio. If the US calls Australia, there is only audio. I have run a test call from the US through VLT and have found the following SDP's (see below). When The US make a video enabled call to australia the message "Video is not available, Remote party has video off" on the US phone screen.
Both clusters have the SIP trunk set up with the same codec settings and video bandwidth between reqions and locations. the SIP trunk is configured pretty much stock standard and identical at both ends, yet the SDP seem to want to negotiate different Video Parameters (again see SDP's below). CUCM in australia is 10.61.2.82.
what other settings can I check to get video to work when calls get initiated from either direction,...................
both phones are SIP 8941's, again audio is no problem in both directions.
=======this is from the phone in Australia to the CUCM in australia phone IP 10.61.4.112======================================
45870304.002 |09:02:07.941 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.61.4.112 on port 34271 index 53563 with 2089 bytes:
[344530309,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.61.2.82:5060;branch=z9hG4bKe0103892bbb75
From: "Anonymous" <sip:[email protected]>;tag=109791678~1b5af941-cea2-4a00-a0bd-15a532224d7d-59374526
To: <sip:[email protected]>;tag=5057a887bfdd550c0d321a20-7f843426
Call-ID: [email protected]
Date: Wed, 29 Apr 2015 23:02:07 GMT
CSeq: 101 INVITE
Server: Cisco-CP8941/9.4.2
Contact: <sip:[email protected]:34271;transport=tcp>;video
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Dennis Mink - 33935" <sip:[email protected]>;party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 966
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 28123 0 IN IP4 10.61.4.112
s=SIP Call
t=0 0
m=audio 16736 RTP/AVP 0 8 18 102 9 116 101
c=IN IP4 10.61.4.112
a=trafficclass:conversational.audio.avconf.aq:admitted
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 16738 RTP/AVP 126 97
c=IN IP4 10.61.4.112
b=TIAS:2000000
a=trafficclass:conversational.video.avconf.aq:admitted <----this is missing from US SDP
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428014;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200;max-rcmd-nalu-size=1300
a=imageattr:126 send * recv [x=640,y=480]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200
a=imageattr:97 send * recv [x=640,y=480]
a=rtcp-fb:* ccm tmmbr
a=sendrecv
============below is coming from the US (phone IP is 10.1.109.81)================
04/30/2015 09:02:08.169 Send 10.61.4.112 SIP ACK bfa99a00-541162ed-71da57-52023d0a NotAvail
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.61.4.112 on port 34271 index 53563
[344530326,NET]
ACK sip:[email protected]:34271;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.61.2.82:5060;branch=z9hG4bKe010481f320b08
From: "Anonymous" <sip:[email protected]>;tag=109791678~1b5af941-cea2-4a00-a0bd-15a532224d7d-59374526
To: <sip:[email protected]>;tag=5057a887bfdd550c0d321a20-7f843426
Date: Wed, 29 Apr 2015 23:02:05 GMT
Call-ID: [email protected]
User-Agent: Cisco-CUCM10.0
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 456
SDP Message
====================================================
v=0
o=CiscoSystemsCCM-SIP 109791678 1 IN IP4 10.61.2.82
s=SIP Call
c=IN IP4 10.1.109.81
b=TIAS:8000
b=AS:8
t=0 0
m=audio 16412 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=trafficclass:conversational.audio.aq:admitted <---what does this do here, and how?
m=video 0 RTP/SAVP 31 34 96 97 <-----------port 0. why?
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=rtpmap:97 H264/90000
a=content:main
a=inactiveHi Dennis,
On US phone SDP media attribute is inactive.
a=rtpmap:97 H264/90000
a=content:main
a=inactive
Are you sure that audio works ? Can you please share all the SIP messages of both the scenarios.
Thanks
Manish -
Issue with instant ringback when using sip trunk to SP
Hi all,
We use CUCM 8.0.2.
We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
c2900-universalk9-mz.SPA.150-1.M3.bin
Cisco CISCO2911/K9 (revision 1.0)
Technology Package License Information for Module:'c2900'
Technology Technology-package
Current Type
ipbase ipbasek9 Permanent
security securityk9 Permanent
uc uck9 Permanent
data None None
We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
We use 7945 and CIPC for our phones.
We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
Any ideas why this happens and how to stop it?
I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
voice service voip
address-hiding
mode border-element
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
header-passing error-passthru
early-offer forced
midcall-signaling passthru
interface GigabitEthernet0/0
ip address x.x.x.x 255.255.255.252
ip access-group acl.SIP-IN in
no ip redirects
no ip unreachables
ip verify unicast reverse-path
ip virtual-reassembly
duplex full
speed 100
no cdp enable
gateway
timer receive-rtp 1200
sip-ua
connection-reuse
gatekeeper
shutdown
dial-peer voice 1 voip
description *** INBOUND CALLS FROM CARRIER ***
translation-profile incoming SIPTRUNK-INCOMING
session protocol sipv2
incoming called-number #blah blah#
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 61 voip
description **** WA, SA AND NT NUMBERS ****
destination-pattern 0[8]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[8]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 81 voip
description **** MOBILE NUMBERS ****
destination-pattern 0[4]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[4]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 500 voip
description *** INBOUND SIP TRUNK TO CUCM PUB ***
translation-profile outgoing SIPTRUNK-CALLING-ADD-0
preference 1
destination-pattern 5[12]..
session protocol sipv2
session target ipv4:<OUR CUCM PUBLISHER IP>
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
Any help or a point in the right direction would be greatly appreciated.
Cheers,
BrettI ended up resolving this issue as follows:
In CUCM, under Device > Device Settings > SIP Profile.
I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
Now, I get the expected delay before hearing ringback.
Solved! -
Lync 2013 with SIP trunk with panasonic kx-tde200
Hi
My company has installed a panasonic ip-pbx kx-tde for multiline with 100 number range for telephone service.
Now my company is going to replace multiline by sip trunk . It will still work with Panasonic pbx box just need to reprogramme to be able to connected to the sip proxy which is managed by internet service provider.
For this scenario , would Lync 2013 voice work if I just add PSTN gateway which is the ip of panasonic pbx address to the frontend in topology ? Or I may need a mediation server as a must requirement to make lync voice work?
Thanks
WenFeiMedia bypass allow a call to basically skip the mediation server once it's established and go directly from gateway (in this case the PBX) and the endpoint (the telephone handset or Lync client) More information here: http://technet.microsoft.com/en-us/library/gg398719.aspx
By having this (if your PBX supports it) you reduce the load on the mediation servers. Before you go too far down the road also make sure that your PBX supports SIP trunks that are SIP over TCP (as Lync doesn't work with SIP over UDP)
Sort of, the easiest way is to add the .com as an additional SIP domain in Topology builder, you will need to create DNS records for it (both internal and external) and you will need to reissue the certs with additional SANs to support the second domain.
YOu will also need to update all the users to use the new suffix of xxx.com. So it's not a small task.
If this helped you please click "Vote As Helpful" if it answered your question please click "Mark As Answer" | Blog
www.lynced.com.au | Twitter
@imlynced -
ILBC calls via SIP Trunk with CUBE and CUCM7
hi there,
our SIP Provider offers the IBLC codec which promises to provide better quality compard to G.729.
I'm using this scenario:
IP-Phone(G711) --- CUCM7 --- (SIP-Trunk1) --- CUBE --- (SIP-Trunk2) --- Provider
Everything workes unless I'm configuring IBLC at the provider and on trunk2.
I have the CUBE router acting as a trancoding device and also specified IBLC as codec to be handled.
SIP trunk 2 was placed in a region with IBLC as codec.
On the trunk configuration in CUCM the media ressource group with XCODE capability is configured
Transcoding workes between two IP Phones in different regions with different codecs within the intranet.
Unfortunately the CUBE router doesn't seem to use the transcoder to change internal G711u calls into IBLC codec
so calls are blocked by the CUBE device:
deb ccsip calls
for incoming call:
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4AE7AC98
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0237892992
Called Number : 036677725231
Source IP Address (Sig ): 10.100.100.50
Destn SIP Req Addr:Port : <IP SIP Provicer>
Destn SIP Resp Addr:Port : <IP SIP Provicer>:5060
Destination Name : <IP SIP Provicer>
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : ilbc
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): <IP CUBE>
Source IP Port (Media): 0
Destn IP Address (Media): <IP SIP Provicer>
Destn IP Port (Media): 22022
Orig Destn IP Address:Port (Media): [ - ]:0
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 488
(Output lookes similar to outgoing calls)
I set up ccm on cube and assigned dsp ressources without success:
Here are the relevant configuration parts:
voice class codec 1
codec preference 1 iblc
voice service voip
address-hiding
allow-connections sip to sip
allow-connections h323 to sip
allow-connections sip to h323
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
h323
sip
header-passing error-passthru
no update-callerid
midcall-signaling passthru
privacy-policy passthru
voice-card 0
dspfarm
dsp services dspfarm
dial-peer voice 40991 voip
description *** Incoming from SIP-Provider
destination-pattern 03667772523.%
session protocol sipv2
session target ipv4:<IP_of_CUCM>
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
ip qos dscp cs5 media
ip qos dscp cs5 signaling
sccp local GigabitEthernet0/0
sccp ccm 10.100.100.50 identifier 11 version 4.1
sccp
sccp ccm group 11
description *** lokaler CCM fuer Codec-Konvertierung von SIP/DUS.NET
associate ccm 11 priority 1
associate profile 21 register DE_WGT_MTP02
dspfarm profile 21 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec ilbc
maximum sessions 10
associate application SCCP
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 10
sdspfarm tag 1 DE_WGT_MTP02
max-ephones 30
max-dn 30
ip source-address 10.100.100.50 port 2000
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 14 2010 02:10:34
sh sccp
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: 10.100.100.50
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.100.100.50, Port Number: 2000
Priority: N/A, Version: 4.1, Identifier: 11
Trustpoint: N/A
Call Manager: 10.1.1.55, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 10
Trustpoint: N/A
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.100.100.50, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 21
Reported Max Streams: 20, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
sh dspfarm dsp all
SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED
0 1 26.3.4 UP N/A FREE xcode 1 - - -
0 1 26.3.4 UP N/A FREE xcode 1 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 1 - - -
1 1 26.3.4 UP N/A FREE xcode 1 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
Thanks in advance,
DavidHi there,
Just wondering whether you ever got this resolved? I seem to have a very similiar problem.
Regards
Karen -
How to Remove port number for SIP trunk in CME
Hi,
I trying to set a SIP trunk with SIP provider, I have CME 7.1
The trunk is registered now but I can´t make calsl via SIP provider. After some debbugs sip provider's staff told me that the solution is not
not append the port in the INVITE.
Is it possible to do this?, How?
I have found some info about normalization but is relating to CM server not CME.
regardswhat port number does your provider use for signalling? They need to provide you the port number if its different from the standard 5060..
You can then configure the signalling ports on your dial-peer as shown in example below..where port 5081 is used here
dial-peer voice 1 voip
destination-pattern .T
session protocol sipv2
session target ipv4:10.10.10.24:5081
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"opportunity is a haughty goddess who waste no time with those who are unprepared" -
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
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