I'm hoping someone can help ? I'm trying to configure a SIP trunk on a test 2851 (adventerprisek9-mz.151-4.M12a ) that terminates on a 7911 phone (ext 100), but the incoming calls are being rejected. The phone is registered. The configuration is relatively simple :
7911 - 2851 - Internet - ITSP
I see an INVITE incoming, with a TRYING response from the 2851, then the 2851 sends the 500 error. So far I've been playing with the dial-peers but don't quite seem to be getting the configuration right, or there's something else wrong. I've attached our configuration and the output from :
debug ccsip messages
debug voice dial-peer all
Could someone point me in the right direction as to what might be wrong or what additional debugging I can collect ?
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I'm hoping someone here will be able to resolve an issue I'm having setting up a SIP trunk on a test 2851 router. I'm unable to make incoming calls to the trunk line, I can see the SIP INVITE but for some reason I'm yet to understand the call is being dropped. I've not got the outgoing dial peer configured yet.
The setup is simple :
7911 -- CE500 -- 2851 -- Internet
The 2851 is directly connected to the internet, no firewall, to make debugging simple, and the CE500 is there to provide power to the phone. The phone is registered and working, I can call 100 from the phone and it rings itself.
I've noticed that SIP Registration fails with "405 Method not allowed", but I'm getting an OPTIONS poll every 10 second with an OK reply from the router (included in the output).
I've attached the configuration and the output from,
debug voice dialpeer all debug ccsip all
and replaced the carriers ip with aaa.bbb.ccc.ddd, and our public with www.xxx.yyy.zzz.
Any suggestions gratefully received !
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Hi All I'm looking for some advice on the correct part numbers to suite our requirements. We have a 2851 (12.4(24)T3) running CME for our company and acting as a VPN server for our mobile users, with the following inventory cme#show inv NAME: "2851 chassis", DESCR: "2851 chassis" PID: CISCO2851 , VID: V04 , NAME: "One port E1 voice interface daughtercard on Slot 0 SubSlot 0", DESCR: "One port E1 voice interface daughtercard" PID: VWIC-1MFT-E1= , VID: 1.0, NAME: "2nd generation two port FXS voice interface daughtercard on Slot 0 SubSlot 1", DESCR: "2nd generation two port FXS voice interface daughtercard" PID: VIC2-2FXS , VID: V01 , NAME: "2nd generation two port FXO voice interface daughtercard on Slot 0 SubSlot 2", DESCR: "2nd generation two port FXO voice interface daughtercard" PID: VIC2-2FXO , VID: V01 , NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 4", DESCR: "PVDMII DSP SIMM with four DSPs" PID: PVDM2-64 , VID: V01 , NAME: "AIM Service Engine 0", DESCR: "AIM Service Engine" PID: AIM-CUE , VID: V02 , We have two ADSL lines with different public networks that I would like to attach to this device, with the intention of setting up a DNS round robin to point our mobile users at when setting up their VPN connection to the office. There aren't enough physical switch ports to do so with this device and so I would like to add some more, possibly with POE to increase the switching capacity in our computer room. Also, would I need an updated PSU to support POE. Please advise on the part number(s) required, and whether a EtherSwitch (NM), Etherwswitch (NME), or Etherswitch (HWIC) are appropriate. thanks in advance Simon
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Hi Marwan Yes, I had to modify the packet filter to allow all UDP traffic from the Skype IP's : ip access-list extended InternetAccess remark Remote UDP access to the Skype SIP peers permit udp host 220.127.116.11 any permit udp host 18.104.22.168 any permit udp host 22.214.171.124 any permit udp host 126.96.36.199 any permit udp host 188.8.131.52 any We don't have a skype ddi for each user, only one to the main attendant, so Also, all the ephone-dn's needed to be ammeded, such that number XXX no-reg primary kind regards Simon
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Hi there I've successfully setup Skype to allow incoming calls to our 2851 (+PVDM2) with CUE. Calls to phones are working in so much as the voice is going both ways and is clear. However, if the call is transfered to the CUE, or if the call is endpoint is the CUE directly, the voice from caller to CUE doesn't seem to be being received. I say that because the CUE doesn't record any message from the caller, in addition, the DTMF is being ignored. The attendant playback on the CUE can be heard. Here is a stripped copy of the configuration, and any help would be appreciated. Simon version 12.4 ! isdn switch-type primary-net5 ! ! voice call send-alert voice rtp send-recv ! voice service voip notify redirect ip2pots 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 moved-temporarily no supplementary-service sip refer supplementary-service ringback h225-info fax protocol cisco cause-code legacy h323 emptycapability h225 signal overlap h225 connect-passthru call service stop h245 passthru tcsnonstd-passthru sip header-passing registrar server expires max 3600 min 120 transport switch udp tcp ! ! ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw ! voice class codec 2 codec preference 4 g729r8 ! ! ! voice iec syslog ! voice translation-rule 1 rule 1 /^90/ /0/ ! voice translation-rule 2 rule 1 /1/ /019327550/ ! voice translation-rule 3 rule 1 // /0/ type national national rule 2 // /00/ type international international ! voice translation-rule 4 rule 1 /7550\(.\)\(\)/ /1\1\2/ rule 7 /7550/ /1/ rule 8 /781151/ /101/ rule 9 /786262/ /100/ rule 10 /1932779997/ /100/ ! voice translation-rule 5 rule 1 /^0$/ /960/ ! voice translation-rule 8 rule 1 /^118...$/ /118247/ ! voice translation-rule 9 rule 1 /^9\(.*\)/ /\1/ ! voice translation-rule 30 rule 1 /.*/ /640/ ! ! voice translation-profile AT_Outbound translate called 6 ! voice translation-profile BT_Outbound translate called 1 ! voice translation-profile DDI_Inbound translate called 4 ! voice translation-profile FAX_Outbound translate calling 2 ! voice translation-profile FAX_Outbound_2 translate calling 10 ! voice translation-profile GSM_Trunk_Outbound translate called 9 ! voice translation-profile Prefix-CLI-In translate calling 3 ! voice translation-profile Zero_Operator translate called 5 ! voice translation-profile skype_to_ccue translate called 30 ! ! voice-card 0 dspfarm dsp services dspfarm ! ! interface Loopback0 description Do not delete - SDM WebVPN generated interface$FW_INSIDE$ ip address 192.168.252.1 255.255.255.252 no ip proxy-arp ip flow ingress ip nat inside ip virtual-reassembly ! interface Loopback1 description $FW_INSIDE$ ip address 172.22.0.1 255.255.255.0 ! interface Loopback2 description Connecting IP gateway for the CUE ip address 172.21.1.1 255.255.255.252 ip flow ingress ! interface Null0 no ip unreachables ! interface GigabitEthernet0/0 description Internet interface$FW_OUTSIDE$$ETH-LAN$ ip address XXXXXXXXXX ip verify unicast reverse-path no ip proxy-arp ip flow ingress ip nat outside ip virtual-reassembly no ip mroute-cache duplex full speed 100 no cdp enable no mop enabled ! interface GigabitEthernet0/0.10 ! interface Service-Engine0/0 description $FW_INSIDE$ ip unnumbered Loopback1 service-module ip address 172.22.0.2 255.255.255.0 service-module ip default-gateway 172.22.0.1 ! interface GigabitEthernet0/1 description Internet interface no ip address no ip proxy-arp ip flow ingress duplex auto speed auto no cdp enable no mop enabled ! interface GigabitEthernet0/1.1 description $FW_INSIDE$ encapsulation dot1Q 1 native ip address 172.20.0.240 255.255.254.0 no ip proxy-arp ip flow ingress ! interface GigabitEthernet0/1.20 description $FW_INSIDE$ encapsulation dot1Q 20 ip address 172.21.0.1 255.255.255.0 ip flow ingress ! voice-port 0/0/0:15 translation-profile incoming Prefix-CLI-In cptone GB ! ! sccp local GigabitEthernet0/1.20 sccp ccm 172.20.0.240 identifier 1 version 4.0 sccp ! sccp ccm group 1 bind interface Loopback1 associate ccm 1 priority 1 associate profile 2 register skype_cue associate profile 1 register confprof1 keepalive retries 5 ! dspfarm profile 2 transcode description CCA transcoding for SIP Trunk Skype for SIP codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 6 associate application SCCP ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 associate application SCCP ! dial-peer voice 196 voip description Cisco Unity Express Pilot Number huntstop preference 1 destination-pattern 640 b2bua session protocol sipv2 session target ipv4:172.22.0.2 dtmf-relay sip-notify no vad ! dial-peer voice 99997 voip description Dial-peer for incoming skype calls translation-profile incoming skype_to_ccue huntstop preference 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server incoming called-number 99051000134036 dtmf-relay sip-notify rtp-nte ip qos dscp cs3 signaling no vad ! ! ! sip-ua credentials username XXXXXXXX password 7 XXXXXXXX realm sip.skype.com authentication username XXXXXXXX password 7 XXXXXXXX retry register 10 mwi-server ipv4:172.22.0.2 expires 120 port 5060 transport udp registrar dns:sip.skype.com expires 3600 presence enable ! ! ! telephony-service sdspfarm units 4 sdspfarm transcode sessions 128 sdspfarm tag 1 confprof1 sdspfarm tag 2 skype_cue conference hardware no auto-reg-ephone xml user admin password cisco 15 max-ephones 90 max-dn 120 ip source-address 172.21.0.1 port 2000 auto assign 1 to 10 calling-number initiator timeouts interdigit 4 timeouts ringing 20 system message Hello cnf-file location flash: network-locale GB load 7906 SCCP11.8-2-2SR2S load 7911 sccp11.8-4-2s load 7960-7940 P00308000500 load 7941 SCCP41.8-2-2SR2S load 7970 SCCP70.8-2-2SR2S load 7971 SCCP70.8-2-2SR2S load ata ATA030203SCCP051201A.zup time-zone 21 time-format 24 date-format dd-mm-yy live-record 222 voicemail 640 max-conferences 15 gain -6 call-forward pattern .T call-forward system redirecting-expanded hunt-group logout HLog dn-webedit time-webedit transfer-system full-consult transfer-pattern 9.T log password abcd login timeout 5 fac custom pickup group **3 create cnf-files version-stamp 7960 Sep 07 2011 11:31:39 ! ! end
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Hi, We've had our CCME/CUE installation running for a couple of months now, but I've noticed that all the IP Communicator clients we have (on 2k sp4 & xp sp2) are all consuming a substantial amount of memory. They start of OK, around 55-60MB consumed (as shown in the VM size column, not "Mem Usage" on task manager). As time goes on, this creeps up. It's not unusual for me to find it running up at 600+MB over 24 hours. Can other users confirm this, or shed light on to why this might be the case ? FWIW, I've disabled the "Enable Logging" feature where I've found it turned on, and we're using IPC 2.1.1, CCME 4.1 on IOS 12.4(11)T, on a 2851ISR. Any suggestions or comments appreciated Simon Delicata
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